Fri Jul 24 00:41:28 2009

Asterisk developer's documentation

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chan_sip.c File Reference

Implementation of Session Initiation Protocol. More...

#include "asterisk.h"
#include <ctype.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/signal.h>
#include <regex.h>
#include <time.h>
#include "asterisk/network.h"
#include "asterisk/paths.h"
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
#include "asterisk/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
#include "asterisk/tcptls.h"

Go to the source code of this file.

Data Structures
struct	__show_chan_arg
	argument for the 'show channels|subscriptions' callback. More...
struct	_map_x_s
	generic struct to map between strings and integers. Fill it with x-s pairs, terminate with an entry with s = NULL; Then you can call map_x_s(...) to map an integer to a string, and map_s_x() for the string -> integer mapping. More...
struct	ast_register_list
	The register list: Other SIP proxies we register with and place calls to. More...
struct	cfsip_methods
	The core structure to setup dialogs. We parse incoming messages by using structure and then route the messages according to the type. More...
struct	cfsip_options
	List of well-known SIP options. If we get this in a require, we should check the list and answer accordingly. More...
struct	cfsubscription_types
	Subscription types that we support. We support dialoginfo updates (really device status, not dialog info as was the original intent of the standard) SIMPLE presence used for device status Voicemail notification subscriptions. More...
struct	domain
	Domain data structure. More...
struct	domain_list
struct	find_call_cb_arg
	argument to the helper function to identify a call More...
struct	invstate2stringtable
	Readable descriptions of device states. More...
struct	peer_finding_info
struct	sip_auth
	sip_auth: Credentials for authentication to other SIP services More...
struct	sip_dual
	structure used in transfers More...
struct	sip_history
	sip_history: Structure for saving transactions within a SIP dialog More...
struct	sip_history_head
struct	sip_invite_param
	Parameters to the transmit_invite function. More...
struct	sip_mailbox
	A peer's mailbox. More...
struct	sip_peer
	Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host). More...
struct	sip_pkt
	sip packet - raw format for outbound packets that are sent or scheduled for transmission Packets are linked in a list, whose head is in the struct sip_pvt they belong to. Each packet holds a reference to the parent struct sip_pvt. This structure is allocated in __sip_reliable_xmit() and only for packets that require retransmissions. More...
struct	sip_proxy
	definition of a sip proxy server More...
struct	sip_pvt
	Structure used for each SIP dialog, ie. a call, a registration, a subscribe. Created and initialized by sip_alloc(), the descriptor goes into the list of descriptors (dialoglist). More...
struct	sip_pvt::request_queue
struct	sip_refer
	Structure to handle SIP transfers. Dynamically allocated when needed. More...
struct	sip_registry
	Registrations with other SIP proxies. More...
struct	sip_request
	sip_request: The data grabbed from the UDP socket More...
struct	sip_route
	Structure to save routing information for a SIP session. More...
struct	sip_settings
	a place to store all global settings for the sip channel driver More...
struct	sip_socket
	The SIP socket definition. More...
struct	sip_st_cfg
	Structure that encapsulates all attributes related to configuration of SIP Session-Timers feature on a per user/peer basis. More...
struct	sip_st_dlg
	Structure that encapsulates all attributes related to running SIP Session-Timers feature on a per dialog basis. More...
struct	sip_threadinfo
	Definition of a thread that handles a socket. More...
struct	t38properties
	T.38 channel settings (at some point we need to make this alloc'ed. More...
struct	threadl
	The thread list of TCP threads. More...
Object counters @{
Bug: These counters are not handled in a thread-safe way ast_atomic_fetchadd_int() should be used to modify these values.
#define	DEC_CALL_LIMIT   0
#define	DEC_CALL_RINGING   2
#define	INC_CALL_LIMIT   1
#define	INC_CALL_RINGING   3
#define	REQ_OFFSET_TO_STR(req, offset)   ((req)->data->str + ((req)->offset))
enum	domain_mode { SIP_DOMAIN_AUTO, SIP_DOMAIN_CONFIG }
	Modes for SIP domain handling in the PBX. More...
static int	apeerobjs = 0
static struct ast_flags	global_flags [2] = {{0}}
static struct io_context *	io
static pthread_t	monitor_thread = AST_PTHREADT_NULL
	This is the thread for the monitor which checks for input on the channels which are not currently in use.
static ast_mutex_t	monlock = ((ast_mutex_t) PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP )
	Protect the monitoring thread, so only one process can kill or start it, and not when it's doing something critical.
static ast_mutex_t	netlock = ((ast_mutex_t) PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP )
static int	regobjs = 0
static int	rpeerobjs = 0
static struct sched_context *	sched
static ast_mutex_t	sip_reload_lock = ((ast_mutex_t) PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP )
static int	sip_reloading = FALSE
static enum channelreloadreason	sip_reloadreason
static int *	sipsock_read_id
static int	speerobjs = 0
static char	used_context [AST_MAX_CONTEXT]
DefaultValues Default values, set and reset in reload_config before reading configuration
These are default values in the source. There are other recommended values in the sip.conf.sample for new installations. These may differ to keep backwards compatibility, yet encouraging new behaviour on new installations
#define	DEFAULT_ALLOW_EXT_DOM   TRUE
#define	DEFAULT_ALLOWGUEST   TRUE
#define	DEFAULT_AUTOCREATEPEER   FALSE
#define	DEFAULT_CALLCOUNTER   FALSE
#define	DEFAULT_CALLERID   "asterisk"
#define	DEFAULT_COMPACTHEADERS   FALSE
#define	DEFAULT_CONTEXT   "default"
#define	DEFAULT_COS_AUDIO   5
#define	DEFAULT_COS_SIP   4
#define	DEFAULT_COS_TEXT   5
#define	DEFAULT_COS_VIDEO   6
#define	DEFAULT_MAX_CALL_BITRATE   (384)
#define	DEFAULT_MOHINTERPRET   "default"
#define	DEFAULT_MOHSUGGEST   ""
#define	DEFAULT_NOTIFYMIME   "application/simple-message-summary"
#define	DEFAULT_NOTIFYRINGING   TRUE
#define	DEFAULT_PEDANTIC   FALSE
#define	DEFAULT_QUALIFY   FALSE
#define	DEFAULT_REALM   "asterisk"
#define	DEFAULT_REGEXTENONQUALIFY   FALSE
#define	DEFAULT_SDPOWNER   "root"
#define	DEFAULT_SDPSESSION   "Asterisk PBX"
#define	DEFAULT_SRVLOOKUP   TRUE
#define	DEFAULT_T1MIN   100
#define	DEFAULT_TOS_AUDIO   0
#define	DEFAULT_TOS_SIP   0
#define	DEFAULT_TOS_TEXT   0
#define	DEFAULT_TOS_VIDEO   0
#define	DEFAULT_USERAGENT   "Asterisk PBX"
#define	DEFAULT_VMEXTEN   "asterisk"
SIPflags
Various flags for the flags field in the pvt structure Trying to sort these up (one or more of the following): D: Dialog P: Peer/user G: Global flag When flags are used by multiple structures, it is important that they have a common layout so it is easy to copy them.
#define	SIP_CALL_LIMIT   (1 << 7)
#define	SIP_CAN_REINVITE   (1 << 20)
#define	SIP_CAN_REINVITE_NAT   (2 << 20)
#define	SIP_DEFER_BYE_ON_TRANSFER   (1 << 10)
#define	SIP_DTMF   (7 << 15)
#define	SIP_DTMF_AUTO   (3 << 15)
#define	SIP_DTMF_INBAND   (1 << 15)
#define	SIP_DTMF_INFO   (2 << 15)
#define	SIP_DTMF_RFC2833   (0 << 15)
#define	SIP_DTMF_SHORTINFO   (4 << 15)
#define	SIP_FLAGS_TO_COPY
	Flags to copy from peer/user to dialog.
#define	SIP_G726_NONSTANDARD   (1 << 31)
#define	SIP_GOTREFER   (1 << 6)
#define	SIP_INC_COUNT   (1 << 8)
#define	SIP_INC_RINGING   (1 << 9)
#define	SIP_INSECURE   (3 << 23)
#define	SIP_INSECURE_INVITE   (1 << 24)
#define	SIP_INSECURE_NONE   (0 << 23)
#define	SIP_INSECURE_PORT   (1 << 23)
#define	SIP_NAT   (3 << 18)
#define	SIP_NAT_ALWAYS   (3 << 18)
#define	SIP_NAT_NEVER   (0 << 18)
#define	SIP_NAT_RFC3581   (1 << 18)
#define	SIP_NAT_ROUTE   (2 << 18)
#define	SIP_NEEDREINVITE   (1 << 4)
#define	SIP_OUTGOING   (1 << 0)
#define	SIP_PENDINGBYE   (1 << 5)
#define	SIP_PROG_INBAND   (3 << 25)
#define	SIP_PROG_INBAND_NEVER   (0 << 25)
#define	SIP_PROG_INBAND_NO   (1 << 25)
#define	SIP_PROG_INBAND_YES   (2 << 25)
#define	SIP_PROGRESS_SENT   (1 << 3)
#define	SIP_PROMISCREDIR   (1 << 11)
#define	SIP_REINVITE   (7 << 20)
#define	SIP_REINVITE_NONE   (0 << 20)
#define	SIP_REINVITE_UPDATE   (4 << 20)
#define	SIP_RINGING   (1 << 2)
#define	SIP_SENDRPID   (1 << 29)
#define	SIP_TRUSTRPID   (1 << 12)
#define	SIP_USECLIENTCODE   (1 << 14)
#define	SIP_USEREQPHONE   (1 << 13)
SIPflags2
a second page of flags (for flags[1]
#define	SIP_PAGE2_ALLOWOVERLAP   (1 << 17)
#define	SIP_PAGE2_ALLOWSUBSCRIBE   (1 << 16)
#define	SIP_PAGE2_BUGGY_MWI   (1 << 26)
#define	SIP_PAGE2_CALL_ONHOLD   (3 << 23)
#define	SIP_PAGE2_CALL_ONHOLD_ACTIVE   (1 << 23)
#define	SIP_PAGE2_CALL_ONHOLD_INACTIVE   (3 << 23)
#define	SIP_PAGE2_CALL_ONHOLD_ONEDIR   (2 << 23)
#define	SIP_PAGE2_DIALOG_ESTABLISHED   (1 << 27)
#define	SIP_PAGE2_FLAGS_TO_COPY
#define	SIP_PAGE2_REGISTERTRYING   (1 << 29)
#define	SIP_PAGE2_RFC2833_COMPENSATE   (1 << 25)
#define	SIP_PAGE2_RTAUTOCLEAR   (1 << 2)
#define	SIP_PAGE2_RTCACHEFRIENDS   (1 << 0)
#define	SIP_PAGE2_STATECHANGEQUEUE   (1 << 9)
#define	SIP_PAGE2_SUBSCRIBEMWIONLY   (1 << 18)
#define	SIP_PAGE2_T38SUPPORT   (7 << 20)
#define	SIP_PAGE2_T38SUPPORT_RTP   (2 << 20)
#define	SIP_PAGE2_T38SUPPORT_TCP   (4 << 20)
#define	SIP_PAGE2_T38SUPPORT_UDPTL   (1 << 20)
#define	SIP_PAGE2_TEXTSUPPORT   (1 << 15)
#define	SIP_PAGE2_UDPTL_DESTINATION   (1 << 30)
#define	SIP_PAGE2_VIDEOSUPPORT   (1 << 14)
#define	SIP_PAGE2_VIDEOSUPPORT_ALWAYS   (1 << 31)
SIPflagsT38
T.38 set of flags
#define	T38FAX_FILL_BIT_REMOVAL   (1 << 0)
#define	T38FAX_RATE_12000   (1 << 12)
#define	T38FAX_RATE_14400   (1 << 13)
#define	T38FAX_RATE_2400   (1 << 8)
#define	T38FAX_RATE_4800   (1 << 9)
#define	T38FAX_RATE_7200   (1 << 10)
#define	T38FAX_RATE_9600   (1 << 11)
#define	T38FAX_RATE_MANAGEMENT_LOCAL_TCF   (1 << 3)
#define	T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF   (0 << 3)
#define	T38FAX_TRANSCODING_JBIG   (1 << 2)
#define	T38FAX_TRANSCODING_MMR   (1 << 1)
#define	T38FAX_UDP_EC_FEC   (1 << 4)
#define	T38FAX_UDP_EC_NONE   (0 << 4)
#define	T38FAX_UDP_EC_REDUNDANCY   (2 << 4)
#define	T38FAX_VERSION   (3 << 6)
#define	T38FAX_VERSION_0   (0 << 6)
#define	T38FAX_VERSION_1   (1 << 6)
static int	global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600
GlobalSettings
Global settings apply to the channel (often settings you can change in the general section of sip.conf
static int	allow_external_domains
static int	autocreatepeer
static int	compactheaders
static int	dumphistory
static int	global_allowguest
static int	global_allowsubscribe
static enum transfermodes	global_allowtransfer
static int	global_alwaysauthreject
static int	global_authfailureevents
static int	global_autoframing
static int	global_callcounter
static int	global_callevents
static int	global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263
	Codecs that we support by default:.
static unsigned int	global_cos_audio
static unsigned int	global_cos_sip
static unsigned int	global_cos_text
static unsigned int	global_cos_video
static int	global_directrtpsetup
static int	global_match_auth_username
static int	global_matchexterniplocally
static int	global_max_se
static int	global_min_se
static int	global_notifyhold
static int	global_notifyringing
static struct sip_proxy	global_outboundproxy
static int	global_qualifyfreq
static char	global_realm [MAXHOSTNAMELEN]
static int	global_reg_timeout
static int	global_regattempts_max
static char	global_regcontext [AST_MAX_CONTEXT]
static int	global_regextenonqualify
static int	global_relaxdtmf
static int	global_rtautoclear
static int	global_rtpholdtimeout
static int	global_rtpkeepalive
static int	global_rtptimeout
static char	global_sdpowner [AST_MAX_EXTENSION]
static char	global_sdpsession [AST_MAX_EXTENSION]
static int	global_srvlookup
static enum st_mode	global_st_mode
static enum st_refresher	global_st_refresher
static int	global_t1
static int	global_t1min
static int	global_timer_b
static unsigned int	global_tos_audio
static unsigned int	global_tos_sip
static unsigned int	global_tos_text
static unsigned int	global_tos_video
static char	global_useragent [AST_MAX_EXTENSION]
static int	pedanticsipchecking
static int	recordhistory
DefaultSettings
Default setttings are used as a channel setting and as a default when configuring devices
static char	default_callerid [AST_MAX_EXTENSION]
static char	default_context [AST_MAX_CONTEXT]
static char	default_fromdomain [AST_MAX_EXTENSION]
static char	default_language [MAX_LANGUAGE]
static int	default_maxcallbitrate
static char	default_mohinterpret [MAX_MUSICCLASS]
static char	default_mohsuggest [MAX_MUSICCLASS]
static char	default_notifymime [AST_MAX_EXTENSION]
static char	default_parkinglot [AST_MAX_CONTEXT]
static struct ast_codec_pref	default_prefs
static int	default_qualify
static char	default_subscribecontext [AST_MAX_CONTEXT]
static char	default_vmexten [AST_MAX_EXTENSION]
static struct sip_settings	sip_cfg
Defines
#define	ALLOWED_METHODS   "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
	SIP Methods we support.
#define	append_history(p, event, fmt, args...)   append_history_full(p, "%-15s " fmt, event, ## args)
	Append to SIP dialog history.
#define	CALLERID_UNKNOWN   "Unknown"
#define	CHECK_AUTH_BUF_INITLEN   256
#define	check_request_transport(peer, tmpl)
	generic function for determining if a correct transport is being used to contact a peer
#define	DEFAULT_DEFAULT_EXPIRY   120
#define	DEFAULT_EXPIRY   900
#define	DEFAULT_FREQ_NOTOK   10 * 1000
#define	DEFAULT_MAX_EXPIRY   3600
#define	DEFAULT_MAX_FORWARDS   "70"
#define	DEFAULT_MAX_SE   1800
#define	DEFAULT_MAXMS   2000
#define	DEFAULT_MIN_EXPIRY   60
#define	DEFAULT_MIN_SE   90
#define	DEFAULT_QUALIFYFREQ   60 * 1000
#define	DEFAULT_REGISTRATION_TIMEOUT   20
#define	DEFAULT_RETRANS   1000
#define	DEFAULT_TRANS_TIMEOUT   -1
#define	EXPIRY_GUARD_LIMIT   30
#define	EXPIRY_GUARD_MIN   500
#define	EXPIRY_GUARD_PCT   0.20
#define	EXPIRY_GUARD_SECS   15
#define	FALSE   0
#define	FINDALLDEVICES   (FINDUSERS | FINDPEERS)
#define	FINDPEERS   (1 << 1)
#define	FINDUSERS   (1 << 0)
#define	FORMAT   "%-15.15s %-10.10s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s\n"
#define	FORMAT   "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"
#define	FORMAT   "%-30.30s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"
#define	FORMAT   "%-40.40s %-20.20s %-16.16s\n"
#define	FORMAT   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"
#define	FORMAT   "%-30.30s %-6d %-9.9s %-6.6s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT2   "%-15.15s %-10.10s %-15.15s %-15.15s %-7.7s %-15.15s %-6.6s\n"
#define	FORMAT2   "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s (%-2.2s) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define	FORMAT2   "%-30.30s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"
#define	FORMAT2   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %s\n"
#define	FORMAT2   "%-30.30s %3.6s %9.9s %6.6s\n"
#define	FORMAT2   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT3   "%-15.15s %-10.10s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"
#define	FORMAT4   "%-15.15s %-10.10s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"
#define	INITIAL_CSEQ   101
#define	IS_SIP_TECH(t)   ((t) == &sip_tech || (t) == &sip_tech_info)
#define	MAX(a, b)   ((a) > (b) ? (a) : (b))
#define	MAX_AUTHTRIES   3
#define	MAX_HISTORY_ENTRIES   50
#define	MAX_RETRANS   6
#define	NO_RTP   0
#define	NOT_SUPPORTED   0
#define	RTP   1
#define	SDP_MAX_RTPMAP_CODECS   32
#define	SDP_SAMPLE_RATE(x)   8000
#define	SIP_MAX_HEADERS   64
#define	SIP_MAX_LINES   64
#define	SIP_MAX_PACKET   4096
#define	SIP_MIN_PACKET   1024
#define	SIP_OPT_100REL   (1 << 1)
#define	SIP_OPT_EARLY_SESSION   (1 << 3)
#define	SIP_OPT_EVENTLIST   (1 << 11)
#define	SIP_OPT_FROMCHANGE   (1 << 17)
#define	SIP_OPT_GRUU   (1 << 12)
#define	SIP_OPT_HISTINFO   (1 << 15)
#define	SIP_OPT_JOIN   (1 << 4)
#define	SIP_OPT_NOREFERSUB   (1 << 14)
#define	SIP_OPT_PATH   (1 << 5)
#define	SIP_OPT_PRECONDITION   (1 << 7)
#define	SIP_OPT_PREF   (1 << 6)
#define	SIP_OPT_PRIVACY   (1 << 8)
#define	SIP_OPT_RECLISTINV   (1 << 18)
#define	SIP_OPT_RECLISTSUB   (1 << 19)
#define	SIP_OPT_REPLACES   (1 << 0)
#define	SIP_OPT_RESPRIORITY   (1 << 16)
#define	SIP_OPT_SDP_ANAT   (1 << 9)
#define	SIP_OPT_SEC_AGREE   (1 << 10)
#define	SIP_OPT_TARGET_DIALOG   (1 << 13)
#define	SIP_OPT_TIMER   (1 << 2)
#define	SIP_OPT_UNKNOWN   (1 << 20)
#define	sip_pvt_lock(x)   ao2_lock(x)
#define	sip_pvt_trylock(x)   ao2_trylock(x)
#define	sip_pvt_unlock(x)   ao2_unlock(x)
#define	SIP_RESERVED   ";/?:@&=+$,# "
#define	SIP_TIMER_T1   500
#define	SIP_TRANS_TIMEOUT   64 * SIP_TIMER_T1
#define	SIPBUFSIZE   512
#define	STANDARD_SIP_PORT   5060
	Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS.
#define	STANDARD_TLS_PORT   5061
	Standard SIP TLS port for sips: from RFC 3261. DO NOT CHANGE THIS.
#define	SUPPORTED   1
#define	SUPPORTED_EXTENSIONS   "replaces, timer"
	SIP Extensions we support.
#define	TRUE   1
#define	UNLINK(element, head, prev)
#define	XMIT_ERROR   -2
Enumerations
enum	can_create_dialog { CAN_NOT_CREATE_DIALOG, CAN_CREATE_DIALOG, CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
	States whether a SIP message can create a dialog in Asterisk. More...
enum	check_auth_result {   AUTH_DONT_KNOW = -100, AUTH_SUCCESSFUL = 0, AUTH_CHALLENGE_SENT = 1, AUTH_SECRET_FAILED = -1,   AUTH_USERNAME_MISMATCH = -2, AUTH_NOT_FOUND = -3, AUTH_FAKE_AUTH = -4, AUTH_UNKNOWN_DOMAIN = -5,   AUTH_PEER_NOT_DYNAMIC = -6, AUTH_ACL_FAILED = -7, AUTH_BAD_TRANSPORT = -8 }
	Authentication result from check_auth* functions. More...
enum	invitestates {   INV_NONE = 0, INV_CALLING = 1, INV_PROCEEDING = 2, INV_EARLY_MEDIA = 3,   INV_COMPLETED = 4, INV_CONFIRMED = 5, INV_TERMINATED = 6, INV_CANCELLED = 7 }
	States for the INVITE transaction, not the dialog. More...
enum	parse_register_result { PARSE_REGISTER_FAILED, PARSE_REGISTER_UPDATE, PARSE_REGISTER_QUERY }
enum	referstatus {   REFER_IDLE, REFER_SENT, REFER_RECEIVED, REFER_CONFIRMED,   REFER_ACCEPTED, REFER_RINGING, REFER_200OK, REFER_FAILED,   REFER_NOAUTH }
	Parameters to know status of transfer. More...
enum	sip_auth_type { PROXY_AUTH = 407, WWW_AUTH = 401 }
	Authentication types - proxy or www authentication. More...
enum	sip_debug_e { sip_debug_none = 0, sip_debug_config = 1, sip_debug_console = 2 }
	debugging state We store separately the debugging requests from the config file and requests from the CLI. Debugging is enabled if either is set (which means that if sipdebug is set in the config file, we can only turn it off by reloading the config). More...
enum	sip_peer_type { SIP_TYPE_PEER = (1 << 0), SIP_TYPE_USER = (1 << 1) }
enum	sip_result { AST_SUCCESS = 0, AST_FAILURE = -1 }
	The result of a lot of functions. More...
enum	sip_transport { SIP_TRANSPORT_UDP = 1, SIP_TRANSPORT_TCP = 1 << 1, SIP_TRANSPORT_TLS = 1 << 2 }
	Define some implemented SIP transports. More...
enum	sipmethod {   SIP_UNKNOWN, SIP_RESPONSE, SIP_REGISTER, SIP_OPTIONS,   SIP_NOTIFY, SIP_INVITE, SIP_ACK, SIP_PRACK,   SIP_BYE, SIP_REFER, SIP_SUBSCRIBE, SIP_MESSAGE,   SIP_UPDATE, SIP_INFO, SIP_CANCEL, SIP_PUBLISH,   SIP_PING }
	SIP Request methods known by Asterisk. More...
enum	sipregistrystate {   REG_STATE_UNREGISTERED = 0, REG_STATE_REGSENT, REG_STATE_AUTHSENT, REG_STATE_REGISTERED,   REG_STATE_REJECTED, REG_STATE_TIMEOUT, REG_STATE_NOAUTH, REG_STATE_FAILED }
	States for outbound registrations (with register= lines in sip.conf. More...
enum	st_mode { SESSION_TIMER_MODE_INVALID = 0, SESSION_TIMER_MODE_ACCEPT, SESSION_TIMER_MODE_ORIGINATE, SESSION_TIMER_MODE_REFUSE }
	Modes in which Asterisk can be configured to run SIP Session-Timers. More...
enum	st_refresher { SESSION_TIMER_REFRESHER_AUTO, SESSION_TIMER_REFRESHER_UAC, SESSION_TIMER_REFRESHER_UAS }
	The entity playing the refresher role for Session-Timers. More...
enum	subscriptiontype {   NONE = 0, XPIDF_XML, DIALOG_INFO_XML, CPIM_PIDF_XML,   PIDF_XML, MWI_NOTIFICATION }
	Type of subscription, based on the packages we do support. More...
enum	t38_action_flag { SDP_T38_NONE = 0, SDP_T38_INITIATE, SDP_T38_ACCEPT }
enum	t38state {   T38_DISABLED = 0, T38_LOCAL_REINVITE, T38_PEER_DIRECT, T38_PEER_REINVITE,   T38_ENABLED }
	T38 States for a call. More...
enum	transfermodes { TRANSFER_OPENFORALL, TRANSFER_CLOSED }
	Authorization scheme for call transfers. More...
enum	xmittype { XMIT_CRITICAL = 2, XMIT_RELIABLE = 1, XMIT_UNRELIABLE = 0 }
	When sending a SIP message, we can send with a few options, depending on type of SIP request. UNRELIABLE is moslty used for responses to repeated requests, where the original response would be sent RELIABLE in an INVITE transaction. More...
Functions
static const char *	__get_header (const struct sip_request *req, const char *name, int *start)
static void	__init_check_auth_buf (void)
static void	__init_ts_temp_pvt (void)
	A per-thread temporary pvt structure.
static void	__reg_module (void)
static int	__set_address_from_contact (const char *fullcontact, struct sockaddr_in *sin, int tcp)
static void	__sip_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)
	Acknowledges receipt of a packet and stops retransmission called with p locked.
static int	__sip_autodestruct (const void *data)
	Kill a SIP dialog (called only by the scheduler) The scheduler has a reference to this dialog when p->autokillid != -1, and we are called using that reference. So if the event is not rescheduled, we need to call dialog_unref().
static void	__sip_destroy (struct sip_pvt *p, int lockowner, int lockdialoglist)
	Execute destruction of SIP dialog structure, release memory.
static int	__sip_do_register (struct sip_registry *r)
	Register with SIP proxy.
static void	__sip_pretend_ack (struct sip_pvt *p)
	Pretend to ack all packets called with p locked.
static int	__sip_reliable_xmit (struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod)
	Transmit packet with retransmits.
static int	__sip_semi_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)
	Acks receipt of packet, keep it around (used for provisional responses).
static int	__sip_xmit (struct sip_pvt *p, struct ast_str *data, int len)
	Transmit SIP message Sends a SIP request or response on a given socket (in the pvt) Called by retrans_pkt, send_request, send_response and __sip_reliable_xmit.
static int	__transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Base transmit response function.
static void	__unreg_module (void)
static char *	_sip_qualify_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Send qualify message to peer from cli or manager. Mostly for debugging.
static char *	_sip_show_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Show one peer in detail (main function).
static char *	_sip_show_peers (int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Execute sip show peers command.
static void *	_sip_tcp_helper_thread (struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session)
	SIP TCP thread management function.
static int	acf_channel_read (struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
static void	add_blank (struct sip_request *req)
	add a blank line if no body
static void	add_codec_to_sdp (const struct sip_pvt *p, int codec, int sample_rate, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_digit (struct sip_request *req, char digit, unsigned int duration, int mode)
	Add DTMF INFO tone to sip message Mode = 0 for application/dtmf-relay (Cisco) 1 for application/dtmf.
static int	add_header (struct sip_request *req, const char *var, const char *value)
	Add header to SIP message.
static int	add_header_contentLength (struct sip_request *req, int len)
	Add 'Content-Length' header to SIP message.
static int	add_line (struct sip_request *req, const char *line)
	Add content (not header) to SIP message.
static void	add_noncodec_to_sdp (const struct sip_pvt *p, int format, int sample_rate, struct ast_str **m_buf, struct ast_str **a_buf, int debug)
	Add RFC 2833 DTMF offer to SDP.
static void	add_peer_mailboxes (struct sip_peer *peer, const char *value)
static void	add_peer_mwi_subs (struct sip_peer *peer)
static struct sip_auth *	add_realm_authentication (struct sip_auth *authlist, const char *configuration, int lineno)
	Add realm authentication in list.
static void	add_route (struct sip_request *req, struct sip_route *route)
	Add route header into request per learned route.
static enum sip_result	add_sdp (struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
	Add Session Description Protocol message.
static int	add_sip_domain (const char *domain, const enum domain_mode mode, const char *context)
	Add SIP domain to list of domains we are responsible for.
static void	add_tcodec_to_sdp (const struct sip_pvt *p, int codec, int sample_rate, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add text codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_text (struct sip_request *req, const char *text)
	Add text body to SIP message.
static struct ast_variable *	add_var (const char *buf, struct ast_variable *list)
	implement the setvar config line
static void	add_vcodec_to_sdp (const struct sip_pvt *p, int codec, int sample_rate, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add video codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_vidupdate (struct sip_request *req)
	add XML encoded media control with update
static void	append_date (struct sip_request *req)
	Append date to SIP message.
static void	append_history_full (struct sip_pvt *p, const char *fmt,...)
	Append to SIP dialog history with arg list.
static void	append_history_va (struct sip_pvt *p, const char *fmt, va_list ap)
	Append to SIP dialog history with arg list.
static void	ast_quiet_chan (struct ast_channel *chan)
	Turn off generator data XXX Does this function belong in the SIP channel?
static void	ast_sip_ouraddrfor (struct in_addr *them, struct sockaddr_in *us)
	NAT fix - decide which IP address to use for Asterisk server?
static int	attempt_transfer (struct sip_dual *transferer, struct sip_dual *target)
	Attempt transfer of SIP call This fix for attended transfers on a local PBX.
static void	auth_headers (enum sip_auth_type code, char **header, char **respheader)
	return the request and response heade for a 401 or 407 code
static int	auto_congest (const void *arg)
	Scheduled congestion on a call. Only called by the scheduler, must return the reference when done.
static void	build_callid_pvt (struct sip_pvt *pvt)
	Build SIP Call-ID value for a non-REGISTER transaction.
static void	build_callid_registry (struct sip_registry *reg, struct in_addr ourip, const char *fromdomain)
	Build SIP Call-ID value for a REGISTER transaction.
static void	build_contact (struct sip_pvt *p)
	Build contact header - the contact header we send out.
static struct sip_peer *	build_peer (const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime)
	Build peer from configuration (file or realtime static/dynamic).
static int	build_reply_digest (struct sip_pvt *p, int method, char *digest, int digest_len)
	Build reply digest.
static void	build_route (struct sip_pvt *p, struct sip_request *req, int backwards)
	Build route list from Record-Route header.
static void	build_rpid (struct sip_pvt *p)
	Build the Remote Party-ID & From using callingpres options.
static void	build_via (struct sip_pvt *p)
	Build a Via header for a request.
static int	cb_extensionstate (char *context, char *exten, int state, void *data)
	Callback for the devicestate notification (SUBSCRIBE) support subsystem.
static void	change_t38_state (struct sip_pvt *p, int state)
	Change the T38 state on a SIP dialog.
static enum check_auth_result	check_auth (struct sip_pvt *p, struct sip_request *req, const char *username, const char *secret, const char *md5secret, int sipmethod, char *uri, enum xmittype reliable, int ignore)
	Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set).
static enum check_auth_result	check_peer_ok (struct sip_pvt *p, char *of, struct sip_request *req, int sipmethod, struct sockaddr_in *sin, struct sip_peer **authpeer, enum xmittype reliable, char *rpid_num, char *calleridname, char *uri2)
	Validate device authentication.
static void	check_pendings (struct sip_pvt *p)
	Check pending actions on SIP call.
static void	check_rtp_timeout (struct sip_pvt *dialog, time_t t)
	helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked
static int	check_sip_domain (const char *domain, char *context, size_t len)
	check_sip_domain: Check if domain part of uri is local to our server
static int	check_user (struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin)
	Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.
static enum check_auth_result	check_user_full (struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin, struct sip_peer **authpeer)
	Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.
static void	check_via (struct sip_pvt *p, struct sip_request *req)
	check Via: header for hostname, port and rport request/answer
static attribute_unused void	check_via_response (struct sip_pvt *p, struct sip_request *req)
	check received= and rport= in a SIP response. If we get a response with received= and/or rport= in the Via: line, use them as 'p->ourip' (see RFC 3581 for rport, and RFC 3261 for received). Using these two fields SIP can produce the correct address and port in the SIP headers without the need for STUN. The address part is also reused for the media sessions. Note that ast_sip_ouraddrfor() still rewrites p->ourip if you specify externip/seternaddr/stunaddr.
static void	cleanup_stale_contexts (char *new, char *old)
	Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.
static void	clear_peer_mailboxes (struct sip_peer *peer)
static int	clear_realm_authentication (struct sip_auth *authlist)
	Clear realm authentication list (at reload).
static void	clear_sip_domains (void)
	Clear our domain list (at reload).
static const char *	cli_yesno (int x)
	return Yes or No depending on the argument. This is used in many places in CLI command, having a function to generate this helps maintaining a consistent output (and possibly emitting the output in other languages, at some point).
static char *	complete_sip_peer (const char *word, int state, int flags2)
	Do completion on peer name.
static char *	complete_sip_registered_peer (const char *word, int state, int flags2)
	Do completion on registered peer name.
static char *	complete_sip_show_history (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show history' CLI.
static char *	complete_sip_show_peer (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show peer' CLI.
static char *	complete_sip_show_user (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show user' CLI.
static char *	complete_sip_unregister (const char *line, const char *word, int pos, int state)
	Support routine for 'sip unregister' CLI.
static char *	complete_sip_user (const char *word, int state)
	Do completion on user name.
static char *	complete_sipch (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show channel' and 'sip show history' CLI This is in charge of generating all strings that match a prefix in the given position. As many functions of this kind, each invokation has O(state) time complexity so be careful in using it.
static char *	complete_sipnotify (const char *line, const char *word, int pos, int state)
	Support routine for 'sip notify' CLI.
static int	copy_all_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy all headers from one request to another.
static int	copy_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy one header field from one request to another.
static void	copy_request (struct sip_request *dst, const struct sip_request *src)
	copy SIP request (mostly used to save request for responses)
static void	copy_socket_data (struct sip_socket *to_sock, const struct sip_socket *from_sock)
static struct ast_variable *	copy_vars (struct ast_variable *src)
static int	copy_via_headers (struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy SIP VIA Headers from the request to the response.
static int	create_addr (struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin, int newdialog)
	create address structure from device name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success
static int	create_addr_from_peer (struct sip_pvt *dialog, struct sip_peer *peer)
	Create address structure from peer reference. This function copies data from peer to the dialog, so we don't have to look up the peer again from memory or database during the life time of the dialog.
static void	destroy_association (struct sip_peer *peer)
	Remove registration data from realtime database or AST/DB when registration expires.
static void	destroy_mailbox (struct sip_mailbox *mailbox)
static int	determine_firstline_parts (struct sip_request *req)
	Parse first line of incoming SIP request.
static int	dialog_cmp_cb (void *obj, void *arg, int flags)
static int	dialog_dump_func (void *userobj, void *arg, int flags)
static int	dialog_hash_cb (const void *obj, const int flags)
static int	dialog_needdestroy (void *dialogobj, void *arg, int flags)
static struct sip_pvt *	dialog_ref (struct sip_pvt *p, char *tag)
	when we create or delete references, make sure to use these functions so we keep track of the refcounts. To simplify the code, we allow a NULL to be passed to dialog_unref().
static void *	dialog_unlink_all (struct sip_pvt *dialog, int lockowner, int lockdialoglist)
	Unlink a dialog from the dialogs container, as well as any other places that it may be currently stored.
static struct sip_pvt *	dialog_unref (struct sip_pvt *p, char *tag)
static void *	do_monitor (void *data)
	The SIP monitoring thread.
static int	do_proxy_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code, int sipmethod, int init)
	Add authentication on outbound SIP packet.
static int	do_register_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code)
	Authenticate for outbound registration.
static void	do_setnat (struct sip_pvt *p, int natflags)
	Set nat mode on the various data sockets.
static const char *	domain_mode_to_text (const enum domain_mode mode)
	Print domain mode to cli.
static const char *	dtmfmode2str (int mode)
	Convert DTMF mode to printable string.
static int	expire_register (const void *data)
	Expire registration of SIP peer.
static void	extract_uri (struct sip_pvt *p, struct sip_request *req)
	Check Contact: URI of SIP message.
static const char *	find_alias (const char *name, const char *_default)
	Find compressed SIP alias.
static int	find_by_name (void *obj, void *arg, int flags)
static struct sip_pvt *	find_call (struct sip_request *req, struct sockaddr_in *sin, const int intended_method)
	find or create a dialog structure for an incoming SIP message. Connect incoming SIP message to current dialog or create new dialog structure Returns a reference to the sip_pvt object, remember to give it back once done. Called by handle_incoming(), sipsock_read
static int	find_call_cb (void *__pvt, void *__arg, int flags)
static const char *	find_closing_quote (const char *start, const char *lim)
	Locate closing quote in a string, skipping escaped quotes. optionally with a limit on the search. start must be past the first quote.
static struct sip_peer *	find_peer (const char *peer, struct sockaddr_in *sin, int realtime, int which_objects, int devstate_only)
	Locate device by name or ip address.
static struct sip_auth *	find_realm_authentication (struct sip_auth *authlist, const char *realm)
	Find authentication for a specific realm.
static int	find_sdp (struct sip_request *req)
	Determine whether a SIP message contains an SDP in its body.
static int	find_sip_method (const char *msg)
	find_sip_method: Find SIP method from header
static struct cfsubscription_types *	find_subscription_type (enum subscriptiontype subtype)
	Find subscription type in array.
static void	free_old_route (struct sip_route *route)
	Remove route from route list.
static int	func_check_sipdomain (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	Dial plan function to check if domain is local.
static int	func_header_read (struct ast_channel *chan, const char *function, char *data, char *buf, size_t len)
	Read SIP header (dialplan function).
static int	function_sipchaninfo_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPCHANINFO()} Dialplan function - reads sip channel data
static int	function_sippeer (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPPEER()} Dialplan function - reads peer data
static char *	generate_random_string (char *buf, size_t size)
	Generate 32 byte random string for callid's etc.
static int	get_also_info (struct sip_pvt *p, struct sip_request *oreq)
	Call transfer support (old way, deprecated by the IETF)--.
static char *	get_body (struct sip_request *req, char *name)
	Get a specific line from the message body.
static char *	get_body_by_line (const char *line, const char *name, int nameLen)
	Reads one line of SIP message body.
static int	get_cached_mwi (struct sip_peer *peer, int *new, int *old)
	Get cached MWI info.
static char *	get_calleridname (const char *input, char *output, size_t outputsize)
	Get caller id name from SIP headers.
static int	get_destination (struct sip_pvt *p, struct sip_request *oreq)
	Find out who the call is for. We use the request uri as a destination. This code assumes authentication has been done, so that the device (peer/user) context is already set.
static const char *	get_header (const struct sip_request *req, const char *name)
	Get header from SIP request.
static char *	get_in_brackets (char *tmp)
	Pick out text in brackets from character string.
static struct ast_variable *	get_insecure_variable_from_config (struct ast_config *config)
static int	get_msg_text (char *buf, int len, struct sip_request *req, int addnewline)
	Get text out of a SIP MESSAGE packet.
static const char *	get_name_from_variable (struct ast_variable *var, const char *newpeername)
static void	get_our_media_address (struct sip_pvt *p, int needvideo, struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *tsin, struct sockaddr_in *dest, struct sockaddr_in *vdest)
	Set all IP media addresses for this call.
static int	get_rdnis (struct sip_pvt *p, struct sip_request *oreq)
	Get referring dnis.
static int	get_refer_info (struct sip_pvt *transferer, struct sip_request *outgoing_req)
	Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.
static int	get_rpid_num (const char *input, char *output, int maxlen)
	Get caller id number from Remote-Party-ID header field Returns true if number should be restricted (privacy setting found) output is set to NULL if no number found.
static const char *	get_sdp (struct sip_request *req, const char *name)
	Get a line from an SDP message body.
static const char *	get_sdp_iterate (int *start, struct sip_request *req, const char *name)
	Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.
static struct sip_pvt *	get_sip_pvt_byid_locked (const char *callid, const char *totag, const char *fromtag)
	Lock dialog lock and find matching pvt lock.
static const char *	get_transport (enum sip_transport t)
static const char *	get_transport_list (struct sip_peer *peer)
static const char *	get_transport_pvt (struct sip_pvt *p)
static const char *	gettag (const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize)
	Get tag from packet.
static int	handle_common_options (struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v)
	Handle flag-type options common to configuration of devices - peers.
static int	handle_incoming (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock)
	Handle incoming SIP requests (methods).
static int	handle_invite_replaces (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin)
	Handle the transfer part of INVITE with a replaces: header, meaning a target pickup or an attended transfer. Used only once. XXX 'ignore' is unused.
static int	handle_request_bye (struct sip_pvt *p, struct sip_request *req)
	Handle incoming BYE request.
static int	handle_request_cancel (struct sip_pvt *p, struct sip_request *req)
	Handle incoming CANCEL request.
static int	handle_request_do (struct sip_request *req, struct sockaddr_in *sin)
static void	handle_request_info (struct sip_pvt *p, struct sip_request *req)
	Receive SIP INFO Message.
static int	handle_request_invite (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e, int *nounlock)
	Handle incoming INVITE request.
static int	handle_request_message (struct sip_pvt *p, struct sip_request *req)
	Handle incoming MESSAGE request.
static int	handle_request_notify (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e)
	Handle incoming notifications.
static int	handle_request_options (struct sip_pvt *p, struct sip_request *req)
	Handle incoming OPTIONS request An OPTIONS request should be answered like an INVITE from the same UA, including SDP.
static int	handle_request_refer (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock)
static int	handle_request_register (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e)
	Handle incoming REGISTER request.
static int	handle_request_subscribe (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e)
	Handle incoming SUBSCRIBE request.
static void	handle_response (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle SIP response in dialogue.
static void	handle_response_invite (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle SIP response to INVITE dialogue.
static void	handle_response_notify (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
static void	handle_response_peerpoke (struct sip_pvt *p, int resp, struct sip_request *req)
	Handle qualification responses (OPTIONS).
static void	handle_response_refer (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
static int	handle_response_register (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle responses on REGISTER to services.
static const char *	hangup_cause2sip (int cause)
	Convert Asterisk hangup causes to SIP codes.
static int	hangup_sip2cause (int cause)
	Convert SIP hangup causes to Asterisk hangup causes.
static int	init_req (struct sip_request *req, int sipmethod, const char *recip)
	Initialize SIP request.
static int	init_resp (struct sip_request *resp, const char *msg)
	Initialize SIP response, based on SIP request.
static void	initialize_initreq (struct sip_pvt *p, struct sip_request *req)
	Initialize the initital request packet in the pvt structure. This packet is used for creating replies and future requests in a dialog.
static void	initreqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod)
	Initiate new SIP request to peer/user.
static const char *	insecure2str (int mode)
	Convert Insecure setting to printable string.
static void	list_route (struct sip_route *route)
	List all routes - mostly for debugging.
static int	load_module (void)
	PBX load module - initialization.
static int	local_attended_transfer (struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno)
	Find all call legs and bridge transferee with target called from handle_request_refer.
static int	lws2sws (char *msgbuf, int len)
	Parse multiline SIP headers into one header This is enabled if pedanticsipchecking is enabled.
static void	make_our_tag (char *tagbuf, size_t len)
	Make our SIP dialog tag.
static int	manager_show_registry (struct mansession *s, const struct message *m)
	Show SIP registrations in the manager API.
static int	manager_sip_qualify_peer (struct mansession *s, const struct message *m)
	Qualify SIP peers in the manager API.
static int	manager_sip_show_peer (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	manager_sip_show_peers (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	manager_sipnotify (struct mansession *s, const struct message *m)
static int	map_s_x (const struct _map_x_s *table, const char *s, int errorvalue)
	map from a string to an integer value, case insensitive. If no match is found, return errorvalue.
static const char *	map_x_s (const struct _map_x_s *table, int x, const char *errorstring)
	map from an integer value to a string. If no match is found, return errorstring
static int	method_match (enum sipmethod id, const char *name)
	returns true if 'name' (with optional trailing whitespace) matches the sip method 'id'. Strictly speaking, SIP methods are case SENSITIVE, but we do a case-insensitive comparison to be more tolerant. following Jon Postel's rule: Be gentle in what you accept, strict with what you send
static void	mwi_event_cb (const struct ast_event *event, void *userdata)
	Receive MWI events that we have subscribed to.
static const char *	nat2str (int nat)
	Convert NAT setting to text string.
static struct sip_proxy *	obproxy_get (struct sip_pvt *dialog, struct sip_peer *peer)
	Get default outbound proxy or global proxy.
static void	parse_copy (struct sip_request *dst, const struct sip_request *src)
	Copy SIP request, parse it.
static int	parse_minse (const char *p_hdrval, int *const p_interval)
	Session-Timers: Function for parsing Min-SE header.
static void	parse_moved_contact (struct sip_pvt *p, struct sip_request *req)
	Parse 302 Moved temporalily response.
static int	parse_ok_contact (struct sip_pvt *pvt, struct sip_request *req)
	Save contact header for 200 OK on INVITE.
static enum parse_register_result	parse_register_contact (struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
	Parse contact header and save registration (peer registration).
static int	parse_request (struct sip_request *req)
	Parse a SIP message.
static int	parse_session_expires (const char *p_hdrval, int *const p_interval, enum st_refresher *const p_ref)
	Session-Timers: Function for parsing Session-Expires header.
static unsigned int	parse_sip_options (struct sip_pvt *pvt, const char *supported)
	Parse supported header in incoming packet.
static int	parse_uri (char *uri, char *scheme, char **ret_name, char **pass, char **domain, char **port, char **options)
	* parses a URI in its components.
static int	peer_cmp_cb (void *obj, void *arg, int flags)
static int	peer_dump_func (void *userobj, void *arg, int flags)
static int	peer_hash_cb (const void *obj, const int flags)
static int	peer_ipcmp_cb (void *obj, void *arg, int flags)
static int	peer_iphash_cb (const void *obj, const int flags)
static int	peer_is_marked (void *peerobj, void *arg, int flags)
static void	peer_mailboxes_to_str (struct ast_str **mailbox_str, struct sip_peer *peer)
	list peer mailboxes to CLI
static int	peer_markall_func (void *device, void *arg, int flags)
static int	peer_status (struct sip_peer *peer, char *status, int statuslen)
int	peercomparefunc (const void *a, const void *b)
static void	print_codec_to_cli (int fd, struct ast_codec_pref *pref)
	Print codec list from preference to CLI/manager.
static void	print_group (int fd, ast_group_t group, int crlf)
	Print call group and pickup group.
static void	proc_422_rsp (struct sip_pvt *p, struct sip_request *rsp)
	Handle 422 response to INVITE with session-timer requested.
static int	proc_session_timer (const void *vp)
	Session-Timers: Process session refresh timeout event.
static void	process_request_queue (struct sip_pvt *p, int *recount, int *nounlock)
static int	process_sdp (struct sip_pvt *p, struct sip_request *req, int t38action)
	Process SIP SDP offer, select formats and activate RTP channels If offer is rejected, we will not change any properties of the call Return 0 on success, a negative value on errors. Must be called after find_sdp().
static struct sip_proxy *	proxy_allocate (char *name, char *port, int force)
	Allocate and initialize sip proxy.
static int	proxy_update (struct sip_proxy *proxy)
static int	queue_request (struct sip_pvt *p, const struct sip_request *req)
static struct sip_peer *	realtime_peer (const char *newpeername, struct sockaddr_in *sin, int devstate_only)
	realtime_peer: Get peer from realtime storage Checks the "sippeers" realtime family from extconfig.conf Checks the "sipregs" realtime family from extconfig.conf if it's configured. This returns a pointer to a peer and because we use build_peer, we can rest assured that the refcount is bumped.
static void	realtime_update_peer (const char *peername, struct sockaddr_in *sin, const char *defaultuser, const char *fullcontact, const char *useragent, int expirey, int deprecated_username, int lastms)
	Update peer object in realtime storage If the Asterisk system name is set in asterisk.conf, we will use that name and store that in the "regserver" field in the sippeers table to facilitate multi-server setups.
static void	receive_message (struct sip_pvt *p, struct sip_request *req)
	Receive SIP MESSAGE method messages.
static struct sip_peer *	ref_peer (struct sip_peer *peer, char *tag)
static struct sip_proxy *	ref_proxy (struct sip_pvt *pvt, struct sip_proxy *proxy)
	maintain proper refcounts for a sip_pvt's outboundproxy
static const char *	referstatus2str (enum referstatus rstatus)
	Convert transfer status to string.
static void	reg_source_db (struct sip_peer *peer)
	Get registration details from Asterisk DB.
static void	register_peer_exten (struct sip_peer *peer, int onoff)
	Automatically add peer extension to dial plan.
static enum check_auth_result	register_verify (struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req, char *uri)
	Verify registration of user Registration is done in several steps, first a REGISTER without auth to get a challenge (nonce) then a second one with auth Registration requests are only matched with peers that are marked as "dynamic".
static struct sip_registry *	registry_addref (struct sip_registry *reg, char *tag)
	Add object reference to SIP registry.
static void *	registry_unref (struct sip_registry *reg, char *tag)
static const char *	regstate2str (enum sipregistrystate regstate)
	Convert registration state status to string.
static int	reload (void)
	Part of Asterisk module interface.
static int	reload_config (enum channelreloadreason reason)
	Re-read SIP.conf config file.
static char *	remove_uri_parameters (char *uri)
static void	replace_cid (struct sip_pvt *p, const char *rpid_num, const char *calleridname)
	helper function for check_{user|peer}_ok()
static int	reply_digest (struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len)
	reply to authentication for outbound registrations
static int	reqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch)
	Initialize a SIP request message (not the initial one in a dialog).
static int	resp_needs_contact (const char *msg, enum sipmethod method)
	Test if this response needs a contact header.
static int	respprep (struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Prepare SIP response packet.
static int	restart_monitor (void)
	Start the channel monitor thread.
static void	restart_session_timer (struct sip_pvt *p)
	Session-Timers: Restart session timer.
static int	retrans_pkt (const void *data)
	Retransmit SIP message if no answer (Called from scheduler).
static int	scheduler_process_request_queue (const void *data)
static int	send_request (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
	Send SIP Request to the other part of the dialogue.
static int	send_response (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
	Transmit response on SIP request.
static int	set_address_from_contact (struct sip_pvt *pvt)
	Change the other partys IP address based on given contact.
static void	set_destination (struct sip_pvt *p, char *uri)
	Set destination from SIP URI.
static void	set_insecure_flags (struct ast_flags *flags, const char *value, int lineno)
	Parse insecure= setting in sip.conf and set flags according to setting.
static void	set_peer_defaults (struct sip_peer *peer)
	Set peer defaults before configuring specific configurations.
static void	set_t38_capabilities (struct sip_pvt *p)
	Set the global T38 capabilities on a SIP dialog structure.
static int	show_channels_cb (void *__cur, void *__arg, int flags)
	callback for show channel|subscription
static int	show_chanstats_cb (void *__cur, void *__arg, int flags)
	Callback for show_chanstats.
static int	sip_addheader (struct ast_channel *chan, void *data)
	Add a SIP header to an outbound INVITE.
static struct sip_pvt *	sip_alloc (ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method)
	Allocate sip_pvt structure, set defaults and link in the container. Returns a reference to the object so whoever uses it later must remember to release the reference.
static void	sip_alreadygone (struct sip_pvt *dialog)
	Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging.
static int	sip_answer (struct ast_channel *ast)
	sip_answer: Answer SIP call , send 200 OK on Invite Part of PBX interface
static int	sip_call (struct ast_channel *ast, char *dest, int timeout)
	Initiate SIP call from PBX used from the dial() application.
static int	sip_cancel_destroy (struct sip_pvt *p)
	Cancel destruction of SIP dialog. Be careful as this also absorbs the reference - if you call it from within the scheduler, this might be the last reference.
static char *	sip_cli_notify (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Cli command to send SIP notify to peer.
static int	sip_debug_test_addr (const struct sockaddr_in *addr)
	See if we pass debug IP filter.
static int	sip_debug_test_pvt (struct sip_pvt *p)
	Test PVT for debugging output.
static struct sip_pvt *	sip_destroy (struct sip_pvt *p)
	Destroy SIP call structure. Make it return NULL so the caller can do things like foo = sip_destroy(foo); and reduce the chance of bugs due to dangling pointers.
static void	sip_destroy_fn (void *p)
static void	sip_destroy_peer (struct sip_peer *peer)
	Destroy peer object from memory.
static void	sip_destroy_peer_fn (void *peer)
static int	sip_devicestate (void *data)
	Part of PBX channel interface.
static char *	sip_do_debug (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Turn on SIP debugging (CLI command).
static char *	sip_do_debug_ip (int fd, char *arg)
	Enable SIP Debugging for a single IP.
static char *	sip_do_debug_peer (int fd, char *arg)
	Turn on SIP debugging for a given peer.
static char *	sip_do_history_deprecated (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Enable/Disable SIP History logging (CLI) - deprecated. use sip_set_history instead.
static int	sip_do_reload (enum channelreloadreason reason)
	Reload module.
static int	sip_dtmfmode (struct ast_channel *chan, void *data)
	Set the DTMFmode for an outbound SIP call (application).
static void	sip_dump_history (struct sip_pvt *dialog)
	Dump SIP history to debug log file at end of lifespan for SIP dialog.
static int	sip_fixup (struct ast_channel *oldchan, struct ast_channel *newchan)
	sip_fixup: Fix up a channel: If a channel is consumed, this is called. Basically update any ->owner links
static const char *	sip_get_callid (struct ast_channel *chan)
	Deliver SIP call ID for the call.
static int	sip_get_codec (struct ast_channel *chan)
	Return SIP UA's codec (part of the RTP interface).
static enum ast_rtp_get_result	sip_get_rtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite audio (part of RTP interface).
static enum ast_rtp_get_result	sip_get_trtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite text (part of RTP interface).
static struct ast_udptl *	sip_get_udptl_peer (struct ast_channel *chan)
static enum ast_rtp_get_result	sip_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite video (part of RTP interface).
static int	sip_hangup (struct ast_channel *ast)
	sip_hangup: Hangup SIP call Part of PBX interface, called from ast_hangup
static int	sip_indicate (struct ast_channel *ast, int condition, const void *data, size_t datalen)
	Play indication to user With SIP a lot of indications is sent as messages, letting the device play the indication - busy signal, congestion etc.
static const char *	sip_nat_mode (const struct sip_pvt *p)
	Display SIP nat mode.
static struct ast_channel *	sip_new (struct sip_pvt *i, int state, const char *title)
	Initiate a call in the SIP channel called from sip_request_call (calls from the pbx ) for outbound channels and from handle_request_invite for inbound channels.
static int	sip_park (struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno)
	Park a call using the subsystem in res_features.c This is executed in a separate thread.
static void *	sip_park_thread (void *stuff)
	Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup.
static int	sip_parse_host (char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport)
	Small function to parse a config line for a host with a transport i.e. tls://www.google.com:8056.
static void	sip_peer_hold (struct sip_pvt *p, int hold)
	Change onhold state of a peer using a pvt structure.
static void	sip_poke_all_peers (void)
	Send a poke to all known peers Space them out 100 ms apart XXX We might have a cool algorithm for this or use random - any suggestions?
static int	sip_poke_noanswer (const void *data)
	React to lack of answer to Qualify poke.
static int	sip_poke_peer (struct sip_peer *peer, int force)
	Check availability of peer, also keep NAT open.
static int	sip_poke_peer_s (const void *data)
	Poke peer (send qualify to check if peer is alive and well).
static int	sip_prepare_socket (struct sip_pvt *p)
static char *	sip_prune_realtime (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Remove temporary realtime objects from memory (CLI).
static char *	sip_qualify_peer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Send an OPTIONS packet to a SIP peer.
static int	sip_queryoption (struct ast_channel *chan, int option, void *data, int *datalen)
	Query an option on a SIP dialog.
static struct ast_frame *	sip_read (struct ast_channel *ast)
	Read SIP RTP from channel.
static struct sockaddr_in *	sip_real_dst (const struct sip_pvt *p)
	The real destination address for a write.
static int	sip_refer_allocate (struct sip_pvt *p)
	Allocate SIP refer structure.
static int	sip_reg_timeout (const void *data)
	Registration timeout, register again Registered as a timeout handler during transmit_register(), to retransmit the packet if a reply does not come back. This is called by the scheduler so the event is not pending anymore when we are called.
static int	sip_register (const char *value, int lineno)
	Parse register=> line in sip.conf and add to registry.
static void	sip_registry_destroy (struct sip_registry *reg)
	Destroy registry object Objects created with the register= statement in static configuration.
static int	sip_reinvite_retry (const void *data)
	Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite to avoid race conditions between asterisk servers. Called from the scheduler.
static char *	sip_reload (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Force reload of module from cli.
static struct ast_channel *	sip_request_call (const char *type, int format, void *data, int *cause)
	PBX interface function -build SIP pvt structure SIP calls initiated by the PBX arrive here.
static int	sip_reregister (const void *data)
	Update registration with SIP Proxy. Called from the scheduler when the previous registration expires, so we don't have to cancel the pending event. We assume the reference so the sip_registry is valid, since it is stored in the scheduled event anyways.
static struct ast_frame *	sip_rtp_read (struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
	Read RTP from network.
static void	sip_scheddestroy (struct sip_pvt *p, int ms)
	Schedule destruction of SIP dialog.
static void	sip_send_all_registers (void)
	Send all known registrations.
static int	sip_send_mwi_to_peer (struct sip_peer *peer, const struct ast_event *event, int cache_only)
	Send message waiting indication to alert peer that they've got voicemail.
static int	sip_senddigit_begin (struct ast_channel *ast, char digit)
static int	sip_senddigit_end (struct ast_channel *ast, char digit, unsigned int duration)
	Send DTMF character on SIP channel within one call, we're able to transmit in many methods simultaneously.
static int	sip_sendhtml (struct ast_channel *chan, int subclass, const char *data, int datalen)
	Send message with Access-URL header, if this is an HTML URL only!
static int	sip_sendtext (struct ast_channel *ast, const char *text)
	Send SIP MESSAGE text within a call Called from PBX core sendtext() application.
static char *	sip_set_history (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Enable/Disable SIP History logging (CLI).
static void	sip_set_redirstr (struct sip_pvt *p, char *reason)
	Translate referring cause.
static int	sip_set_rtp_peer (struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
	Set the RTP peer for this call.
static int	sip_set_udptl_peer (struct ast_channel *chan, struct ast_udptl *udptl)
static char *	sip_show_channel (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show details of one active dialog.
static char *	sip_show_channels (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI for show channels or subscriptions. This is a new-style CLI handler so a single function contains the prototype for the function, the 'generator' to produce multiple entries in case it is required, and the actual handler for the command.
static char *	sip_show_channelstats (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	SIP show channelstats CLI (main function).
static char *	sip_show_domains (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI command to list local domains.
static char *	sip_show_history (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show history details of one dialog.
static char *	sip_show_inuse (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Command to show calls within limits set by call_limit.
static char *	sip_show_objects (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	List all allocated SIP Objects (realtime or static).
static char *	sip_show_peer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show one peer in detail.
static char *	sip_show_peers (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Show Peers command.
static char *	sip_show_registry (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show SIP Registry (registrations with other SIP proxies.
static char *	sip_show_sched (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	sip_show_settings (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	List global settings for the SIP channel.
static char *	sip_show_tcp (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show active TCP connections.
static char *	sip_show_user (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show one user in detail.
static char *	sip_show_users (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Command 'SIP Show Users'.
static int	sip_sipredirect (struct sip_pvt *p, const char *dest)
	Transfer call before connect with a 302 redirect.
static struct sip_st_dlg *	sip_st_alloc (struct sip_pvt *const p)
	Allocate Session-Timers struct w/in dialog.
static int	sip_standard_port (enum sip_transport type, int port)
	Returns the port to use for this socket.
static int	sip_t38_abort (const void *data)
static struct ast_tcptls_session_instance *	sip_tcp_locate (struct sockaddr_in *s)
	Find thread for TCP/TLS session (based on IP/Port.
static void *	sip_tcp_worker_fn (void *data)
	SIP TCP connection handler.
static int	sip_transfer (struct ast_channel *ast, const char *dest)
	Transfer SIP call.
static char *	sip_unregister (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Unregister (force expiration) a SIP peer in the registry via CLI.
static int	sip_uri_cmp (const char *input1, const char *input2)
static int	sip_uri_headers_cmp (const char *input1, const char *input2)
	helper routine for sip_uri_cmp
static int	sip_uri_params_cmp (const char *input1, const char *input2)
	helper routine for sip_uri_cmp
static int	sip_write (struct ast_channel *ast, struct ast_frame *frame)
	Send frame to media channel (rtp).
static int	sipsock_read (int *id, int fd, short events, void *ignore)
	Read data from SIP socket.
static enum st_mode	st_get_mode (struct sip_pvt *p)
	Get the session-timer mode.
static enum st_refresher	st_get_refresher (struct sip_pvt *p)
	Get the entity (UAC or UAS) that's acting as the session-timer refresher.
static int	st_get_se (struct sip_pvt *p, int max)
	Get Max or Min SE (session timer expiry).
static void	start_session_timer (struct sip_pvt *p)
	Session-Timers: Start session timer.
static const char *	stmode2str (enum st_mode m)
static void	stop_media_flows (struct sip_pvt *p)
	Immediately stop RTP, VRTP and UDPTL as applicable.
static void	stop_session_timer (struct sip_pvt *p)
	Session-Timers: Stop session timer.
static int	str2dtmfmode (const char *str)
	maps a string to dtmfmode, returns -1 on error
static enum st_mode	str2stmode (const char *s)
static enum st_refresher	str2strefresher (const char *s)
static const char *	strefresher2str (enum st_refresher r)
static const char *	subscription_type2str (enum subscriptiontype subtype)
	Show subscription type in string format.
static int	t38_get_rate (int t38cap)
	Get Max T.38 Transmission rate from T38 capabilities.
static struct sip_peer *	temp_peer (const char *name)
	Create temporary peer (used in autocreatepeer mode).
static void	temp_pvt_cleanup (void *)
static int	temp_pvt_init (void *)
static char *	terminate_uri (char *uri)
static char *	transfermode2str (enum transfermodes mode)
	Convert transfer mode to text string.
static void	transmit_fake_auth_response (struct sip_pvt *p, int sipmethod, struct sip_request *req, enum xmittype reliable)
	Send a fake 401 Unauthorized response when the administrator wants to hide the names of local devices from fishers.
static int	transmit_info_with_digit (struct sip_pvt *p, const char digit, unsigned int duration)
	Send SIP INFO dtmf message, see Cisco documentation on cisco.com.
static int	transmit_info_with_vidupdate (struct sip_pvt *p)
	Send SIP INFO with video update request.
static int	transmit_invite (struct sip_pvt *p, int sipmethod, int sdp, int init)
	Build REFER/INVITE/OPTIONS message and transmit it.
static int	transmit_message_with_text (struct sip_pvt *p, const char *text)
	Transmit text with SIP MESSAGE method.
static int	transmit_notify_custom (struct sip_pvt *p, struct ast_variable *vars)
	Notify device with custom headers from sip_notify.conf.
static int	transmit_notify_with_mwi (struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten)
	Notify user of messages waiting in voicemail (RFC3842).
static int	transmit_notify_with_sipfrag (struct sip_pvt *p, int cseq, char *message, int terminate)
	Notify a transferring party of the status of transfer (RFC3515).
static int	transmit_refer (struct sip_pvt *p, const char *dest)
	Transmit SIP REFER message (initiated by the transfer() dialplan application.
static int	transmit_register (struct sip_registry *r, int sipmethod, const char *auth, const char *authheader)
	Transmit register to SIP proxy or UA auth = NULL on the initial registration (from sip_reregister()).
static int	transmit_reinvite_with_sdp (struct sip_pvt *p, int t38version, int oldsdp)
	Transmit reinvite with SDP.
static int	transmit_request (struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
	Transmit generic SIP request returns XMIT_ERROR if transmit failed with a critical error (don't retry).
static int	transmit_request_with_auth (struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
	Transmit SIP request, auth added.
static int	transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, no retransmits.
static int	transmit_response_reliable (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, Make sure you get an ACK This is only used for responses to INVITEs, where we need to make sure we get an ACK.
static int	transmit_response_using_temp (ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg)
	Transmit response, no retransmits, using a temporary pvt structure.
static int	transmit_response_with_allow (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Append Accept header, content length before transmitting response.
static int	transmit_response_with_auth (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale)
	Respond with authorization request.
static int	transmit_response_with_date (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Append date and content length before transmitting response.
static int	transmit_response_with_minse (struct sip_pvt *p, const char *msg, const struct sip_request *req, int minse_int)
	Transmit 422 response with Min-SE header (Session-Timers).
static int	transmit_response_with_sdp (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_t38_sdp (struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_unsupported (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported)
	Transmit response, no retransmits.
static int	transmit_state_notify (struct sip_pvt *p, int state, int full, int timeout)
	Used in the SUBSCRIBE notification subsystem (RFC3265).
static void	try_suggested_sip_codec (struct sip_pvt *p)
	Try setting codec suggested by the SIP_CODEC channel variable.
static int	unload_module (void)
	PBX unload module API.
static void *	unref_peer (struct sip_peer *peer, char *tag)
static int	update_call_counter (struct sip_pvt *fup, int event)
	update_call_counter: Handle call_limit for SIP devices Setting a call-limit will cause calls above the limit not to be accepted.
static void	update_peer (struct sip_peer *p, int expire)
	Update peer data in database (if used).
Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_DEFAULT , .description = "Session Initiation Protocol (SIP)" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "a9c98e5d177805051735cb5b0b16b0a0" , .load = load_module, .unload = unload_module, .reload = reload, }
static char *	app_dtmfmode = "SIPDtmfMode"
static char *	app_sipaddheader = "SIPAddHeader"
static struct ast_module_info *	ast_module_info = &__mod_info
static struct sip_auth *	authl = NULL
	Authentication list for realm authentication.
static struct sockaddr_in	bindaddr
static struct ast_threadstorage	check_auth_buf = { .once = PTHREAD_ONCE_INIT, .key_init = __init_check_auth_buf , .custom_init = NULL , }
static struct ast_custom_function	checksipdomain_function
static struct ast_cli_entry	cli_sip []
	SIP Cli commands definition.
static struct ast_cli_entry	cli_sip_do_history_deprecated = { .handler = sip_do_history_deprecated , .summary = "Enable/Disable SIP history" ,__VA_ARGS__ }
static const char	config [] = "sip.conf"
static struct sockaddr_in	debugaddr
static int	default_expiry = DEFAULT_DEFAULT_EXPIRY
static struct ast_jb_conf	default_jbconf
	Global jitterbuffer configuration - by default, jb is disabled.
static struct ast_tls_config	default_tls_cfg
	Default TLS connection configuration.
static char *	descrip_dtmfmode = " SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call\n"
static char *	descrip_sipaddheader
ao2_container *	dialogs
	Here we implement the container for dialogs (sip_pvt), defining generic wrapper functions to ease the transition from the current implementation (a single linked list) to a different container. In addition to a reference to the container, we need functions to lock/unlock the container and individual items, and functions to add/remove references to the individual items.
static struct _map_x_s	dtmfstr []
	mapping between dtmf flags and strings
static time_t	externexpire
static char	externhost [MAXHOSTNAMELEN]
static struct sockaddr_in	externip
	our external IP address/port for SIP sessions. externip.sin_addr is only set when we know we might be behind a NAT, and this is done using a variety of (mutually exclusive) ways from the config file:
static int	externrefresh = 10
static struct ast_ha *	global_contact_ha = NULL
	Global list of addresses dynamic peers are not allowed to use.
static int	global_dynamic_exclude_static = 0
static struct ast_jb_conf	global_jbconf
static int	hash_dialog_size = 563
static int	hash_peer_size = 563
static int	hash_user_size = 563
static struct _map_x_s	insecurestr []
static struct sockaddr_in	internip
	our (internal) default address/port to put in SIP/SDP messages internip is initialized picking a suitable address from one of the interfaces, and the same port number we bind to. It is used as the default address/port in SIP messages, and as the default address (but not port) in SDP messages.
static struct invstate2stringtable	invitestate2string []
	Readable descriptions of device states.
static struct ast_ha *	localaddr
	List of local networks We store "localnet" addresses from the config file into an access list, marked as 'DENY', so the call to ast_apply_ha() will return AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local' (i.e. presumably public) addresses.
static char	mandescr_show_peer []
static char	mandescr_show_peers []
static char	mandescr_show_registry []
	Manager Action SIPShowRegistry description.
static char	mandescr_sipnotify []
static int	max_expiry = DEFAULT_MAX_EXPIRY
static int	min_expiry = DEFAULT_MIN_EXPIRY
static struct _map_x_s	natmodes []
static const char	notify_config [] = "sip_notify.conf"
static struct ast_config *	notify_types
static int	ourport_tcp
static int	ourport_tls
ao2_container *	peers
	The peer list: Users, Peers and Friends.
ao2_container *	peers_by_ip
static struct _map_x_s	referstatusstrings []
static struct ast_register_list	regl
	The register list: Other SIP proxies we register with and place calls to.
static struct _map_x_s	regstatestrings []
static struct ast_custom_function	sip_header_function
static struct cfsip_methods	sip_methods []
	The core structure to setup dialogs. We parse incoming messages by using structure and then route the messages according to the type.
static struct cfsip_options	sip_options []
	List of well-known SIP options. If we get this in a require, we should check the list and answer accordingly.
static struct ast_rtp_protocol	sip_rtp
	Interface structure with callbacks used to connect to RTP module.
static struct ast_tcptls_session_args	sip_tcp_desc
	The TCP server definition.
static struct ast_channel_tech	sip_tech
	Definition of this channel for PBX channel registration.
static struct ast_channel_tech	sip_tech_info
	This version of the sip channel tech has no send_digit_begin callback so that the core knows that the channel does not want DTMF BEGIN frames. The struct is initialized just before registering the channel driver, and is for use with channels using SIP INFO DTMF.
static struct ast_tls_config	sip_tls_cfg
	Working TLS connection configuration.
static struct ast_tcptls_session_args	sip_tls_desc
	The TCP/TLS server definition.
static struct ast_udptl_protocol	sip_udptl
	Interface structure with callbacks used to connect to UDPTL module.
static struct ast_custom_function	sipchaninfo_function
	Structure to declare a dialplan function: SIPCHANINFO.
static enum sip_debug_e	sipdebug
static int	sipdebug_text
	extra debugging for 'text' related events. At the moment this is set together with sip_debug_console.
ast_custom_function	sippeer_function
	Structure to declare a dialplan function: SIPPEER.
static int	sipsock = -1
	Main socket for SIP communication.
static struct _map_x_s	stmodes []
	Report Peer status in character string.
static struct _map_x_s	strefreshers []
static struct sockaddr_in	stunaddr
static struct cfsubscription_types	subscription_types []
	Subscription types that we support. We support dialoginfo updates (really device status, not dialog info as was the original intent of the standard) SIMPLE presence used for device status Voicemail notification subscriptions.
static char *	synopsis_dtmfmode = "Change the dtmfmode for a SIP call"
static char *	synopsis_sipaddheader = "Add a SIP header to the outbound call"
static struct ast_threadstorage	ts_temp_pvt = { .once = PTHREAD_ONCE_INIT, .key_init = __init_ts_temp_pvt , .custom_init = temp_pvt_init , }

---

Detailed Description

Implementation of Session Initiation Protocol.

Author:

Mark Spencer <markster@digium.com>

See Also:

• CREDITS

Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support Configuration file sip.conf

********** IMPORTANT *

Note:

TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration settings, dialplan commands and dialplans apps/functions

TODO:s

Todo:

Better support of forking

VIA branch tag transaction checking

Transaction support

We need to test TCP sessions with SIP proxies and in regards to the SIP outbound specs.

Fix TCP/TLS handling in dialplan, SRV records, transfers and much more

Save TCP/TLS sessions in registry

Add TCP/TLS information to function SIPPEER and SIPCHANINFO

Overview of the handling of SIP sessions

The SIP channel handles several types of SIP sessions, or dialogs, not all of them being "telephone calls".

• Incoming calls that will be sent to the PBX core
• Outgoing calls, generated by the PBX
• SIP subscriptions and notifications of states and voicemail messages
• SIP registrations, both inbound and outbound
• SIP peer management (peerpoke, OPTIONS)
• SIP text messages

In the SIP channel, there's a list of active SIP dialogs, which includes all of these when they are active. "sip show channels" in the CLI will show most of these, excluding subscriptions which are shown by "sip show subscriptions"

incoming packets

Incoming packets are received in the monitoring thread, then handled by sipsock_read(). This function parses the packet and matches an existing dialog or starts a new SIP dialog.

sipsock_read sends the packet to handle_incoming(), that parses a bit more. If it is a response to an outbound request, the packet is sent to handle_response(). If it is a request, handle_incoming() sends it to one of a list of functions depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc sipsock_read locks the ast_channel if it exists (an active call) and unlocks it after we have processed the SIP message.

A new INVITE is sent to handle_request_invite(), that will end up starting a new channel in the PBX, the new channel after that executing in a separate channel thread. This is an incoming "call". When the call is answered, either by a bridged channel or the PBX itself the sip_answer() function is called.

The actual media - Video or Audio - is mostly handled by the RTP subsystem in rtp.c

Outbound calls

Outbound calls are set up by the PBX through the sip_request_call() function. After that, they are activated by sip_call().

Hanging up

The PBX issues a hangup on both incoming and outgoing calls through the sip_hangup() function

Definition in file chan_sip.c.

---

Define Documentation

#define ALLOWED_METHODS   "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"

SIP Methods we support.

Todo:

This string should be set dynamically. We only support REFER and SUBSCRIBE is we have allowsubscribe and allowrefer on in sip.conf.

Definition at line 654 of file chan_sip.c.

Referenced by respprep(), transmit_invite(), transmit_notify_with_sipfrag(), transmit_refer(), and transmit_reinvite_with_sdp().

#define append_history	(	p,
		event,
		fmt,
		args...		)	append_history_full(p, "%-15s " fmt, event, ## args)

Append to SIP dialog history.

Returns:

Always returns 0

Definition at line 2096 of file chan_sip.c.

Referenced by __sip_autodestruct(), __sip_reliable_xmit(), auto_congest(), build_reply_digest(), cb_extensionstate(), check_auth(), do_register_auth(), handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_do(), handle_request_invite(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), handle_response_invite(), local_attended_transfer(), obproxy_get(), process_sdp(), retrans_pkt(), send_request(), send_response(), sip_cancel_destroy(), sip_fixup(), sip_hangup(), sip_new(), sip_park_thread(), sip_reregister(), sip_scheddestroy(), sip_set_rtp_peer(), transmit_register(), transmit_reinvite_with_sdp(), and transmit_response_with_auth().

#define CALLERID_UNKNOWN   "Unknown"

Definition at line 250 of file chan_sip.c.

Referenced by initreqprep().

#define CHECK_AUTH_BUF_INITLEN   256

Definition at line 10883 of file chan_sip.c.

Referenced by check_auth(), and transmit_fake_auth_response().

#define check_request_transport	(	peer,
		tmpl		)

generic function for determining if a correct transport is being used to contact a peer

this is done as a macro so that the "tmpl" var can be passed either a sip_request or a sip_peer

Definition at line 2003 of file chan_sip.c.

Referenced by create_addr_from_peer(), and register_verify().

#define DEC_CALL_LIMIT   0

Definition at line 850 of file chan_sip.c.

Referenced by __sip_destroy(), handle_request_cancel(), handle_request_invite(), handle_response_invite(), sip_hangup(), and update_call_counter().

#define DEC_CALL_RINGING   2

Definition at line 852 of file chan_sip.c.

Referenced by handle_response_invite(), and update_call_counter().

#define DEFAULT_ALLOW_EXT_DOM   TRUE

Allow external domains

Definition at line 702 of file chan_sip.c.

#define DEFAULT_ALLOWGUEST   TRUE

Definition at line 690 of file chan_sip.c.

#define DEFAULT_AUTOCREATEPEER   FALSE

Definition at line 706 of file chan_sip.c.

#define DEFAULT_CALLCOUNTER   FALSE

Definition at line 691 of file chan_sip.c.

#define DEFAULT_CALLERID   "asterisk"

Definition at line 688 of file chan_sip.c.

#define DEFAULT_COMPACTHEADERS   FALSE

Definition at line 693 of file chan_sip.c.

#define DEFAULT_CONTEXT   "default"

Definition at line 684 of file chan_sip.c.

#define DEFAULT_COS_AUDIO   5

Level 2 class of service for audio media

Definition at line 699 of file chan_sip.c.

#define DEFAULT_COS_SIP   4

Level 2 class of service for SIP signalling

Definition at line 698 of file chan_sip.c.

#define DEFAULT_COS_TEXT   5

Level 2 class of service for text media (T.140)

Definition at line 701 of file chan_sip.c.

#define DEFAULT_COS_VIDEO   6

Level 2 class of service for video media

Definition at line 700 of file chan_sip.c.

#define DEFAULT_DEFAULT_EXPIRY   120

Definition at line 223 of file chan_sip.c.

#define DEFAULT_EXPIRY   900

Expire slowly

Definition at line 240 of file chan_sip.c.

#define DEFAULT_FREQ_NOTOK   10 * 1000

Qualification: How often to check, if the host is down...

Definition at line 254 of file chan_sip.c.

#define DEFAULT_MAX_CALL_BITRATE   (384)

Max bitrate for video

Definition at line 710 of file chan_sip.c.

#define DEFAULT_MAX_EXPIRY   3600

Definition at line 225 of file chan_sip.c.

#define DEFAULT_MAX_FORWARDS   "70"

Definition at line 227 of file chan_sip.c.

Referenced by initreqprep(), reqprep(), and transmit_register().

#define DEFAULT_MAX_SE   1800

Session-Timer Default Session-Expires period (RFC 4028)

Definition at line 272 of file chan_sip.c.

#define DEFAULT_MAXMS   2000

Qualification: Must be faster than 2 seconds by default

Definition at line 252 of file chan_sip.c.

#define DEFAULT_MIN_EXPIRY   60

Definition at line 224 of file chan_sip.c.

#define DEFAULT_MIN_SE   90

Session-Timer Default Min-SE period (RFC 4028)

Definition at line 273 of file chan_sip.c.

#define DEFAULT_MOHINTERPRET   "default"

Definition at line 685 of file chan_sip.c.

#define DEFAULT_MOHSUGGEST   ""

Definition at line 686 of file chan_sip.c.

#define DEFAULT_NOTIFYMIME   "application/simple-message-summary"

Definition at line 689 of file chan_sip.c.

#define DEFAULT_NOTIFYRINGING   TRUE

Definition at line 704 of file chan_sip.c.

#define DEFAULT_PEDANTIC   FALSE

Definition at line 705 of file chan_sip.c.

#define DEFAULT_QUALIFY   FALSE

Definition at line 707 of file chan_sip.c.

#define DEFAULT_QUALIFYFREQ   60 * 1000

Qualification: How often to check for the host to be up

Definition at line 253 of file chan_sip.c.

#define DEFAULT_REALM   "asterisk"

Realm for HTTP digest authentication

Definition at line 703 of file chan_sip.c.

#define DEFAULT_REGEXTENONQUALIFY   FALSE

Definition at line 708 of file chan_sip.c.

#define DEFAULT_REGISTRATION_TIMEOUT   20

Definition at line 226 of file chan_sip.c.

#define DEFAULT_RETRANS   1000

How frequently to retransmit Default: 2 * 500 ms in RFC 3261

Definition at line 256 of file chan_sip.c.

#define DEFAULT_SDPOWNER   "root"

Default SDP username field in (o=) header unless re-defined in sip.conf

Definition at line 714 of file chan_sip.c.

#define DEFAULT_SDPSESSION   "Asterisk PBX"

Default SDP session name, (s=) header unless re-defined in sip.conf

Definition at line 713 of file chan_sip.c.

#define DEFAULT_SRVLOOKUP   TRUE

Recommended setting is ON

Definition at line 692 of file chan_sip.c.

#define DEFAULT_T1MIN   100

100 MS for minimal roundtrip time

Definition at line 709 of file chan_sip.c.

#define DEFAULT_TOS_AUDIO   0

Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions.

Definition at line 695 of file chan_sip.c.

#define DEFAULT_TOS_SIP   0

Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions.

Definition at line 694 of file chan_sip.c.

#define DEFAULT_TOS_TEXT   0

Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions.

Definition at line 697 of file chan_sip.c.

#define DEFAULT_TOS_VIDEO   0

Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions.

Definition at line 696 of file chan_sip.c.

#define DEFAULT_TRANS_TIMEOUT   -1

Definition at line 262 of file chan_sip.c.

Referenced by __sip_autodestruct(), cb_extensionstate(), check_auth(), check_pendings(), handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_notify(), handle_request_options(), handle_request_register(), handle_response_invite(), handle_response_register(), receive_message(), sip_hangup(), sip_send_mwi_to_peer(), and transmit_fake_auth_response().

#define DEFAULT_USERAGENT   "Asterisk PBX"

Default Useragent: header unless re-defined in sip.conf

Definition at line 712 of file chan_sip.c.

#define DEFAULT_VMEXTEN   "asterisk"

Definition at line 687 of file chan_sip.c.

#define EXPIRY_GUARD_LIMIT   30

Below here, we use EXPIRY_GUARD_PCT instead of EXPIRY_GUARD_SECS

Definition at line 232 of file chan_sip.c.

Referenced by handle_response_register().

#define EXPIRY_GUARD_MIN   500

This is the minimum guard time applied. If GUARD_PCT turns out to be lower than this, it will use this time instead. This is in milliseconds.

Definition at line 234 of file chan_sip.c.

Referenced by handle_response_register().

#define EXPIRY_GUARD_PCT   0.20

Percentage of expires timeout to use when below EXPIRY_GUARD_LIMIT

Definition at line 238 of file chan_sip.c.

Referenced by handle_response_register().

#define EXPIRY_GUARD_SECS   15

How long before expiry do we reregister

Definition at line 231 of file chan_sip.c.

Referenced by handle_response_register().

#define FALSE   0

Definition at line 203 of file chan_sip.c.

#define FINDALLDEVICES   (FINDUSERS | FINDPEERS)

Definition at line 215 of file chan_sip.c.

Referenced by check_peer_ok(), find_by_name(), sip_devicestate(), sip_peer_hold(), and update_call_counter().

#define FINDPEERS   (1 << 1)

Definition at line 214 of file chan_sip.c.

Referenced by _sip_qualify_peer(), _sip_show_peer(), check_peer_ok(), create_addr(), find_by_name(), function_sippeer(), register_verify(), sip_do_debug_peer(), sip_unregister(), st_get_mode(), st_get_refresher(), and st_get_se().

#define FINDUSERS   (1 << 0)

Definition at line 213 of file chan_sip.c.

Referenced by check_peer_ok(), find_by_name(), and sip_show_user().

#define FORMAT   "%-15.15s %-10.10s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s\n"

Definition at line 14408 of file chan_sip.c.

#define FORMAT   "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"

Definition at line 14408 of file chan_sip.c.

#define FORMAT   "%-30.30s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"

Definition at line 14408 of file chan_sip.c.

#define FORMAT   "%-40.40s %-20.20s %-16.16s\n"

Definition at line 14408 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"

Definition at line 14408 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"

Definition at line 14408 of file chan_sip.c.

#define FORMAT   "%-30.30s %-6d %-9.9s %-6.6s\n"

Definition at line 14408 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-15.15s \n"

Definition at line 14408 of file chan_sip.c.

#define FORMAT2   "%-15.15s %-10.10s %-15.15s %-15.15s %-7.7s %-15.15s %-6.6s\n"

Definition at line 14407 of file chan_sip.c.

#define FORMAT2   "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s (%-2.2s) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"

Definition at line 14407 of file chan_sip.c.

#define FORMAT2   "%-30.30s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"

Definition at line 14407 of file chan_sip.c.

#define FORMAT2   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %s\n"

Definition at line 14407 of file chan_sip.c.

#define FORMAT2   "%-30.30s %3.6s %9.9s %6.6s\n"

Definition at line 14407 of file chan_sip.c.

#define FORMAT2   "%-25.25s %-15.15s %-15.15s \n"

Definition at line 14407 of file chan_sip.c.

#define FORMAT3   "%-15.15s %-10.10s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"

Definition at line 14406 of file chan_sip.c.

Referenced by sip_show_channels().

#define FORMAT4   "%-15.15s %-10.10s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"

Definition at line 14405 of file chan_sip.c.

Referenced by show_channels_cb().

#define INC_CALL_LIMIT   1

Definition at line 851 of file chan_sip.c.

Referenced by handle_request_invite(), and update_call_counter().

#define INC_CALL_RINGING   3

Definition at line 853 of file chan_sip.c.

Referenced by sip_call(), and update_call_counter().

#define INITIAL_CSEQ   101

our initial sip sequence number

Definition at line 270 of file chan_sip.c.

Referenced by sip_alloc(), sip_register(), and transmit_response_using_temp().

#define IS_SIP_TECH

(

t

)

((t) == &sip_tech || (t) == &sip_tech_info)

Definition at line 2319 of file chan_sip.c.

Referenced by acf_channel_read(), func_header_read(), function_sipchaninfo_read(), handle_request_bye(), sip_dtmfmode(), and sip_hangup().

#define MAX	(	a,
		b		)	((a) > (b) ? (a) : (b))

Definition at line 247 of file chan_sip.c.

#define MAX_AUTHTRIES   3

Try authentication three times, then fail

Definition at line 263 of file chan_sip.c.

Referenced by handle_response(), handle_response_invite(), and handle_response_register().

#define MAX_HISTORY_ENTRIES   50

Max entires in the history list for a sip_pvt

Definition at line 1409 of file chan_sip.c.

Referenced by append_history_va().

#define MAX_RETRANS   6

Try only 6 times for retransmissions, a total of 7 transmissions

Definition at line 257 of file chan_sip.c.

#define NO_RTP   0

Definition at line 291 of file chan_sip.c.

#define NOT_SUPPORTED   0

Definition at line 572 of file chan_sip.c.

#define REQ_OFFSET_TO_STR	(	req,
		offset		)	((req)->data->str + ((req)->offset))

Definition at line 916 of file chan_sip.c.

Referenced by __get_header(), _sip_tcp_helper_thread(), find_sdp(), get_body(), get_destination(), get_msg_text(), get_sdp_iterate(), handle_incoming(), handle_request_do(), handle_request_invite(), reqprep(), and send_response().

#define RTP   1

Definition at line 290 of file chan_sip.c.

#define SDP_MAX_RTPMAP_CODECS   32

Maximum number of codecs allowed in received SDP

Definition at line 275 of file chan_sip.c.

Referenced by process_sdp().

#define SDP_SAMPLE_RATE

(

x

)

8000

Note:

G.722 actually is supposed to specified as 8 kHz, even though it is really 16 kHz. Update this macro for other formats as they are added in the future.

Definition at line 8596 of file chan_sip.c.

Referenced by add_sdp().

#define SIP_CALL_LIMIT   (1 << 7)

D: Call limit enforced for this call

Definition at line 999 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), and update_call_counter().

#define SIP_CAN_REINVITE   (1 << 20)

DP: allow peers to be reinvited to send media directly p2p

Definition at line 1027 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), and sip_get_vrtp_peer().

#define SIP_CAN_REINVITE_NAT   (2 << 20)

DP: allow media reinvite when new peer is behind NAT

Definition at line 1028 of file chan_sip.c.

Referenced by handle_common_options(), sip_get_rtp_peer(), and sip_set_rtp_peer().

#define SIP_DEFER_BYE_ON_TRANSFER   (1 << 10)

D: Do not hangup at first ast_hangup

Definition at line 1002 of file chan_sip.c.

Referenced by handle_invite_replaces(), handle_request_refer(), local_attended_transfer(), sip_hangup(), and sip_set_rtp_peer().

#define SIP_DTMF   (7 << 15)

DP: DTMF Support: five settings, uses three bits

Definition at line 1010 of file chan_sip.c.

Referenced by _sip_show_peer(), check_peer_ok(), create_addr_from_peer(), handle_common_options(), handle_request_invite(), process_sdp(), sip_alloc(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), sip_senddigit_end(), sip_show_channel(), sip_show_settings(), and transmit_info_with_digit().

#define SIP_DTMF_AUTO   (3 << 15)

DP: DTMF Support: AUTO switch between rfc2833 and in-band DTMF

Definition at line 1014 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), handle_common_options(), process_sdp(), sip_alloc(), and sip_new().

#define SIP_DTMF_INBAND   (1 << 15)

DP: DTMF Support: Inband audio, only for ULAW/ALAW - "inband"

Definition at line 1012 of file chan_sip.c.

Referenced by handle_common_options(), process_sdp(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), and sip_senddigit_end().

#define SIP_DTMF_INFO   (2 << 15)

DP: DTMF Support: SIP Info messages - "info"

Definition at line 1013 of file chan_sip.c.

Referenced by handle_common_options(), sip_dtmfmode(), sip_new(), and sip_senddigit_end().

#define SIP_DTMF_RFC2833   (0 << 15)

DP: DTMF Support: RTP DTMF - "rfc2833"

Definition at line 1011 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), handle_common_options(), handle_request_invite(), process_sdp(), sip_alloc(), sip_dtmfmode(), sip_rtp_read(), sip_senddigit_begin(), and sip_senddigit_end().

#define SIP_DTMF_SHORTINFO   (4 << 15)

DP: DTMF Support: SIP Info messages - "info" - short variant

Definition at line 1015 of file chan_sip.c.

Referenced by handle_common_options(), sip_dtmfmode(), sip_new(), sip_senddigit_end(), and transmit_info_with_digit().

#define SIP_FLAGS_TO_COPY

Value:

(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
    SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
    SIP_USEREQPHONE | SIP_INSECURE)

Flags to copy from peer/user to dialog.

Definition at line 1047 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), set_peer_defaults(), sip_alloc(), and sip_poke_peer().

#define SIP_G726_NONSTANDARD   (1 << 31)

DP: Use non-standard packing for G726-32 data

Definition at line 1044 of file chan_sip.c.

Referenced by add_codec_to_sdp(), handle_common_options(), and process_sdp().

#define SIP_GOTREFER   (1 << 6)

D: Got a refer?

Definition at line 998 of file chan_sip.c.

Referenced by handle_request_refer(), local_attended_transfer(), sip_read(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_INC_COUNT   (1 << 8)

D: Did this dialog increment the counter of in-use calls?

Definition at line 1000 of file chan_sip.c.

Referenced by __sip_destroy(), handle_request_cancel(), sip_hangup(), and update_call_counter().

#define SIP_INC_RINGING   (1 << 9)

D: Did this connection increment the counter of in-use calls?

Definition at line 1001 of file chan_sip.c.

Referenced by update_call_counter().

#define SIP_INSECURE   (3 << 23)

DP: three settings, uses two bits

Definition at line 1032 of file chan_sip.c.

Referenced by _sip_show_peer(), and handle_common_options().

#define SIP_INSECURE_INVITE   (1 << 24)

DP: don't require authentication for incoming INVITEs

Definition at line 1035 of file chan_sip.c.

Referenced by check_peer_ok(), and set_insecure_flags().

#define SIP_INSECURE_NONE   (0 << 23)

DP: secure mode

Definition at line 1033 of file chan_sip.c.

#define SIP_INSECURE_PORT   (1 << 23)

DP: don't require matching port for incoming requests

Definition at line 1034 of file chan_sip.c.

Referenced by find_peer(), get_insecure_variable_from_config(), peer_ipcmp_cb(), peer_iphash_cb(), and set_insecure_flags().

#define SIP_MAX_HEADERS   64

Max amount of SIP headers to read

Definition at line 265 of file chan_sip.c.

Referenced by add_header(), and parse_request().

#define SIP_MAX_LINES   64

Max amount of lines in SIP attachment (like SDP)

Definition at line 266 of file chan_sip.c.

Referenced by add_line(), and parse_request().

#define SIP_MAX_PACKET   4096

Also from RFC 3261 (2543), should sub headers tho

Definition at line 267 of file chan_sip.c.

#define SIP_MIN_PACKET   1024

Initialize size of memory to allocate for packets

Definition at line 268 of file chan_sip.c.

Referenced by _sip_tcp_helper_thread(), init_req(), init_resp(), and sipsock_read().

#define SIP_NAT   (3 << 18)

DP: four settings, uses two bits

Definition at line 1018 of file chan_sip.c.

Referenced by _sip_show_peer(), build_via(), copy_via_headers(), create_addr(), create_addr_from_peer(), handle_common_options(), process_sdp(), register_verify(), sip_alloc(), sip_nat_mode(), sip_real_dst(), sip_show_channel(), sip_show_settings(), sip_show_users(), and transmit_response_using_temp().

#define SIP_NAT_ALWAYS   (3 << 18)

DP: NAT Both ROUTE and RFC3581

Definition at line 1022 of file chan_sip.c.

Referenced by copy_via_headers(), and handle_common_options().

#define SIP_NAT_NEVER   (0 << 18)

DP: No nat support

Definition at line 1019 of file chan_sip.c.

Referenced by handle_common_options().

#define SIP_NAT_RFC3581   (1 << 18)

DP: NAT RFC3581

Definition at line 1020 of file chan_sip.c.

Referenced by build_via(), copy_via_headers(), and handle_common_options().

#define SIP_NAT_ROUTE   (2 << 18)

DP: NAT Only ROUTE

Definition at line 1021 of file chan_sip.c.

Referenced by _sip_show_peers(), check_peer_ok(), check_via(), create_addr(), create_addr_from_peer(), handle_common_options(), parse_register_contact(), send_request(), set_address_from_contact(), sip_alloc(), sip_nat_mode(), sip_real_dst(), and transmit_response_using_temp().

#define SIP_NEEDREINVITE   (1 << 4)

D: Do we need to send another reinvite?

Definition at line 996 of file chan_sip.c.

Referenced by check_pendings(), sip_hangup(), sip_indicate(), sip_read(), sip_reinvite_retry(), sip_sendhtml(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_OPT_100REL   (1 << 1)

Definition at line 576 of file chan_sip.c.

#define SIP_OPT_EARLY_SESSION   (1 << 3)

Definition at line 578 of file chan_sip.c.

#define SIP_OPT_EVENTLIST   (1 << 11)

Definition at line 586 of file chan_sip.c.

#define SIP_OPT_FROMCHANGE   (1 << 17)

Definition at line 592 of file chan_sip.c.

#define SIP_OPT_GRUU   (1 << 12)

Definition at line 587 of file chan_sip.c.

#define SIP_OPT_HISTINFO   (1 << 15)

Definition at line 590 of file chan_sip.c.

#define SIP_OPT_JOIN   (1 << 4)

Definition at line 579 of file chan_sip.c.

#define SIP_OPT_NOREFERSUB   (1 << 14)

Definition at line 589 of file chan_sip.c.

#define SIP_OPT_PATH   (1 << 5)

Definition at line 580 of file chan_sip.c.

#define SIP_OPT_PRECONDITION   (1 << 7)

Definition at line 582 of file chan_sip.c.

#define SIP_OPT_PREF   (1 << 6)

Definition at line 581 of file chan_sip.c.

#define SIP_OPT_PRIVACY   (1 << 8)

Definition at line 583 of file chan_sip.c.

#define SIP_OPT_RECLISTINV   (1 << 18)

Definition at line 593 of file chan_sip.c.

#define SIP_OPT_RECLISTSUB   (1 << 19)

Definition at line 594 of file chan_sip.c.

#define SIP_OPT_REPLACES   (1 << 0)

Definition at line 575 of file chan_sip.c.

Referenced by handle_request_invite().

#define SIP_OPT_RESPRIORITY   (1 << 16)

Definition at line 591 of file chan_sip.c.

#define SIP_OPT_SDP_ANAT   (1 << 9)

Definition at line 584 of file chan_sip.c.

#define SIP_OPT_SEC_AGREE   (1 << 10)

Definition at line 585 of file chan_sip.c.

#define SIP_OPT_TARGET_DIALOG   (1 << 13)

Definition at line 588 of file chan_sip.c.

#define SIP_OPT_TIMER   (1 << 2)

Definition at line 577 of file chan_sip.c.

Referenced by handle_request_invite().

#define SIP_OPT_UNKNOWN   (1 << 20)

Definition at line 595 of file chan_sip.c.

Referenced by parse_sip_options().

#define SIP_OUTGOING   (1 << 0)

D: Direction of the last transaction in this dialog

Definition at line 993 of file chan_sip.c.

Referenced by handle_request_bye(), handle_request_invite(), handle_request_refer(), handle_response(), handle_response_invite(), manager_sipnotify(), reqprep(), respprep(), sip_call(), sip_cli_notify(), sip_hangup(), sip_indicate(), sip_poke_peer(), sip_send_mwi_to_peer(), sip_show_channel(), sip_write(), transmit_refer(), transmit_register(), transmit_reinvite_with_sdp(), and transmit_response_with_sdp().

#define SIP_PAGE2_ALLOWOVERLAP   (1 << 17)

DP: Allow overlap dialing ?

Definition at line 1065 of file chan_sip.c.

Referenced by _sip_show_peer(), get_destination(), handle_common_options(), handle_request_invite(), and sip_show_settings().

#define SIP_PAGE2_ALLOWSUBSCRIBE   (1 << 16)

GP: Allow subscriptions from this peer?

Definition at line 1064 of file chan_sip.c.

Referenced by _sip_show_peer(), build_peer(), handle_common_options(), handle_request_subscribe(), and sip_show_settings().

#define SIP_PAGE2_BUGGY_MWI   (1 << 26)

DP: Buggy CISCO MWI fix

Definition at line 1079 of file chan_sip.c.

Referenced by handle_common_options(), and transmit_notify_with_mwi().

#define SIP_PAGE2_CALL_ONHOLD   (3 << 23)

D: Call hold states:

Definition at line 1073 of file chan_sip.c.

Referenced by __sip_destroy(), add_sdp(), check_rtp_timeout(), handle_request_cancel(), process_sdp(), show_channels_cb(), sip_hangup(), and update_call_counter().

#define SIP_PAGE2_CALL_ONHOLD_ACTIVE   (1 << 23)

D: Active hold

Definition at line 1074 of file chan_sip.c.

Referenced by process_sdp().

#define SIP_PAGE2_CALL_ONHOLD_INACTIVE   (3 << 23)

D: Inactive hold

Definition at line 1076 of file chan_sip.c.

Referenced by add_sdp(), and process_sdp().

#define SIP_PAGE2_CALL_ONHOLD_ONEDIR   (2 << 23)

D: One directional hold

Definition at line 1075 of file chan_sip.c.

Referenced by add_sdp(), and process_sdp().

#define SIP_PAGE2_DIALOG_ESTABLISHED   (1 << 27)

29: Has a dialog been established?

Definition at line 1080 of file chan_sip.c.

Referenced by find_call_cb(), handle_request_bye(), handle_request_invite(), handle_request_subscribe(), handle_response(), handle_response_invite(), and sip_answer().

#define SIP_PAGE2_FLAGS_TO_COPY

Value:

(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
   SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
   SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION | \
   SIP_PAGE2_VIDEOSUPPORT_ALWAYS)

Definition at line 1085 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), set_peer_defaults(), sip_alloc(), and sip_poke_peer().

#define SIP_PAGE2_REGISTERTRYING   (1 << 29)

DP: Send 100 Trying on REGISTER attempts

Definition at line 1081 of file chan_sip.c.

Referenced by _sip_show_peer(), build_peer(), and register_verify().

#define SIP_PAGE2_RFC2833_COMPENSATE   (1 << 25)

DP: Compensate for buggy RFC2833 implementations

Definition at line 1078 of file chan_sip.c.

Referenced by create_addr_from_peer(), handle_common_options(), handle_request_invite(), process_sdp(), sip_alloc(), and sip_show_settings().

#define SIP_PAGE2_RTAUTOCLEAR   (1 << 2)

GP: Should we clean memory from peers after expiry?

Definition at line 1058 of file chan_sip.c.

Referenced by expire_register().

#define SIP_PAGE2_RTCACHEFRIENDS   (1 << 0)

GP: Should we keep RT objects in memory for extended time?

Definition at line 1057 of file chan_sip.c.

Referenced by build_peer(), parse_register_contact(), sip_prune_realtime(), sip_show_settings(), and update_peer().

#define SIP_PAGE2_STATECHANGEQUEUE   (1 << 9)

D: Unsent state pending change exists

Definition at line 1060 of file chan_sip.c.

Referenced by cb_extensionstate(), handle_response(), and handle_response_notify().

#define SIP_PAGE2_SUBSCRIBEMWIONLY   (1 << 18)

GP: Only issue MWI notification if subscribed to

Definition at line 1066 of file chan_sip.c.

Referenced by build_peer(), handle_request_subscribe(), register_verify(), and sip_send_mwi_to_peer().

#define SIP_PAGE2_T38SUPPORT   (7 << 20)

GDP: T38 Fax Passthrough Support

Definition at line 1068 of file chan_sip.c.

Referenced by create_addr_from_peer(), sip_alloc(), sip_indicate(), sip_queryoption(), and sip_write().

#define SIP_PAGE2_T38SUPPORT_RTP   (2 << 20)

GDP: T38 Fax Passthrough Support (not implemented)

Definition at line 1070 of file chan_sip.c.

Referenced by _sip_show_peer(), add_sdp(), handle_common_options(), and sip_show_settings().

#define SIP_PAGE2_T38SUPPORT_TCP   (4 << 20)

GDP: T38 Fax Passthrough Support (not implemented)

Definition at line 1071 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), and sip_show_settings().

#define SIP_PAGE2_T38SUPPORT_UDPTL   (1 << 20)

GDP: T38 Fax Passthrough Support

Definition at line 1069 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), sip_read(), sip_rtp_read(), and sip_show_settings().

#define SIP_PAGE2_TEXTSUPPORT   (1 << 15)

GDP: Global text enable

Definition at line 1063 of file chan_sip.c.

Referenced by _sip_show_peer(), _sip_show_peers(), check_peer_ok(), create_addr_from_peer(), handle_common_options(), sip_alloc(), and sip_show_settings().

#define SIP_PAGE2_UDPTL_DESTINATION   (1 << 30)

DP: Use source IP of RTP as destination if NAT is enabled

Definition at line 1082 of file chan_sip.c.

Referenced by handle_common_options(), and process_sdp().

#define SIP_PAGE2_VIDEOSUPPORT   (1 << 14)

DP: Video supported if offered?

Definition at line 1062 of file chan_sip.c.

Referenced by _sip_show_peer(), _sip_show_peers(), check_peer_ok(), create_addr_from_peer(), handle_common_options(), sip_alloc(), sip_new(), and sip_show_settings().

#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS   (1 << 31)

DP: Always set up video, even if endpoints don't support it

Definition at line 1083 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), and handle_common_options().

#define SIP_PENDINGBYE   (1 << 5)

D: Need to send bye after we ack?

Definition at line 997 of file chan_sip.c.

Referenced by check_pendings(), handle_response_invite(), register_verify(), sip_hangup(), sip_indicate(), sip_read(), sip_sendhtml(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_PROG_INBAND   (3 << 25)

DP: three settings, uses two bits

Definition at line 1038 of file chan_sip.c.

Referenced by handle_common_options(), sip_indicate(), and sip_show_settings().

#define SIP_PROG_INBAND_NEVER   (0 << 25)

Definition at line 1039 of file chan_sip.c.

Referenced by sip_indicate(), and sip_show_settings().

#define SIP_PROG_INBAND_NO   (1 << 25)

Definition at line 1040 of file chan_sip.c.

Referenced by handle_common_options(), and sip_show_settings().

#define SIP_PROG_INBAND_YES   (2 << 25)

Definition at line 1041 of file chan_sip.c.

Referenced by handle_common_options(), and sip_indicate().

#define SIP_PROGRESS_SENT   (1 << 3)

D: Have sent 183 message progress

Definition at line 995 of file chan_sip.c.

Referenced by sip_indicate(), and sip_write().

#define SIP_PROMISCREDIR   (1 << 11)

DP: Promiscuous redirection

Definition at line 1004 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), parse_moved_contact(), sip_show_channel(), and sip_show_settings().

#define sip_pvt_lock

(

x

)

ao2_lock(x)

Definition at line 1421 of file chan_sip.c.

Referenced by __sip_ack(), auto_congest(), cb_extensionstate(), check_rtp_timeout(), complete_sipch(), find_call(), get_sip_pvt_byid_locked(), handle_request_invite(), handle_request_subscribe(), proc_session_timer(), retrans_pkt(), scheduler_process_request_queue(), sip_answer(), sip_dtmfmode(), sip_fixup(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), sip_hangup(), sip_indicate(), sip_new(), sip_park(), sip_queryoption(), sip_read(), sip_reg_timeout(), sip_reinvite_retry(), sip_request_call(), sip_senddigit_begin(), sip_senddigit_end(), sip_set_rtp_peer(), sip_set_udptl_peer(), sip_show_channel(), sip_show_history(), sip_transfer(), sip_write(), and update_call_counter().

#define sip_pvt_trylock

(

x

)

ao2_trylock(x)

Definition at line 1422 of file chan_sip.c.

Referenced by dialog_needdestroy(), and find_call().

#define sip_pvt_unlock

(

x

)

ao2_unlock(x)

Definition at line 1423 of file chan_sip.c.

Referenced by __sip_ack(), auto_congest(), cb_extensionstate(), check_rtp_timeout(), complete_sipch(), dialog_needdestroy(), get_sip_pvt_byid_locked(), handle_invite_replaces(), handle_request_do(), handle_request_invite(), handle_request_subscribe(), local_attended_transfer(), proc_session_timer(), retrans_pkt(), scheduler_process_request_queue(), sip_answer(), sip_dtmfmode(), sip_fixup(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), sip_hangup(), sip_indicate(), sip_new(), sip_park(), sip_queryoption(), sip_read(), sip_reg_timeout(), sip_reinvite_retry(), sip_request_call(), sip_senddigit_begin(), sip_senddigit_end(), sip_set_rtp_peer(), sip_set_udptl_peer(), sip_show_channel(), sip_show_history(), sip_transfer(), sip_write(), and update_call_counter().

#define SIP_REINVITE   (7 << 20)

DP: four settings, uses three bits

Definition at line 1025 of file chan_sip.c.

Referenced by handle_common_options().

#define SIP_REINVITE_NONE   (0 << 20)

DP: no reinvite allowed

Definition at line 1026 of file chan_sip.c.

#define SIP_REINVITE_UPDATE   (4 << 20)

DP: use UPDATE (RFC3311) when reinviting this peer

Definition at line 1029 of file chan_sip.c.

Referenced by handle_common_options(), and transmit_reinvite_with_sdp().

#define SIP_RESERVED   ";/?:@&=+$,# "

Definition at line 219 of file chan_sip.c.

#define SIP_RINGING   (1 << 2)

D: Have sent 180 ringing

Definition at line 994 of file chan_sip.c.

Referenced by sip_indicate().

#define SIP_SENDRPID   (1 << 29)

DP: Remote Party-ID Support

Definition at line 1043 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), and initreqprep().

#define SIP_TIMER_T1   500

Definition at line 258 of file chan_sip.c.

#define SIP_TRANS_TIMEOUT   64 * SIP_TIMER_T1

SIP request timeout (rfc 3261) 64*T1

Todo:

Use known T1 for timeout (peerpoke)

Definition at line 259 of file chan_sip.c.

Referenced by manager_sipnotify(), sip_cli_notify(), and sip_sipredirect().

#define SIP_TRUSTRPID   (1 << 12)

DP: Trust RPID headers?

Definition at line 1005 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), and replace_cid().

#define SIP_USECLIENTCODE   (1 << 14)

DP: Trust X-ClientCode info message

Definition at line 1007 of file chan_sip.c.

Referenced by handle_common_options(), handle_request_info(), and sip_show_settings().

#define SIP_USEREQPHONE   (1 << 13)

DP: Add user=phone to numeric URI. Default off

Definition at line 1006 of file chan_sip.c.

Referenced by _sip_show_peer(), build_peer(), initreqprep(), and sip_show_settings().

#define SIPBUFSIZE   512

Definition at line 210 of file chan_sip.c.

Referenced by add_route(), add_sdp(), extract_uri(), handle_request_refer(), initreqprep(), parse_moved_contact(), parse_ok_contact(), parse_register_contact(), process_sdp(), respprep(), show_channels_cb(), sip_call(), sip_new(), sip_show_channel(), sip_show_settings(), transmit_invite(), and transmit_notify_with_sipfrag().

#define STANDARD_SIP_PORT   5060

Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS.

Definition at line 664 of file chan_sip.c.

Referenced by __set_address_from_contact(), build_peer(), check_via(), create_addr(), initreqprep(), manager_show_registry(), parse_register_contact(), proxy_allocate(), reload_config(), set_destination(), set_peer_defaults(), sip_parse_host(), sip_show_registry(), sip_standard_port(), and transmit_register().

#define STANDARD_TLS_PORT   5061

Standard SIP TLS port for sips: from RFC 3261. DO NOT CHANGE THIS.

Definition at line 666 of file chan_sip.c.

Referenced by __set_address_from_contact(), build_peer(), create_addr(), parse_register_contact(), reload_config(), sip_parse_host(), and sip_standard_port().

#define SUPPORTED   1

Define SIP option tags, used in Require: and Supported: headers We need to be aware of these properties in the phones to use the replace: header. We should not do that without knowing that the other end supports it... This is nothing we can configure, we learn by the dialog Supported: header on the REGISTER (peer) or the INVITE (other devices) We are not using many of these today, but will in the future. This is documented in RFC 3261

Definition at line 571 of file chan_sip.c.

#define SUPPORTED_EXTENSIONS   "replaces, timer"

SIP Extensions we support.

Note:

This should be generated based on the previous array in combination with settings.

Todo:

We should not have "timer" if it's disabled in the configuration file.

Definition at line 661 of file chan_sip.c.

Referenced by respprep(), transmit_invite(), transmit_notify_with_sipfrag(), transmit_refer(), and transmit_reinvite_with_sdp().

#define T38FAX_FILL_BIT_REMOVAL   (1 << 0)

Default: 0 (unset)

Definition at line 1097 of file chan_sip.c.

Referenced by add_sdp(), and process_sdp().

#define T38FAX_RATE_12000   (1 << 12)

12000 bps t38FaxRate

Definition at line 1116 of file chan_sip.c.

Referenced by process_sdp(), and t38_get_rate().

#define T38FAX_RATE_14400   (1 << 13)

14400 bps t38FaxRate This is default: NO MMR and JBIG transcoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate

Definition at line 1117 of file chan_sip.c.

Referenced by process_sdp(), and t38_get_rate().

#define T38FAX_RATE_2400   (1 << 8)

2400 bps t38FaxRate

Definition at line 1112 of file chan_sip.c.

Referenced by process_sdp(), and t38_get_rate().

#define T38FAX_RATE_4800   (1 << 9)

4800 bps t38FaxRate

Definition at line 1113 of file chan_sip.c.

Referenced by process_sdp(), and t38_get_rate().

#define T38FAX_RATE_7200   (1 << 10)

7200 bps t38FaxRate

Definition at line 1114 of file chan_sip.c.

Referenced by process_sdp(), and t38_get_rate().

#define T38FAX_RATE_9600   (1 << 11)

9600 bps t38FaxRate

Definition at line 1115 of file chan_sip.c.

Referenced by process_sdp(), and t38_get_rate().

#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF   (1 << 3)

Unset for transferredTCF (UDPTL), set for localTCF (TPKT)

Definition at line 1102 of file chan_sip.c.

Referenced by add_sdp(), and process_sdp().

#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF   (0 << 3)

Definition at line 1101 of file chan_sip.c.

Referenced by process_sdp(), and set_t38_capabilities().

#define T38FAX_TRANSCODING_JBIG   (1 << 2)

Default: 0 (unset)

Definition at line 1099 of file chan_sip.c.

Referenced by add_sdp(), and process_sdp().

#define T38FAX_TRANSCODING_MMR   (1 << 1)

Default: 0 (unset)

Definition at line 1098 of file chan_sip.c.

Referenced by add_sdp(), and process_sdp().

#define T38FAX_UDP_EC_FEC   (1 << 4)

Set for t38UDPFEC

Definition at line 1105 of file chan_sip.c.

Referenced by process_sdp(), and set_t38_capabilities().

#define T38FAX_UDP_EC_NONE   (0 << 4)

two bits, if unset NO t38UDPEC field in T38 SDP

Definition at line 1104 of file chan_sip.c.

Referenced by add_sdp(), process_sdp(), and set_t38_capabilities().

#define T38FAX_UDP_EC_REDUNDANCY   (2 << 4)

Set for t38UDPRedundancy

Definition at line 1106 of file chan_sip.c.

Referenced by add_sdp(), process_sdp(), and set_t38_capabilities().

#define T38FAX_VERSION   (3 << 6)

two bits, 2 values so far, up to 4 values max

Definition at line 1108 of file chan_sip.c.

Referenced by add_sdp().

#define T38FAX_VERSION_0   (0 << 6)

Version 0

Definition at line 1109 of file chan_sip.c.

Referenced by add_sdp(), and process_sdp().

#define T38FAX_VERSION_1   (1 << 6)

Version 1

Definition at line 1110 of file chan_sip.c.

Referenced by add_sdp(), and process_sdp().

#define TRUE   1

Definition at line 207 of file chan_sip.c.

#define UNLINK	(	element,
		head,
		prev		)

some list management macros.

Definition at line 1844 of file chan_sip.c.

Referenced by __sip_ack(), and retrans_pkt().

#define XMIT_ERROR   -2

Definition at line 217 of file chan_sip.c.

Referenced by __sip_reliable_xmit(), __sip_xmit(), handle_response_invite(), retrans_pkt(), sip_call(), and sip_poke_peer().

---

Enumeration Type Documentation

enum can_create_dialog

States whether a SIP message can create a dialog in Asterisk.

Enumerator:

CAN_NOT_CREATE_DIALOG
CAN_CREATE_DIALOG
CAN_CREATE_DIALOG_UNSUPPORTED_METHOD

Definition at line 496 of file chan_sip.c.

                       {
   CAN_NOT_CREATE_DIALOG,
   CAN_CREATE_DIALOG,
   CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
};

enum check_auth_result

Authentication result from check_auth* functions.

Enumerator:

AUTH_DONT_KNOW	no result, need to check further
AUTH_SUCCESSFUL
AUTH_CHALLENGE_SENT
AUTH_SECRET_FAILED
AUTH_USERNAME_MISMATCH
AUTH_NOT_FOUND	returned by register_verify
AUTH_FAKE_AUTH
AUTH_UNKNOWN_DOMAIN
AUTH_PEER_NOT_DYNAMIC
AUTH_ACL_FAILED
AUTH_BAD_TRANSPORT

Definition at line 397 of file chan_sip.c.

                       {
   AUTH_DONT_KNOW = -100,  /*!< no result, need to check further */
      /* XXX maybe this is the same as AUTH_NOT_FOUND */

   AUTH_SUCCESSFUL = 0,
   AUTH_CHALLENGE_SENT = 1,
   AUTH_SECRET_FAILED = -1,
   AUTH_USERNAME_MISMATCH = -2,
   AUTH_NOT_FOUND = -3, /*!< returned by register_verify */
   AUTH_FAKE_AUTH = -4,
   AUTH_UNKNOWN_DOMAIN = -5,
   AUTH_PEER_NOT_DYNAMIC = -6,
   AUTH_ACL_FAILED = -7,
   AUTH_BAD_TRANSPORT = -8,
};

enum domain_mode

Modes for SIP domain handling in the PBX.

Enumerator:

SIP_DOMAIN_AUTO	This domain is auto-configured
SIP_DOMAIN_CONFIG	This domain is from configuration

Definition at line 947 of file chan_sip.c.

                 {
   SIP_DOMAIN_AUTO,     /*!< This domain is auto-configured */
   SIP_DOMAIN_CONFIG,      /*!< This domain is from configuration */
};

enum invitestates

States for the INVITE transaction, not the dialog.

Note:

this is for the INVITE that sets up the dialog

Enumerator:

INV_NONE	No state at all, maybe not an INVITE dialog
INV_CALLING	Invite sent, no answer
INV_PROCEEDING	We got/sent 1xx message
INV_EARLY_MEDIA	We got 18x message with to-tag back
INV_COMPLETED	Got final response with error. Wait for ACK, then CONFIRMED
INV_CONFIRMED	Confirmed response - we've got an ack (Incoming calls only)
INV_TERMINATED	Transaction done - either successful (AST_STATE_UP) or failed, but done The only way out of this is a BYE from one side
INV_CANCELLED	Transaction cancelled by client or server in non-terminated state

Definition at line 311 of file chan_sip.c.

                  {
   INV_NONE = 0,          /*!< No state at all, maybe not an INVITE dialog */
   INV_CALLING = 1,  /*!< Invite sent, no answer */
   INV_PROCEEDING = 2,  /*!< We got/sent 1xx message */
   INV_EARLY_MEDIA = 3,    /*!< We got 18x message with to-tag back */
   INV_COMPLETED = 4,   /*!< Got final response with error. Wait for ACK, then CONFIRMED */
   INV_CONFIRMED = 5,   /*!< Confirmed response - we've got an ack (Incoming calls only) */
   INV_TERMINATED = 6,  /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done 
                 The only way out of this is a BYE from one side */
   INV_CANCELLED = 7,   /*!< Transaction cancelled by client or server in non-terminated state */
};

enum parse_register_result

Enumerator:

PARSE_REGISTER_FAILED
PARSE_REGISTER_UPDATE
PARSE_REGISTER_QUERY

Definition at line 349 of file chan_sip.c.

                           {
   PARSE_REGISTER_FAILED,
   PARSE_REGISTER_UPDATE,
   PARSE_REGISTER_QUERY,
};

enum referstatus

Parameters to know status of transfer.

Enumerator:

REFER_IDLE	No REFER is in progress
REFER_SENT	Sent REFER to transferee
REFER_RECEIVED	Received REFER from transferrer
REFER_CONFIRMED	Refer confirmed with a 100 TRYING (unused)
REFER_ACCEPTED	Accepted by transferee
REFER_RINGING	Target Ringing
REFER_200OK	Answered by transfer target
REFER_FAILED	REFER declined - go on
REFER_NOAUTH	We had no auth for REFER

Definition at line 1163 of file chan_sip.c.

                 {
   REFER_IDLE,                    /*!< No REFER is in progress */
   REFER_SENT,                    /*!< Sent REFER to transferee */
   REFER_RECEIVED,                /*!< Received REFER from transferrer */
   REFER_CONFIRMED,               /*!< Refer confirmed with a 100 TRYING (unused) */
   REFER_ACCEPTED,                /*!< Accepted by transferee */
   REFER_RINGING,                 /*!< Target Ringing */
   REFER_200OK,                   /*!< Answered by transfer target */
   REFER_FAILED,                  /*!< REFER declined - go on */
   REFER_NOAUTH                   /*!< We had no auth for REFER */
};

enum sip_auth_type

Authentication types - proxy or www authentication.

Note:

Endpoints, like Asterisk, should always use WWW authentication to allow multiple authentications in the same call - to the proxy and to the end point.

Enumerator:

PROXY_AUTH
WWW_AUTH

Definition at line 391 of file chan_sip.c.

                   {
   PROXY_AUTH = 407,
   WWW_AUTH = 401,
};

enum sip_debug_e

debugging state We store separately the debugging requests from the config file and requests from the CLI. Debugging is enabled if either is set (which means that if sipdebug is set in the config file, we can only turn it off by reloading the config).

Enumerator:

sip_debug_none
sip_debug_config
sip_debug_console

Definition at line 1129 of file chan_sip.c.

                 {
   sip_debug_none = 0,
   sip_debug_config = 1,
   sip_debug_console = 2,
};

enum sip_peer_type

Enumerator:

SIP_TYPE_PEER
SIP_TYPE_USER

Definition at line 1502 of file chan_sip.c.

                   {
   SIP_TYPE_PEER = (1 << 0),
   SIP_TYPE_USER = (1 << 1),
};

enum sip_result

The result of a lot of functions.

Enumerator:

AST_SUCCESS
AST_FAILURE	FALSE means success, funny enough

Definition at line 303 of file chan_sip.c.

                {
   AST_SUCCESS = 0,     /*! FALSE means success, funny enough */
   AST_FAILURE = -1,    
};

enum sip_transport

Define some implemented SIP transports.

Note:

Asterisk does not support SCTP or UDP/DTLS

Enumerator:

SIP_TRANSPORT_UDP	Unreliable transport for SIP, needs retransmissions
SIP_TRANSPORT_TCP	Reliable, but unsecure
SIP_TRANSPORT_TLS	TCP/TLS - reliable and secure transport for signalling

Definition at line 466 of file chan_sip.c.

                   {
   SIP_TRANSPORT_UDP = 1,     /*!< Unreliable transport for SIP, needs retransmissions */
   SIP_TRANSPORT_TCP = 1 << 1,   /*!< Reliable, but unsecure */
   SIP_TRANSPORT_TLS = 1 << 2,   /*!< TCP/TLS - reliable and secure transport for signalling */
};

enum sipmethod

SIP Request methods known by Asterisk.

Note:

Do _NOT_ make any changes to this enum, or the array following it; if you think you are doing the right thing, you are probably not doing the right thing. If you think there are changes needed, get someone else to review them first _before_ submitting a patch. If these two lists do not match properly bad things will happen.

Enumerator:

SIP_UNKNOWN	Unknown response
SIP_RESPONSE	Not request, response to outbound request
SIP_REGISTER	Registration to the mothership, tell us where you are located
SIP_OPTIONS	Check capabilities of a device, used for "ping" too
SIP_NOTIFY	Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER
SIP_INVITE	Set up a session
SIP_ACK	End of a three-way handshake started with INVITE.
SIP_PRACK	Reliable pre-call signalling. Not supported in Asterisk.
SIP_BYE	End of a session
SIP_REFER	Refer to another URI (transfer)
SIP_SUBSCRIBE	Subscribe for updates (voicemail, session status, device status, presence)
SIP_MESSAGE	Text messaging
SIP_UPDATE	Update a dialog. We can send UPDATE; but not accept it
SIP_INFO	Information updates during a session
SIP_CANCEL	Cancel an INVITE
SIP_PUBLISH	Not supported in Asterisk
SIP_PING	Not supported at all, no standard but still implemented out there

Definition at line 512 of file chan_sip.c.

               {
   SIP_UNKNOWN,      /*!< Unknown response */
   SIP_RESPONSE,     /*!< Not request, response to outbound request */
   SIP_REGISTER,     /*!< Registration to the mothership, tell us where you are located */
   SIP_OPTIONS,      /*!< Check capabilities of a device, used for "ping" too */
   SIP_NOTIFY,    /*!< Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER */
   SIP_INVITE,    /*!< Set up a session */
   SIP_ACK,    /*!< End of a three-way handshake started with INVITE. */
   SIP_PRACK,     /*!< Reliable pre-call signalling. Not supported in Asterisk. */
   SIP_BYE,    /*!< End of a session */
   SIP_REFER,     /*!< Refer to another URI (transfer) */
   SIP_SUBSCRIBE,    /*!< Subscribe for updates (voicemail, session status, device status, presence) */
   SIP_MESSAGE,      /*!< Text messaging */
   SIP_UPDATE,    /*!< Update a dialog. We can send UPDATE; but not accept it */
   SIP_INFO,      /*!< Information updates during a session */
   SIP_CANCEL,    /*!< Cancel an INVITE */
   SIP_PUBLISH,      /*!< Not supported in Asterisk */
   SIP_PING,      /*!< Not supported at all, no standard but still implemented out there */
};

enum sipregistrystate

States for outbound registrations (with register= lines in sip.conf.

Enumerator:

REG_STATE_UNREGISTERED	We are not registered Note: Initial state. We should have a timeout scheduled for the initial (or next) registration transmission, calling sip_reregister
REG_STATE_REGSENT	Registration request sent Note: sent initial request, waiting for an ack or a timeout to retransmit the initial request.
REG_STATE_AUTHSENT	We have tried to authenticate Note: entered after transmit_register with auth info, waiting for an ack.
REG_STATE_REGISTERED	Registered and done
REG_STATE_REJECTED	Registration rejected * Note: only used when the remote party has an expire larger than our max-expire. This is a final state from which we do not recover (not sure how correctly).
REG_STATE_TIMEOUT	Registration timed out * Note: XXX unused
REG_STATE_NOAUTH	We have no accepted credentials Note: fatal - no chance to proceed
REG_STATE_FAILED	Registration failed after several tries Note: fatal - no chance to proceed

Definition at line 414 of file chan_sip.c.

                      {
   REG_STATE_UNREGISTERED = 0,   /*!< We are not registered 
       *  \note Initial state. We should have a timeout scheduled for the initial
       * (or next) registration transmission, calling sip_reregister
       */

   REG_STATE_REGSENT,   /*!< Registration request sent 
       * \note sent initial request, waiting for an ack or a timeout to
       * retransmit the initial request.
      */

   REG_STATE_AUTHSENT,  /*!< We have tried to authenticate 
       * \note entered after transmit_register with auth info,
       * waiting for an ack.
       */

   REG_STATE_REGISTERED,   /*!< Registered and done */

   REG_STATE_REJECTED,  /*!< Registration rejected *
       * \note only used when the remote party has an expire larger than
       * our max-expire. This is a final state from which we do not
       * recover (not sure how correctly).
       */

   REG_STATE_TIMEOUT,   /*!< Registration timed out *
      * \note XXX unused */

   REG_STATE_NOAUTH, /*!< We have no accepted credentials
       * \note fatal - no chance to proceed */

   REG_STATE_FAILED, /*!< Registration failed after several tries
       * \note fatal - no chance to proceed */
};

enum st_mode

Modes in which Asterisk can be configured to run SIP Session-Timers.

Enumerator:

SESSION_TIMER_MODE_INVALID	Invalid value
SESSION_TIMER_MODE_ACCEPT	Honor inbound Session-Timer requests
SESSION_TIMER_MODE_ORIGINATE	Originate outbound and honor inbound requests
SESSION_TIMER_MODE_REFUSE	Ignore inbound Session-Timers requests

Definition at line 449 of file chan_sip.c.

             {
        SESSION_TIMER_MODE_INVALID = 0, /*!< Invalid value */ 
        SESSION_TIMER_MODE_ACCEPT,      /*!< Honor inbound Session-Timer requests */
        SESSION_TIMER_MODE_ORIGINATE,   /*!< Originate outbound and honor inbound requests */
        SESSION_TIMER_MODE_REFUSE       /*!< Ignore inbound Session-Timers requests */
};

enum st_refresher

The entity playing the refresher role for Session-Timers.

Enumerator:

SESSION_TIMER_REFRESHER_AUTO	Negotiated
SESSION_TIMER_REFRESHER_UAC	Session is refreshed by the UAC
SESSION_TIMER_REFRESHER_UAS	Session is refreshed by the UAS

Definition at line 457 of file chan_sip.c.

                  {
        SESSION_TIMER_REFRESHER_AUTO,    /*!< Negotiated                      */
        SESSION_TIMER_REFRESHER_UAC,     /*!< Session is refreshed by the UAC */
        SESSION_TIMER_REFRESHER_UAS      /*!< Session is refreshed by the UAS */
};

enum subscriptiontype

Type of subscription, based on the packages we do support.

Enumerator:

NONE
XPIDF_XML
DIALOG_INFO_XML
CPIM_PIDF_XML
PIDF_XML
MWI_NOTIFICATION

Definition at line 356 of file chan_sip.c.

                      { 
   NONE = 0,
   XPIDF_XML,
   DIALOG_INFO_XML,
   CPIM_PIDF_XML,
   PIDF_XML,
   MWI_NOTIFICATION
};

enum t38_action_flag

Enumerator:

SDP_T38_NONE	Do not modify T38 information at all
SDP_T38_INITIATE	Remote side has requested T38 with us
SDP_T38_ACCEPT	Remote side accepted our T38 request

Definition at line 1851 of file chan_sip.c.

                     {
   SDP_T38_NONE = 0, /*!< Do not modify T38 information at all */
   SDP_T38_INITIATE, /*!< Remote side has requested T38 with us */
   SDP_T38_ACCEPT,   /*!< Remote side accepted our T38 request */
};

enum t38state

T38 States for a call.

Enumerator:

T38_DISABLED	Not enabled
T38_LOCAL_REINVITE	Offered from local - REINVITE
T38_PEER_DIRECT	Offered from peer
T38_PEER_REINVITE	Offered from peer - REINVITE
T38_ENABLED	Negotiated (enabled)

Definition at line 1144 of file chan_sip.c.

              {
   T38_DISABLED = 0,                /*!< Not enabled */
   T38_LOCAL_REINVITE,              /*!< Offered from local - REINVITE */
   T38_PEER_DIRECT,                 /*!< Offered from peer */
   T38_PEER_REINVITE,               /*!< Offered from peer - REINVITE */
   T38_ENABLED                      /*!< Negotiated (enabled) */
};

enum transfermodes

Authorization scheme for call transfers.

Note:

Not a bitfield flag, since there are plans for other modes, like "only allow transfers for authenticated devices"

Enumerator:

TRANSFER_OPENFORALL	Allow all SIP transfers
TRANSFER_CLOSED	Allow no SIP transfers

Definition at line 296 of file chan_sip.c.

                   {
   TRANSFER_OPENFORALL,            /*!< Allow all SIP transfers */
   TRANSFER_CLOSED,                /*!< Allow no SIP transfers */
};

enum xmittype

When sending a SIP message, we can send with a few options, depending on type of SIP request. UNRELIABLE is moslty used for responses to repeated requests, where the original response would be sent RELIABLE in an INVITE transaction.

Enumerator:

XMIT_CRITICAL	Transmit critical SIP message reliably, with re-transmits. If it fails, it's critical and will cause a teardown of the session
XMIT_RELIABLE	Transmit SIP message reliably, with re-transmits
XMIT_UNRELIABLE	Transmit SIP message without bothering with re-transmits

Definition at line 342 of file chan_sip.c.

              {
   XMIT_CRITICAL = 2,              /*!< Transmit critical SIP message reliably, with re-transmits.
                                              If it fails, it's critical and will cause a teardown of the session */
   XMIT_RELIABLE = 1,              /*!< Transmit SIP message reliably, with re-transmits */
   XMIT_UNRELIABLE = 0,            /*!< Transmit SIP message without bothering with re-transmits */
};

---

Function Documentation

static const char * __get_header	(	const struct sip_request *	req,
		const char *	name,
		int *	start
	)			[static]

Definition at line 5987 of file chan_sip.c.

References ast_skip_blanks(), find_alias(), sip_request::header, sip_request::headers, sip_request::len, pass, and REQ_OFFSET_TO_STR.

{
   int pass;

   /*
    * Technically you can place arbitrary whitespace both before and after the ':' in
    * a header, although RFC3261 clearly says you shouldn't before, and place just
    * one afterwards.  If you shouldn't do it, what absolute idiot decided it was 
    * a good idea to say you can do it, and if you can do it, why in the hell would.
    * you say you shouldn't.
    * Anyways, pedanticsipchecking controls whether we allow spaces before ':',
    * and we always allow spaces after that for compatibility.
    */
   for (pass = 0; name && pass < 2;pass++) {
      int x, len = strlen(name);
      for (x = *start; x < req->headers; x++) {
         char *header = REQ_OFFSET_TO_STR(req, header[x]);
         if (!strncasecmp(header, name, len)) {
            char *r = header + len; /* skip name */
            if (pedanticsipchecking)
               r = ast_skip_blanks(r);

            if (*r == ':') {
               *start = x+1;
               return ast_skip_blanks(r+1);
            }
         }
      }
      if (pass == 0) /* Try aliases */
         name = find_alias(name, NULL);
   }

   /* Don't return NULL, so get_header is always a valid pointer */
   return "";
}

static void __init_check_auth_buf

(

void

)

[static]

Definition at line 10882 of file chan_sip.c.

{

static void __init_ts_temp_pvt

(

void

)

[static]

A per-thread temporary pvt structure.

Definition at line 1760 of file chan_sip.c.

: The address we bind to */

static void __reg_module

(

void

)

[static]

Definition at line 23599 of file chan_sip.c.

static int __set_address_from_contact	(	const char *	fullcontact,
		struct sockaddr_in *	sin,
		int	tcp
	)			[static]

Definition at line 10524 of file chan_sip.c.

References ahp, ast_copy_string(), ast_gethostbyname(), ast_log(), ast_strlen_zero(), hp, LOG_NOTICE, LOG_WARNING, parse_uri(), STANDARD_SIP_PORT, and STANDARD_TLS_PORT.

Referenced by build_peer(), and set_address_from_contact().

{
   struct hostent *hp;
   struct ast_hostent ahp;
   int port;
   char *host, *pt;
   char contact_buf[256];
   char contact2_buf[256];
   char *contact, *contact2;

   /* Work on a copy */
   ast_copy_string(contact_buf, fullcontact, sizeof(contact_buf));
   ast_copy_string(contact2_buf, fullcontact, sizeof(contact2_buf));
   contact = contact_buf;
   contact2 = contact2_buf;

   /* We have only the part in <brackets> here so we just need to parse a SIP URI.*/
   if (tcp) {
      if (parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL)) {
         if (parse_uri(contact2, "sip:", &contact, NULL, &host, &pt, NULL))
            ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
      }
      port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
   } else {
      if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL))
         ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
      port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
   }

   /* XXX This could block for a long time XXX */
   /* We should only do this if it's a name, not an IP */
   hp = ast_gethostbyname(host, &ahp);
   if (!hp)  {
      ast_log(LOG_WARNING, "Invalid host name in Contact: (can't resolve in DNS) : '%s'\n", host);
      return -1;
   }
   sin->sin_family = AF_INET;
   memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
   sin->sin_port = htons(port);

   return 0;
}

static void __sip_ack	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		int	sipmethod
	)			[static]

Acknowledges receipt of a packet and stops retransmission called with p locked.

Definition at line 3310 of file chan_sip.c.

References ast_debug, ast_free, ast_sched_del(), sip_pvt::callid, sip_pkt::data, dialog_unref(), sip_proxy::force, sip_pkt::is_resp, sip_pkt::method, msg, sip_pkt::next, sip_pvt::outboundproxy, sip_pkt::owner, sip_pvt::packets, sip_pvt::pendinginvite, ref_proxy(), sip_pkt::retransid, sched, sip_pkt::seqno, sip_pvt_lock, sip_pvt_unlock, and UNLINK.

Referenced by __sip_pretend_ack(), handle_incoming(), handle_request_invite(), and handle_response().

{
   struct sip_pkt *cur, *prev = NULL;
   const char *msg = "Not Found";   /* used only for debugging */

   /* If we have an outbound proxy for this dialog, then delete it now since
     the rest of the requests in this dialog needs to follow the routing.
     If obforcing is set, we will keep the outbound proxy during the whole
     dialog, regardless of what the SIP rfc says
   */
   if (p->outboundproxy && !p->outboundproxy->force){
      ref_proxy(p, NULL);
   }

   for (cur = p->packets; cur; prev = cur, cur = cur->next) {
      if (cur->seqno != seqno || cur->is_resp != resp)
         continue;
      if (cur->is_resp || cur->method == sipmethod) {
         msg = "Found";
         if (!resp && (seqno == p->pendinginvite)) {
            ast_debug(1, "Acked pending invite %d\n", p->pendinginvite);
            p->pendinginvite = 0;
         }
         if (cur->retransid > -1) {
            if (sipdebug)
               ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
         }
         /* This odd section is designed to thwart a 
          * race condition in the packet scheduler. There are
          * two conditions under which deleting the packet from the
          * scheduler can fail.
          *
          * 1. The packet has been removed from the scheduler because retransmission
          * is being attempted. The problem is that if the packet is currently attempting
          * retransmission and we are at this point in the code, then that MUST mean
          * that retrans_pkt is waiting on p's lock. Therefore we will relinquish the
          * lock temporarily to allow retransmission.
          *
          * 2. The packet has reached its maximum number of retransmissions and has
          * been permanently removed from the packet scheduler. If this is the case, then
          * the packet's retransid will be set to -1. The atomicity of the setting and checking
          * of the retransid to -1 is ensured since in both cases p's lock is held.
          */
         while (cur->retransid > -1 && ast_sched_del(sched, cur->retransid)) {
            sip_pvt_unlock(p);
            usleep(1);
            sip_pvt_lock(p);
         }
         UNLINK(cur, p->packets, prev);
         dialog_unref(cur->owner, "unref pkt cur->owner dialog from sip ack before freeing pkt");
         if (cur->data)
            ast_free(cur->data);
         ast_free(cur);
         break;
      }
   }
   ast_debug(1, "Stopping retransmission on '%s' of %s %d: Match %s\n",
      p->callid, resp ? "Response" : "Request", seqno, msg);
}

static int __sip_autodestruct

(

const void *

data

)

[static]

Kill a SIP dialog (called only by the scheduler) The scheduler has a reference to this dialog when p->autokillid != -1, and we are called using that reference. So if the event is not rescheduled, we need to call dialog_unref().

Definition at line 3219 of file chan_sip.c.

References sip_pvt::alreadygone, append_history, AST_CAUSE_PROTOCOL_ERROR, ast_debug, AST_EXTENSION_DEACTIVATED, ast_log(), ast_queue_hangup_with_cause(), sip_pvt::autokillid, sip_pvt::callid, DEFAULT_TRANS_TIMEOUT, dialog_unlink_all(), dialog_unref(), LOG_WARNING, sip_pvt::method, NONE, sip_pvt::owner, sip_pvt::packets, sip_pvt::refer, sip_pvt::relatedpeer, SIP_BYE, sip_methods, sip_scheddestroy(), sip_pvt::subscribed, transmit_request_with_auth(), transmit_state_notify(), TRUE, unref_peer(), and XMIT_RELIABLE.

Referenced by sip_scheddestroy(), and sip_show_sched().

{
   struct sip_pvt *p = (struct sip_pvt *)data;

   /* If this is a subscription, tell the phone that we got a timeout */
   if (p->subscribed) {
      transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE);  /* Send last notification */
      p->subscribed = NONE;
      append_history(p, "Subscribestatus", "timeout");
      ast_debug(3, "Re-scheduled destruction of SIP subscription %s\n", p->callid ? p->callid : "<unknown>");
      return 10000;  /* Reschedule this destruction so that we know that it's gone */
   }

   /* If there are packets still waiting for delivery, delay the destruction */
   if (p->packets) {
      ast_debug(3, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>");
      append_history(p, "ReliableXmit", "timeout");
      return 10000;
   }

   if (p->subscribed == MWI_NOTIFICATION)
      if (p->relatedpeer)
         p->relatedpeer = unref_peer(p->relatedpeer, "__sip_autodestruct: unref peer p->relatedpeer");   /* Remove link to peer. If it's realtime, make sure it's gone from memory) */

   /* Reset schedule ID */
   p->autokillid = -1;

   if (p->owner) {
      ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
      ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
   } else if (p->refer && !p->alreadygone) {
      ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
      transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
      append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else {
      append_history(p, "AutoDestroy", "%s", p->callid);
      ast_debug(3, "Auto destroying SIP dialog '%s'\n", p->callid);
      dialog_unlink_all(p, TRUE, TRUE); /* once it's unlinked and unrefd everywhere, it'll be freed automagically */
      /* dialog_unref(p, "unref dialog-- no other matching conditions"); -- unlink all now should finish off the dialog's references and free it. */
      /* sip_destroy(p); */      /* Go ahead and destroy dialog. All attempts to recover is done */
      /* sip_destroy also absorbs the reference */
   }
   dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");
   return 0;
}

static void __sip_destroy	(	struct sip_pvt *	p,
		int	lockowner,
		int	lockdialoglist
	)			[static]

Execute destruction of SIP dialog structure, release memory.

Definition at line 4678 of file chan_sip.c.

References ast_channel::_softhangup, ao2_ref, ast_channel_lock, ast_channel_unlock, ast_debug, ast_free, AST_LIST_REMOVE_HEAD, ast_log(), ast_rtp_destroy(), ast_rtp_get_bridged(), AST_SCHED_DEL_UNREF, AST_SOFTHANGUP_DEV, ast_string_field_free_memory, ast_test_flag, ast_udptl_destroy(), ast_variables_destroy(), ast_verbose, sip_registry::call, sip_peer::call, sip_pvt::callid, DEC_CALL_LIMIT, dialog_unref(), sip_pvt::flags, free_old_route(), sip_history::list, LOG_DEBUG, sip_pvt::method, sip_peer::mwipvt, ast_channel::name, sip_request::next, option_debug, sip_pvt::owner, sip_pvt::registry, registry_unref(), sip_pvt::relatedpeer, sched, sip_debug_test_pvt(), sip_dump_history(), SIP_INC_COUNT, sip_methods, SIP_PAGE2_CALL_ONHOLD, ast_channel::tech_pvt, TRUE, unref_peer(), and update_call_counter().

Referenced by sip_destroy().

{
   struct sip_request *req;

   if (sip_debug_test_pvt(p))
      ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);

   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
      update_call_counter(p, DEC_CALL_LIMIT);
      ast_debug(2, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
   }

   /* Unlink us from the owner if we have one */
   if (p->owner) {
      if (lockowner)
         ast_channel_lock(p->owner);
      if (option_debug)
         ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
      p->owner->tech_pvt = NULL;
      /* Make sure that the channel knows its backend is going away */
      p->owner->_softhangup |= AST_SOFTHANGUP_DEV;
      if (lockowner)
         ast_channel_unlock(p->owner);
      /* Give the channel a chance to react before deallocation */
      usleep(1);
   }

   /* Remove link from peer to subscription of MWI */
   if (p->relatedpeer && p->relatedpeer->mwipvt)
      p->relatedpeer->mwipvt = dialog_unref(p->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
   if (p->relatedpeer && p->relatedpeer->call == p)
      p->relatedpeer->call = dialog_unref(p->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
   
   if (p->relatedpeer)
      p->relatedpeer = unref_peer(p->relatedpeer,"unsetting a dialog relatedpeer field in sip_destroy");
   
   if (p->registry) {
      if (p->registry->call == p)
         p->registry->call = dialog_unref(p->registry->call, "nulling out the registry's call dialog field in unlink_all");
      p->registry = registry_unref(p->registry, "delete p->registry");
   }
   
   if (dumphistory)
      sip_dump_history(p);

   if (p->options)
      ast_free(p->options);

   if (p->notify_headers) {
      ast_variables_destroy(p->notify_headers);
      p->notify_headers = NULL;
   }
   if (p->rtp) {
      ast_rtp_destroy(p->rtp);
   }
   if (p->vrtp) {
      ast_rtp_destroy(p->vrtp);
   }
   if (p->trtp) {
      while (ast_rtp_get_bridged(p->trtp))
         usleep(1);
      ast_rtp_destroy(p->trtp);
   }
   if (p->udptl)
      ast_udptl_destroy(p->udptl);
   if (p->refer)
      ast_free(p->refer);
   if (p->route) {
      free_old_route(p->route);
      p->route = NULL;
   }
   if (p->initreq.data)
      ast_free(p->initreq.data);

   /* Destroy Session-Timers if allocated */
   if (p->stimer) {
      if (p->stimer->st_active == TRUE && p->stimer->st_schedid > -1) {
         AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
               dialog_unref(p, "removing session timer ref"));
      }
      ast_free(p->stimer);
      p->stimer = NULL;
   }

   /* Clear history */
   if (p->history) {
      struct sip_history *hist;
      while ( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) ) {
         ast_free(hist);
         p->history_entries--;
      }
      ast_free(p->history);
      p->history = NULL;
   }

   while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
      ast_free(req);
   }

   if (p->chanvars) {
      ast_variables_destroy(p->chanvars);
      p->chanvars = NULL;
   }

   ast_string_field_free_memory(p);

   if (p->socket.tcptls_session) {
      ao2_ref(p->socket.tcptls_session, -1);
      p->socket.tcptls_session = NULL;
   }
}

static int __sip_do_register

(

struct sip_registry *

r

)

[static]

Register with SIP proxy.

Definition at line 9892 of file chan_sip.c.

References SIP_REGISTER, and transmit_register().

Referenced by sip_reregister().

{
   int res;

   res = transmit_register(r, SIP_REGISTER, NULL, NULL);
   return res;
}

static void __sip_pretend_ack

(

struct sip_pvt *

p

)

[static]

Pretend to ack all packets called with p locked.

Definition at line 3372 of file chan_sip.c.

References __sip_ack(), ast_log(), sip_pkt::data, find_sip_method(), sip_pkt::is_resp, LOG_WARNING, sip_pkt::method, sip_request::method, sip_pvt::packets, sip_pkt::seqno, sip_methods, and ast_str::str.

Referenced by handle_request_bye(), handle_request_cancel(), sip_hangup(), and sip_reg_timeout().

{
   struct sip_pkt *cur = NULL;

   while (p->packets) {
      int method;
      if (cur == p->packets) {
         ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
         return;
      }
      cur = p->packets;
      method = (cur->method) ? cur->method : find_sip_method(cur->data->str);
      __sip_ack(p, cur->seqno, cur->is_resp, method);
   }
}

static enum sip_result __sip_reliable_xmit	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		struct ast_str *	data,
		int	len,
		int	fatal,
		int	sipmethod
	)			[static]

Transmit packet with retransmits.

Returns:

0 on success, -1 on failure to allocate packet

Definition at line 3148 of file chan_sip.c.

References __sip_xmit(), append_history, ast_calloc, ast_debug, AST_FAILURE, ast_free, ast_log(), AST_SCHED_DEL, AST_SCHED_REPLACE_VARIABLE, ast_str_create(), ast_str_set(), AST_SUCCESS, sip_request::data, DEFAULT_RETRANS, dialog_ref(), dialog_unref(), LOG_ERROR, sip_pkt::next, sip_pkt::owner, sip_pvt::packets, sip_pvt::pendinginvite, retrans_pkt(), sched, SIP_INVITE, SIP_TRANSPORT_UDP, sip_pvt::socket, ast_str::str, sip_pvt::timer_t1, sip_pkt::timer_t1, sip_socket::type, and XMIT_ERROR.

Referenced by send_request(), and send_response().

{
   struct sip_pkt *pkt = NULL;
   int siptimer_a = DEFAULT_RETRANS;
   int xmitres = 0;

   if (sipmethod == SIP_INVITE) {
      /* Note this is a pending invite */
      p->pendinginvite = seqno;
   }

   /* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
   /* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
   /* According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
   if (!(p->socket.type & SIP_TRANSPORT_UDP)) {
      xmitres = __sip_xmit(p, data, len); /* Send packet */
      if (xmitres == XMIT_ERROR) {  /* Serious network trouble, no need to try again */
         append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
         return AST_FAILURE;
      } else
         return AST_SUCCESS;
   }

   if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
      return AST_FAILURE;
   /* copy data, add a terminator and save length */
   if (!(pkt->data = ast_str_create(len))) {
      ast_free(pkt);
      return AST_FAILURE;
   }
   ast_str_set(&pkt->data, 0, "%s%s", data->str, "\0");
   pkt->packetlen = len;
   /* copy other parameters from the caller */
   pkt->method = sipmethod;
   pkt->seqno = seqno;
   pkt->is_resp = resp;
   pkt->is_fatal = fatal;
   pkt->owner = dialog_ref(p, "__sip_reliable_xmit: setting pkt->owner");
   pkt->next = p->packets;
   p->packets = pkt; /* Add it to the queue */
   pkt->timer_t1 = p->timer_t1;  /* Set SIP timer T1 */
   pkt->retransid = -1;
   if (pkt->timer_t1)
      siptimer_a = pkt->timer_t1 * 2;

   /* Schedule retransmission */
   AST_SCHED_REPLACE_VARIABLE(pkt->retransid, sched, siptimer_a, retrans_pkt, pkt, 1);
   if (sipdebug)
      ast_debug(4, "*** SIP TIMER: Initializing retransmit timer on packet: Id  #%d\n", pkt->retransid);

   xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);   /* Send packet */

   if (xmitres == XMIT_ERROR) {  /* Serious network trouble, no need to try again */
      append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
      ast_log(LOG_ERROR, "Serious Network Trouble; __sip_xmit returns error for pkt data\n");
      AST_SCHED_DEL(sched, pkt->retransid);
      p->packets = pkt->next;
      pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
      ast_free(pkt->data);
      ast_free(pkt);
      return AST_FAILURE;
   } else {
      return AST_SUCCESS;
   }
}

static int __sip_semi_ack	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		int	sipmethod
	)			[static]

Acks receipt of packet, keep it around (used for provisional responses).

Definition at line 3389 of file chan_sip.c.

References ast_debug, AST_SCHED_DEL, sip_pvt::callid, sip_pkt::data, sip_pkt::is_resp, method_match(), sip_pkt::next, sip_pvt::packets, sip_pkt::retransid, sched, sip_pkt::seqno, sip_methods, ast_str::str, and cfsip_methods::text.

Referenced by handle_response().

{
   struct sip_pkt *cur;
   int res = -1;

   for (cur = p->packets; cur; cur = cur->next) {
      if (cur->seqno == seqno && cur->is_resp == resp &&
         (cur->is_resp || method_match(sipmethod, cur->data->str))) {
         /* this is our baby */
         if (cur->retransid > -1) {
            if (sipdebug)
               ast_debug(4, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
         }
         AST_SCHED_DEL(sched, cur->retransid);
         res = 0;
         break;
      }
   }
   ast_debug(1, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res == -1 ? "Not Found" : "Found");
   return res;
}

static int __sip_xmit	(	struct sip_pvt *	p,
		struct ast_str *	data,
		int	len
	)			[static]

Transmit SIP message Sends a SIP request or response on a given socket (in the pvt) Called by retrans_pkt, send_request, send_response and __sip_reliable_xmit.

Definition at line 2865 of file chan_sip.c.

References ast_debug, ast_inet_ntoa(), ast_log(), ast_mutex_lock(), ast_mutex_unlock(), ast_tcptls_server_write(), sip_request::data, errno, ast_tcptls_session_instance::f, sip_socket::fd, get_transport_pvt(), ast_tcptls_session_instance::lock, LOG_WARNING, sip_prepare_socket(), sip_real_dst(), SIP_TRANSPORT_UDP, sip_pvt::socket, ast_str::str, sip_socket::tcptls_session, sip_socket::type, and XMIT_ERROR.

Referenced by __sip_reliable_xmit(), retrans_pkt(), send_request(), and send_response().

{
   int res = 0;
   const struct sockaddr_in *dst = sip_real_dst(p);

   ast_debug(1, "Trying to put '%.10s' onto %s socket destined for %s:%d\n", data->str, get_transport_pvt(p), ast_inet_ntoa(dst->sin_addr), htons(dst->sin_port));

   if (sip_prepare_socket(p) < 0)
      return XMIT_ERROR;

   if (p->socket.tcptls_session)
      ast_mutex_lock(&p->socket.tcptls_session->lock);

   if (p->socket.type & SIP_TRANSPORT_UDP) 
      res = sendto(p->socket.fd, data->str, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
   else {
      if (p->socket.tcptls_session->f) 
         res = ast_tcptls_server_write(p->socket.tcptls_session, data->str, len);
      else
         ast_debug(2, "No p->socket.tcptls_session->f len=%d\n", len);
   } 

   if (p->socket.tcptls_session)
      ast_mutex_unlock(&p->socket.tcptls_session->lock);

   if (res == -1) {
      switch (errno) {
      case EBADF:       /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
      case EHOSTUNREACH:   /* Host can't be reached */
      case ENETDOWN:       /* Inteface down */
      case ENETUNREACH: /* Network failure */
      case ECONNREFUSED:      /* ICMP port unreachable */ 
         res = XMIT_ERROR; /* Don't bother with trying to transmit again */
      }
   }
   if (res != len)
      ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));

   return res;
}

static int __transmit_response	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		enum xmittype	reliable
	)			[static]

Base transmit response function.

Definition at line 8160 of file chan_sip.c.

References add_header(), add_header_contentLength(), ast_cause2str(), ast_log(), buf, get_header(), ast_channel::hangupcause, LOG_WARNING, sip_pvt::method, sip_pvt::owner, respprep(), send_response(), and SIP_INVITE.

Referenced by transmit_response(), transmit_response_reliable(), and transmit_response_using_temp().

{
   struct sip_request resp;
   int seqno = 0;

   if (reliable && (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1)) {
      ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
      return -1;
   }
   respprep(&resp, p, msg, req);
   add_header_contentLength(&resp, 0);
   /* If we are cancelling an incoming invite for some reason, add information
      about the reason why we are doing this in clear text */
   if (p->method == SIP_INVITE && msg[0] != '1' && p->owner && p->owner->hangupcause) {
      char buf[10];

      add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
      snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
      add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
   }
   return send_response(p, &resp, reliable, seqno);
}

static void __unreg_module

(

void

)

[static]

Definition at line 23599 of file chan_sip.c.

static char * _sip_qualify_peer	(	int	type,
		int	fd,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Send qualify message to peer from cli or manager. Mostly for debugging.

Definition at line 13548 of file chan_sip.c.

References ast_cli(), astman_send_error(), CLI_SHOWUSAGE, CLI_SUCCESS, FALSE, find_peer(), FINDPEERS, s, sip_poke_peer(), TRUE, and unref_peer().

Referenced by manager_sip_qualify_peer(), and sip_qualify_peer().

{
   struct sip_peer *peer;
   int load_realtime;

   if (argc < 4)
      return CLI_SHOWUSAGE;

   load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
   if ((peer = find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE))) {
      sip_poke_peer(peer, 1);
      unref_peer(peer, "qualify: done with peer");
   } else if (type == 0) {
      ast_cli(fd, "Peer '%s' not found\n", argv[3]);
   } else {
      astman_send_error(s, m, "Peer not found");
   }
   return CLI_SUCCESS;
}

static char * _sip_show_peer	(	int	type,
		int	fd,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Show one peer in detail (main function).

Definition at line 13621 of file chan_sip.c.

References sip_peer::accountcode, sip_peer::addr, sip_peer::allowtransfer, sip_peer::amaflags, ARRAY_LEN, ast_callerid_merge(), ast_cdr_flags2str(), ast_check_realtime(), ast_cli(), ast_codec_pref_index(), ast_describe_caller_presentation(), ast_getformatname(), ast_getformatname_multiple(), ast_inet_ntoa(), ast_print_group(), ast_sched_when(), ast_str_alloca, ast_strlen_zero(), ast_test_flag, astman_append(), astman_get_header(), astman_send_error(), sip_peer::auth, sip_peer::autoframing, sip_peer::busy_level, sip_peer::call_limit, sip_peer::callgroup, sip_peer::callingpres, sip_peer::capability, sip_peer::chanvars, sip_peer::cid_name, sip_peer::cid_num, CLI_SHOWUSAGE, CLI_SUCCESS, cli_yesno(), sip_peer::context, sip_peer::defaddr, dtmfmode2str(), sip_peer::expire, FALSE, find_peer(), FINDPEERS, sip_peer::flags, sip_proxy::force, sip_peer::fromdomain, sip_peer::fromuser, sip_peer::fullcontact, get_transport(), sip_peer::ha, sip_peer::host_dynamic, cfsip_options::id, insecure2str(), sip_peer::is_realtime, sip_peer::language, sip_peer::lastmsgssent, sip_peer::maxcallbitrate, sip_auth::md5secret, sip_peer::md5secret, ast_variable::name, sip_proxy::name, sip_peer::name, nat2str(), ast_variable::next, sip_auth::next, sip_peer::outboundproxy, peer_mailboxes_to_str(), peer_status(), sip_peer::pickupgroup, sip_peer::prefs, print_codec_to_cli(), print_group(), sip_peer::qualifyfreq, sip_auth::realm, sip_peer::regexten, s, S_OR, sched, sip_auth::secret, sip_peer::secret, SIP_CAN_REINVITE, SIP_DTMF, SIP_INSECURE, SIP_NAT, sip_options, SIP_PAGE2_ALLOWOVERLAP, SIP_PAGE2_ALLOWSUBSCRIBE, SIP_PAGE2_REGISTERTRYING, SIP_PAGE2_T38SUPPORT_RTP, SIP_PAGE2_T38SUPPORT_TCP, SIP_PAGE2_T38SUPPORT_UDPTL, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_VIDEOSUPPORT, SIP_PROMISCREDIR, SIP_SENDRPID, SIP_TRUSTRPID, SIP_USEREQPHONE, sip_peer::sipoptions, sip_peer::socket, sip_st_cfg::st_max_se, sip_st_cfg::st_min_se, sip_st_cfg::st_mode_oper, sip_st_cfg::st_ref, status, sip_peer::stimer, stmode2str(), ast_str::str, strefresher2str(), sip_peer::subscribecontext, text, sip_peer::timer_b, sip_peer::timer_t1, sip_peer::tohost, transfermode2str(), TRUE, sip_socket::type, unref_peer(), sip_peer::useragent, sip_peer::username, sip_auth::username, ast_variable::value, and sip_peer::vmexten.

Referenced by manager_sip_show_peer(), and sip_show_peer().

{
   char status[30] = "";
   char cbuf[256];
   struct sip_peer *peer;
   char codec_buf[512];
   struct ast_codec_pref *pref;
   struct ast_variable *v;
   struct sip_auth *auth;
   int x = 0, codec = 0, load_realtime;
   int realtimepeers;

   realtimepeers = ast_check_realtime("sippeers");

   if (argc < 4)
      return CLI_SHOWUSAGE;

   load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
   peer = find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE);

   if (s) {    /* Manager */
      if (peer) {
         const char *id = astman_get_header(m, "ActionID");

         astman_append(s, "Response: Success\r\n");
         if (!ast_strlen_zero(id))
            astman_append(s, "ActionID: %s\r\n", id);
      } else {
         snprintf (cbuf, sizeof(cbuf), "Peer %s not found.", argv[3]);
         astman_send_error(s, m, cbuf);
         return CLI_SUCCESS;
      }
   }
   if (peer && type==0 ) { /* Normal listing */
      struct ast_str *mailbox_str = ast_str_alloca(512);
      ast_cli(fd, "\n\n");
      ast_cli(fd, "  * Name       : %s\n", peer->name);
      if (realtimepeers) { /* Realtime is enabled */
         ast_cli(fd, "  Realtime peer: %s\n", peer->is_realtime ? "Yes, cached" : "No");
      }
      ast_cli(fd, "  Secret       : %s\n", ast_strlen_zero(peer->secret)?"<Not set>":"<Set>");
      ast_cli(fd, "  MD5Secret    : %s\n", ast_strlen_zero(peer->md5secret)?"<Not set>":"<Set>");
      for (auth = peer->auth; auth; auth = auth->next) {
         ast_cli(fd, "  Realm-auth   : Realm %-15.15s User %-10.20s ", auth->realm, auth->username);
         ast_cli(fd, "%s\n", !ast_strlen_zero(auth->secret)?"<Secret set>":(!ast_strlen_zero(auth->md5secret)?"<MD5secret set>" : "<Not set>"));
      }
      ast_cli(fd, "  Context      : %s\n", peer->context);
      ast_cli(fd, "  Subscr.Cont. : %s\n", S_OR(peer->subscribecontext, "<Not set>") );
      ast_cli(fd, "  Language     : %s\n", peer->language);
      if (!ast_strlen_zero(peer->accountcode))
         ast_cli(fd, "  Accountcode  : %s\n", peer->accountcode);
      ast_cli(fd, "  AMA flags    : %s\n", ast_cdr_flags2str(peer->amaflags));
      ast_cli(fd, "  Transfer mode: %s\n", transfermode2str(peer->allowtransfer));
      ast_cli(fd, "  CallingPres  : %s\n", ast_describe_caller_presentation(peer->callingpres));
      if (!ast_strlen_zero(peer->fromuser))
         ast_cli(fd, "  FromUser     : %s\n", peer->fromuser);
      if (!ast_strlen_zero(peer->fromdomain))
         ast_cli(fd, "  FromDomain   : %s\n", peer->fromdomain);
      ast_cli(fd, "  Callgroup    : ");
      print_group(fd, peer->callgroup, 0);
      ast_cli(fd, "  Pickupgroup  : ");
      print_group(fd, peer->pickupgroup, 0);
      peer_mailboxes_to_str(&mailbox_str, peer);
      ast_cli(fd, "  Mailbox      : %s\n", mailbox_str->str);
      ast_cli(fd, "  VM Extension : %s\n", peer->vmexten);
      ast_cli(fd, "  LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
      ast_cli(fd, "  Call limit   : %d\n", peer->call_limit);
      if (peer->busy_level)
         ast_cli(fd, "  Busy level   : %d\n", peer->busy_level);
      ast_cli(fd, "  Dynamic      : %s\n", cli_yesno(peer->host_dynamic));
      ast_cli(fd, "  Callerid     : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
      ast_cli(fd, "  MaxCallBR    : %d kbps\n", peer->maxcallbitrate);
      ast_cli(fd, "  Expire       : %ld\n", ast_sched_when(sched, peer->expire));
      ast_cli(fd, "  Insecure     : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
      ast_cli(fd, "  Nat          : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
      ast_cli(fd, "  ACL          : %s\n", cli_yesno(peer->ha != NULL));
      ast_cli(fd, "  T38 pt UDPTL : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
      ast_cli(fd, "  T38 pt RTP   : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP)));
      ast_cli(fd, "  T38 pt TCP   : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP)));
#endif
      ast_cli(fd, "  CanReinvite  : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)));
      ast_cli(fd, "  PromiscRedir : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
      ast_cli(fd, "  User=Phone   : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
      ast_cli(fd, "  Video Support: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)));
      ast_cli(fd, "  Text Support : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)));
      ast_cli(fd, "  Trust RPID   : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
      ast_cli(fd, "  Send RPID    : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
      ast_cli(fd, "  Subscriptions: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
      ast_cli(fd, "  Overlap dial : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
      if (peer->outboundproxy)
         ast_cli(fd, "  Outb. proxy  : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "<not set>" : peer->outboundproxy->name,
                     peer->outboundproxy->force ? "(forced)" : "");

      /* - is enumerated */
      ast_cli(fd, "  DTMFmode     : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
      ast_cli(fd, "  Timer T1     : %d\n", peer->timer_t1);
      ast_cli(fd, "  Timer B      : %d\n", peer->timer_b);
      ast_cli(fd, "  ToHost       : %s\n", peer->tohost);
      ast_cli(fd, "  Addr->IP     : %s Port %d\n",  peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port));
      ast_cli(fd, "  Defaddr->IP  : %s Port %d\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
      ast_cli(fd, "  Transport    : %s\n", get_transport(peer->socket.type));
      if (!ast_strlen_zero(global_regcontext))
         ast_cli(fd, "  Reg. exten   : %s\n", peer->regexten);
      ast_cli(fd, "  Def. Username: %s\n", peer->username);
      ast_cli(fd, "  SIP Options  : ");
      if (peer->sipoptions) {
         int lastoption = -1;
         for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
            if (sip_options[x].id != lastoption) {
               if (peer->sipoptions & sip_options[x].id)
                  ast_cli(fd, "%s ", sip_options[x].text);
               lastoption = x;
            }
         }
      } else
         ast_cli(fd, "(none)");

      ast_cli(fd, "\n");
      ast_cli(fd, "  Codecs       : ");
      ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
      ast_cli(fd, "%s\n", codec_buf);
      ast_cli(fd, "  Codec Order  : (");
      print_codec_to_cli(fd, &peer->prefs);
      ast_cli(fd, ")\n");

      ast_cli(fd, "  Auto-Framing :  %s \n", cli_yesno(peer->autoframing));
      ast_cli(fd, "  100 on REG   : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_REGISTERTRYING) ? "Yes" : "No");
      ast_cli(fd, "  Status       : ");
      peer_status(peer, status, sizeof(status));
      ast_cli(fd, "%s\n", status);
      ast_cli(fd, "  Useragent    : %s\n", peer->useragent);
      ast_cli(fd, "  Reg. Contact : %s\n", peer->fullcontact);
      ast_cli(fd, "  Qualify Freq : %d ms\n", peer->qualifyfreq);
      if (peer->chanvars) {
         ast_cli(fd, "  Variables    :\n");
         for (v = peer->chanvars ; v ; v = v->next)
            ast_cli(fd, "                 %s = %s\n", v->name, v->value);
      }

      ast_cli(fd, "  Sess-Timers  : %s\n", stmode2str(peer->stimer.st_mode_oper));
      ast_cli(fd, "  Sess-Refresh : %s\n", strefresher2str(peer->stimer.st_ref));
      ast_cli(fd, "  Sess-Expires : %d secs\n", peer->stimer.st_max_se);
      ast_cli(fd, "  Min-Sess     : %d secs\n", peer->stimer.st_min_se);
      ast_cli(fd, "\n");
      peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer ptr");
   } else  if (peer && type == 1) { /* manager listing */
      char buffer[256];
      struct ast_str *mailbox_str = ast_str_alloca(512);
      astman_append(s, "Channeltype: SIP\r\n");
      astman_append(s, "ObjectName: %s\r\n", peer->name);
      astman_append(s, "ChanObjectType: peer\r\n");
      astman_append(s, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y");
      astman_append(s, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y");
      astman_append(s, "Context: %s\r\n", peer->context);
      astman_append(s, "Language: %s\r\n", peer->language);
      if (!ast_strlen_zero(peer->accountcode))
         astman_append(s, "Accountcode: %s\r\n", peer->accountcode);
      astman_append(s, "AMAflags: %s\r\n", ast_cdr_flags2str(peer->amaflags));
      astman_append(s, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres));
      if (!ast_strlen_zero(peer->fromuser))
         astman_append(s, "SIP-FromUser: %s\r\n", peer->fromuser);
      if (!ast_strlen_zero(peer->fromdomain))
         astman_append(s, "SIP-FromDomain: %s\r\n", peer->fromdomain);
      astman_append(s, "Callgroup: ");
      astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->callgroup));
      astman_append(s, "Pickupgroup: ");
      astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->pickupgroup));
      peer_mailboxes_to_str(&mailbox_str, peer);
      astman_append(s, "VoiceMailbox: %s\r\n", mailbox_str->str);
      astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
      astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
      astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
      astman_append(s, "Busy-level: %d\r\n", peer->busy_level);
      astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
      astman_append(s, "Dynamic: %s\r\n", peer->host_dynamic?"Y":"N");
      astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
      astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire));
      astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
      astman_append(s, "SIP-NatSupport: %s\r\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
      astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N"));
      astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Y":"N"));
      astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
      astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
      astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
      astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
      astman_append(s, "SIP-Sess-Timers: %s\r\n", stmode2str(peer->stimer.st_mode_oper));
      astman_append(s, "SIP-Sess-Refresh: %s\r\n", strefresher2str(peer->stimer.st_ref));
      astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
      astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);

      /* - is enumerated */
      astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
      astman_append(s, "ToHost: %s\r\n", peer->tohost);
      astman_append(s, "Address-IP: %s\r\nAddress-Port: %d\r\n",  peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "", ntohs(peer->addr.sin_port));
      astman_append(s, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
      astman_append(s, "Default-Username: %s\r\n", peer->username);
      if (!ast_strlen_zero(global_regcontext))
         astman_append(s, "RegExtension: %s\r\n", peer->regexten);
      astman_append(s, "Codecs: ");
      ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
      astman_append(s, "%s\r\n", codec_buf);
      astman_append(s, "CodecOrder: ");
      pref = &peer->prefs;
      for(x = 0; x < 32 ; x++) {
         codec = ast_codec_pref_index(pref, x);
         if (!codec)
            break;
         astman_append(s, "%s", ast_getformatname(codec));
         if (x < 31 && ast_codec_pref_index(pref, x+1))
            astman_append(s, ",");
      }

      astman_append(s, "\r\n");
      astman_append(s, "Status: ");
      peer_status(peer, status, sizeof(status));
      astman_append(s, "%s\r\n", status);
      astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
      astman_append(s, "Reg-Contact: %s\r\n", peer->fullcontact);
      astman_append(s, "QualifyFreq: %d ms\r\n", peer->qualifyfreq);
      if (peer->chanvars) {
         for (v = peer->chanvars ; v ; v = v->next) {
            astman_append(s, "ChanVariable: %s=%s\r\n", v->name, v->value);
         }
      }

      peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer");

   } else {
      ast_cli(fd, "Peer %s not found.\n", argv[3]);
      ast_cli(fd, "\n");
   }

   return CLI_SUCCESS;
}

static char * _sip_show_peers	(	int	fd,
		int *	total,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Execute sip show peers command.

Definition at line 12943 of file chan_sip.c.

References sip_peer::addr, ao2_container_count(), ao2_iterator_init(), ao2_lock(), ao2_t_iterator_next, ao2_unlock(), ast_calloc, ast_check_realtime(), ast_cli(), ast_copy_string(), ast_free, ast_inet_ntoa(), ast_strlen_zero(), ast_test_flag, astman_append(), astman_get_header(), CLI_SHOWUSAGE, CLI_SUCCESS, FALSE, sip_peer::flags, FORMAT, FORMAT2, sip_peer::ha, sip_peer::host_dynamic, id, sip_peer::is_realtime, sip_peer::name, name, peer_status(), peercomparefunc(), s, SIP_NAT_ROUTE, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_VIDEOSUPPORT, SIP_TYPE_PEER, status, TRUE, sip_peer::type, unref_peer(), and sip_peer::username.

Referenced by manager_sip_show_peers(), and sip_show_peers().

{
   regex_t regexbuf;
   int havepattern = FALSE;
   struct sip_peer *peer;
   struct ao2_iterator i;
   
/* the last argument is left-aligned, so we don't need a size anyways */
#define FORMAT2 "%-25.25s  %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %s\n"
#define FORMAT  "%-25.25s  %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"

   char name[256];
   int total_peers = 0;
   int peers_mon_online = 0;
   int peers_mon_offline = 0;
   int peers_unmon_offline = 0;
   int peers_unmon_online = 0;
   const char *id;
   char idtext[256] = "";
   int realtimepeers;
   int objcount = ao2_container_count(peers);
   struct sip_peer **peerarray;
   int k;
   
   
   realtimepeers = ast_check_realtime("sippeers");
   peerarray = ast_calloc(sizeof(struct sip_peer *), objcount);

   if (s) { /* Manager - get ActionID */
      id = astman_get_header(m, "ActionID");
      if (!ast_strlen_zero(id))
         snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);
   }

   switch (argc) {
   case 5:
      if (!strcasecmp(argv[3], "like")) {
         if (regcomp(&regexbuf, argv[4], REG_EXTENDED | REG_NOSUB))
            return CLI_SHOWUSAGE;
         havepattern = TRUE;
      } else
         return CLI_SHOWUSAGE;
   case 3:
      break;
   default:
      return CLI_SHOWUSAGE;
   }

   if (!s) /* Normal list */
      ast_cli(fd, FORMAT2, "Name/username", "Host", "Dyn", "Nat", "ACL", "Port", "Status", (realtimepeers ? "Realtime" : ""));
   

   i = ao2_iterator_init(peers, 0);
   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {  
      ao2_lock(peer);

      if (!(peer->type & SIP_TYPE_PEER)) {
         ao2_unlock(peer);
         unref_peer(peer, "unref peer because it's actually a user");
         continue;
      }

      if (havepattern && regexec(&regexbuf, peer->name, 0, NULL, 0)) {
         objcount--;
         ao2_unlock(peer);
         unref_peer(peer, "toss iterator peer ptr before continue");
         continue;
      }

      peerarray[total_peers++] = peer;
      ao2_unlock(peer);
   }
   
   qsort(peerarray, total_peers, sizeof(struct sip_peer *), peercomparefunc);

   for(k=0; k < total_peers; k++) {
      char status[20] = "";
      char srch[2000];
      char pstatus;
      peer = peerarray[k];
      
      ao2_lock(peer);
      if (havepattern && regexec(&regexbuf, peer->name, 0, NULL, 0)) {
         ao2_unlock(peer);
         unref_peer(peer, "toss iterator peer ptr before continue");
         continue;
      }

      if (!ast_strlen_zero(peer->username) && !s)
         snprintf(name, sizeof(name), "%s/%s", peer->name, peer->username);
      else
         ast_copy_string(name, peer->name, sizeof(name));
      
      pstatus = peer_status(peer, status, sizeof(status));
      if (pstatus == 1)
         peers_mon_online++;
      else if (pstatus == 0)
         peers_mon_offline++;
      else {
         if (peer->addr.sin_port == 0)
            peers_unmon_offline++;
         else
            peers_unmon_online++;
      }

      snprintf(srch, sizeof(srch), FORMAT, name,
         peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)",
         peer->host_dynamic ? " D " : "   ",    /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? " N " : "   ", /* NAT=yes? */
         peer->ha ? " A " : "   ",  /* permit/deny */
         ntohs(peer->addr.sin_port), status,
         realtimepeers ? (peer->is_realtime ? "Cached RT":"") : "");

      if (!s)  {/* Normal CLI list */
         ast_cli(fd, FORMAT, name, 
         peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)",
         peer->host_dynamic ? " D " : "   ",    /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? " N " : "   ", /* NAT=yes? */
         peer->ha ? " A " : "   ",       /* permit/deny */
         
         ntohs(peer->addr.sin_port), status,
         realtimepeers ? (peer->is_realtime ? "Cached RT":"") : "");
      } else { /* Manager format */
         /* The names here need to be the same as other channels */
         astman_append(s, 
         "Event: PeerEntry\r\n%s"
         "Channeltype: SIP\r\n"
         "ObjectName: %s\r\n"
         "ChanObjectType: peer\r\n" /* "peer" or "user" */
         "IPaddress: %s\r\n"
         "IPport: %d\r\n"
         "Dynamic: %s\r\n"
         "Natsupport: %s\r\n"
         "VideoSupport: %s\r\n"
         "TextSupport: %s\r\n"
         "ACL: %s\r\n"
         "Status: %s\r\n"
         "RealtimeDevice: %s\r\n\r\n", 
         idtext,
         peer->name, 
         peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "-none-",
         ntohs(peer->addr.sin_port), 
         peer->host_dynamic ? "yes" : "no",  /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? "yes" : "no",  /* NAT=yes? */
         ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no",  /* VIDEOSUPPORT=yes? */
         ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no",   /* TEXTSUPPORT=yes? */
         peer->ha ? "yes" : "no",       /* permit/deny */
         status,
         realtimepeers ? (peer->is_realtime ? "yes":"no") : "no");
      }
      ao2_unlock(peer);
      unref_peer(peer, "toss iterator peer ptr");
   }
   
   if (!s)
      ast_cli(fd, "%d sip peers [Monitored: %d online, %d offline Unmonitored: %d online, %d offline]\n",
              total_peers, peers_mon_online, peers_mon_offline, peers_unmon_online, peers_unmon_offline);

   if (havepattern)
      regfree(&regexbuf);

   if (total)
      *total = total_peers;
   
   ast_free(peerarray);
   
   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static void * _sip_tcp_helper_thread	(	struct sip_pvt *	pvt,
		struct ast_tcptls_session_instance *	tcptls_session
	)			[static]

SIP TCP thread management function.

Definition at line 2368 of file chan_sip.c.

References ao2_ref, ast_calloc, ast_debug, ast_free, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, ast_mutex_lock(), ast_mutex_unlock(), ast_str_append(), ast_str_create(), ast_str_reset(), ast_wait_for_input(), buf, cleanup(), copy_request(), sip_request::data, ast_tcptls_session_instance::f, ast_tcptls_session_instance::fd, get_header(), handle_request_do(), sip_request::len, sip_threadinfo::list, ast_tcptls_session_instance::lock, me, ourport_tcp, ourport_tls, parse_request(), ast_tcptls_session_instance::remote_address, REQ_OFFSET_TO_STR, SIP_MIN_PACKET, SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, ast_tcptls_session_instance::ssl, and sip_threadinfo::tcptls_session.

Referenced by sip_tcp_worker_fn().

{
   int res, cl;
   struct sip_request req = { 0, } , reqcpy = { 0, };
   struct sip_threadinfo *me;
   char buf[1024] = "";

   me = ast_calloc(1, sizeof(*me));

   if (!me)
      goto cleanup2;

   me->threadid = pthread_self();
   me->tcptls_session = tcptls_session;
   if (tcptls_session->ssl)
      me->type = SIP_TRANSPORT_TLS;
   else
      me->type = SIP_TRANSPORT_TCP;

   ast_debug(2, "Starting thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");

   AST_LIST_LOCK(&threadl);
   AST_LIST_INSERT_TAIL(&threadl, me, list);
   AST_LIST_UNLOCK(&threadl);

   if (!(req.data = ast_str_create(SIP_MIN_PACKET)))
      goto cleanup;
   if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET)))
      goto cleanup;

   for (;;) {
      struct ast_str *str_save;

      str_save = req.data;
      memset(&req, 0, sizeof(req));
      req.data = str_save;
      ast_str_reset(req.data);

      str_save = reqcpy.data;
      memset(&reqcpy, 0, sizeof(reqcpy));
      reqcpy.data = str_save;
      ast_str_reset(reqcpy.data);

      req.socket.fd = tcptls_session->fd;
      if (tcptls_session->ssl) {
         req.socket.type = SIP_TRANSPORT_TLS;
         req.socket.port = htons(ourport_tls);
      } else {
         req.socket.type = SIP_TRANSPORT_TCP;
         req.socket.port = htons(ourport_tcp);
      }
      res = ast_wait_for_input(tcptls_session->fd, -1);
      if (res < 0) {
         ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "SSL": "TCP", res);
         goto cleanup;
      }

      /* Read in headers one line at a time */
      while (req.len < 4 || strncmp(REQ_OFFSET_TO_STR(&req, len - 4), "\r\n\r\n", 4)) {
         ast_mutex_lock(&tcptls_session->lock);
         if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
            ast_mutex_unlock(&tcptls_session->lock);
            goto cleanup;
         }
         ast_mutex_unlock(&tcptls_session->lock);
         if (me->stop) 
             goto cleanup;
         ast_str_append(&req.data, 0, "%s", buf);
         req.len = req.data->used;
      }
      copy_request(&reqcpy, &req);
      parse_request(&reqcpy);
      if (sscanf(get_header(&reqcpy, "Content-Length"), "%d", &cl)) {
         while (cl > 0) {
            ast_mutex_lock(&tcptls_session->lock);
            if (!fread(buf, (cl < sizeof(buf)) ? cl : sizeof(buf), 1, tcptls_session->f)) {
               ast_mutex_unlock(&tcptls_session->lock);
               goto cleanup;
            }
            ast_mutex_unlock(&tcptls_session->lock);
            if (me->stop)
               goto cleanup;
            cl -= strlen(buf);
            ast_str_append(&req.data, 0, "%s", buf);
            req.len = req.data->used;
         }
      }
      req.socket.tcptls_session = tcptls_session;
      handle_request_do(&req, &tcptls_session->remote_address);
   }

cleanup:
   AST_LIST_LOCK(&threadl);
   AST_LIST_REMOVE(&threadl, me, list);
   AST_LIST_UNLOCK(&threadl);
   ast_free(me);
cleanup2:
   fclose(tcptls_session->f);
   tcptls_session->f = NULL;
   tcptls_session->fd = -1;
   if (reqcpy.data) {
      ast_free(reqcpy.data);
   }

   if (req.data) {
      ast_free(req.data);
      req.data = NULL;
   }

   ast_debug(2, "Shutting down thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");

   ao2_ref(tcptls_session, -1);
   tcptls_session = NULL;

   return NULL;
}

static int acf_channel_read	(	struct ast_channel *	chan,
		const char *	funcname,
		char *	preparse,
		char *	buf,
		size_t	buflen
	)			[static]

Definition at line 18939 of file chan_sip.c.

References AST_APP_ARG, ast_copy_string(), AST_DECLARE_APP_ARGS, ast_inet_ntoa(), ast_log(), ast_rtp_get_peer(), ast_rtp_get_qos(), ast_rtp_get_quality(), AST_STANDARD_APP_ARGS, ast_strdupa, ast_strlen_zero(), chan, sip_pvt::from, IS_SIP_TECH, LOG_ERROR, LOG_WARNING, parse(), sip_pvt::peername, qos, sip_pvt::recv, sip_pvt::rtp, RTPQOS_SUMMARY, sip_pvt::sa, t38properties::state, sip_pvt::t38, T38_DISABLED, ast_channel::tech, ast_channel::tech_pvt, sip_pvt::trtp, sip_pvt::uri, sip_pvt::useragent, and sip_pvt::vrtp.

{
   struct sip_pvt *p = chan->tech_pvt;
   char *all = "", *parse = ast_strdupa(preparse);
   int res = 0;
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(param);
      AST_APP_ARG(type);
      AST_APP_ARG(field);
   );
   AST_STANDARD_APP_ARGS(args, parse);

   /* Sanity check */
   if (!IS_SIP_TECH(chan->tech)) {
      ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
      return 0;
   }

   memset(buf, 0, buflen);

   if (!strcasecmp(args.param, "peerip")) {
      ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", buflen);
   } else if (!strcasecmp(args.param, "recvip")) {
      ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", buflen);
   } else if (!strcasecmp(args.param, "from")) {
      ast_copy_string(buf, p->from, buflen);
   } else if (!strcasecmp(args.param, "uri")) {
      ast_copy_string(buf, p->uri, buflen);
   } else if (!strcasecmp(args.param, "useragent")) {
      ast_copy_string(buf, p->useragent, buflen);
   } else if (!strcasecmp(args.param, "peername")) {
      ast_copy_string(buf, p->peername, buflen);
   } else if (!strcasecmp(args.param, "t38passthrough")) {
      ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen);
   } else if (!strcasecmp(args.param, "rtpdest")) {
      struct sockaddr_in sin;

      if (ast_strlen_zero(args.type))
         args.type = "audio";

      if (!strcasecmp(args.type, "audio"))
         ast_rtp_get_peer(p->rtp, &sin);
      else if (!strcasecmp(args.type, "video"))
         ast_rtp_get_peer(p->vrtp, &sin);
      else if (!strcasecmp(args.type, "text"))
         ast_rtp_get_peer(p->trtp, &sin);
      else
         return -1;

      snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
   } else if (!strcasecmp(args.param, "rtpqos")) {
      struct ast_rtp_quality qos;
      struct ast_rtp *rtp = p->rtp;
      
      memset(&qos, 0, sizeof(qos));

      if (ast_strlen_zero(args.type))
         args.type = "audio";
      if (ast_strlen_zero(args.field))
         args.field = "all";
      
      if (!strcasecmp(args.type, "AUDIO")) {
         all = ast_rtp_get_quality(rtp = p->rtp, &qos, RTPQOS_SUMMARY);
      } else if (!strcasecmp(args.type, "VIDEO")) {
         all = ast_rtp_get_quality(rtp = p->vrtp, &qos, RTPQOS_SUMMARY);
      } else if (!strcasecmp(args.type, "TEXT")) {
         all = ast_rtp_get_quality(rtp = p->trtp, &qos, RTPQOS_SUMMARY);
      } else {
         return -1;
      }
      
      if (!strcasecmp(args.field, "local_ssrc"))
         snprintf(buf, buflen, "%u", qos.local_ssrc);
      else if (!strcasecmp(args.field, "local_lostpackets"))
         snprintf(buf, buflen, "%u", qos.local_lostpackets);
      else if (!strcasecmp(args.field, "local_jitter"))
         snprintf(buf, buflen, "%.0f", qos.local_jitter * 1000.0);
      else if (!strcasecmp(args.field, "local_count"))
         snprintf(buf, buflen, "%u", qos.local_count);
      else if (!strcasecmp(args.field, "remote_ssrc"))
         snprintf(buf, buflen, "%u", qos.remote_ssrc);
      else if (!strcasecmp(args.field, "remote_lostpackets"))
         snprintf(buf, buflen, "%u", qos.remote_lostpackets);
      else if (!strcasecmp(args.field, "remote_jitter"))
         snprintf(buf, buflen, "%.0f", qos.remote_jitter * 1000.0);
      else if (!strcasecmp(args.field, "remote_count"))
         snprintf(buf, buflen, "%u", qos.remote_count);
      else if (!strcasecmp(args.field, "rtt"))
         snprintf(buf, buflen, "%.0f", qos.rtt * 1000.0);
      else if (!strcasecmp(args.field, "all"))
         ast_copy_string(buf, all, buflen);
      else if (!ast_rtp_get_qos(rtp, args.field, buf, buflen))
          ;
      else {
         ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
         return -1;
      }
   } else {
      res = -1;
   }
   return res;
}

static void add_blank

(

struct sip_request *

req

)

[static]

add a blank line if no body

Definition at line 3420 of file chan_sip.c.

References ast_str_append(), sip_request::data, sip_request::len, sip_request::lines, and ast_str::used.

Referenced by send_request(), and send_response().

{
   if (!req->lines) {
      /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
      ast_str_append(&req->data, 0, "\r\n");
      req->len = req->data->used;
   }
}

static void add_codec_to_sdp	(	const struct sip_pvt *	p,
		int	codec,
		int	sample_rate,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)			[static]

Add codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 8410 of file chan_sip.c.

References ast_codec_pref_getsize(), AST_FORMAT_G723_1, AST_FORMAT_G729A, AST_FORMAT_ILBC, ast_getformatname(), ast_rtp_codec_getpref(), ast_rtp_lookup_code(), ast_rtp_lookup_mime_subtype(), AST_RTP_OPT_G726_NONSTANDARD, ast_str_append(), ast_test_flag, ast_verbose, ast_format_list::cur_ms, sip_pvt::flags, sip_pvt::rtp, and SIP_G726_NONSTANDARD.

Referenced by add_sdp().

{
   int rtp_code;
   struct ast_format_list fmt;


   if (debug)
      ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
   if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1)
      return;

   if (p->rtp) {
      struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
      fmt = ast_codec_pref_getsize(pref, codec);
   } else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
      return;
   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
          ast_rtp_lookup_mime_subtype(1, codec,
                       ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
          sample_rate);
   if (codec == AST_FORMAT_G729A) {
      /* Indicate that we don't support VAD (G.729 annex B) */
      ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
   } else if (codec == AST_FORMAT_G723_1) {
      /* Indicate that we don't support VAD (G.723.1 annex A) */
      ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
   } else if (codec == AST_FORMAT_ILBC) {
      /* Add information about us using only 20/30 ms packetization */
      ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
   }

   if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
      *min_packet_size = fmt.cur_ms;

   /* Our first codec packetization processed cannot be zero */
   if ((*min_packet_size)==0 && fmt.cur_ms)
      *min_packet_size = fmt.cur_ms;
}

static int add_digit	(	struct sip_request *	req,
		char	digit,
		unsigned int	duration,
		int	mode
	)			[static]

Add DTMF INFO tone to sip message Mode = 0 for application/dtmf-relay (Cisco) 1 for application/dtmf.

Definition at line 8361 of file chan_sip.c.

References add_header(), add_header_contentLength(), and add_line().

Referenced by transmit_info_with_digit().

{
   char tmp[256];
   int event;
   if (mode) {
      /* Application/dtmf short version used by some implementations */
      if (digit == '*')
         event = 10;
      else if (digit == '#')
         event = 11;
      else if ((digit >= 'A') && (digit <= 'D'))
         event = 12 + digit - 'A';
      else
         event = atoi(&digit);
      snprintf(tmp, sizeof(tmp), "%d\r\n", event);
      add_header(req, "Content-Type", "application/dtmf");
      add_header_contentLength(req, strlen(tmp));
      add_line(req, tmp);
   } else {
      /* Application/dtmf-relay as documented by Cisco */
      snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=%u\r\n", digit, duration);
      add_header(req, "Content-Type", "application/dtmf-relay");
      add_header_contentLength(req, strlen(tmp));
      add_line(req, tmp);
   }
   return 0;
}

static int add_header	(	struct sip_request *	req,
		const char *	var,
		const char *	value
	)			[static]

Add header to SIP message.

Definition at line 7631 of file chan_sip.c.

References ast_log(), ast_str_append(), sip_request::data, find_alias(), sip_request::header, sip_request::headers, sip_request::len, sip_request::lines, LOG_WARNING, SIP_MAX_HEADERS, and ast_str::used.

{
   if (req->headers == SIP_MAX_HEADERS) {
      ast_log(LOG_WARNING, "Out of SIP header space\n");
      return -1;
   }

   if (req->lines) {
      ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n");
      return -1;
   }

   if (compactheaders)
      var = find_alias(var, var);

   ast_str_append(&req->data, 0, "%s: %s\r\n", var, value);
   req->header[req->headers] = req->len;

   req->len = req->data->used;
   req->headers++;

   return 0;   
}

static int add_header_contentLength	(	struct sip_request *	req,
		int	len
	)			[static]

Add 'Content-Length' header to SIP message.

Definition at line 7656 of file chan_sip.c.

References add_header().

Referenced by __transmit_response(), add_digit(), add_sdp(), add_text(), add_vidupdate(), transmit_invite(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), transmit_register(), transmit_request(), transmit_request_with_auth(), transmit_response_with_allow(), transmit_response_with_auth(), transmit_response_with_date(), transmit_response_with_minse(), transmit_response_with_unsupported(), and transmit_state_notify().

{
   char clen[10];

   snprintf(clen, sizeof(clen), "%d", len);
   return add_header(req, "Content-Length", clen);
}

static int add_line	(	struct sip_request *	req,
		const char *	line
	)			[static]

Add content (not header) to SIP message.

Definition at line 7665 of file chan_sip.c.

References ast_log(), ast_str_append(), sip_request::data, sip_request::len, sip_request::line, sip_request::lines, LOG_WARNING, SIP_MAX_LINES, and ast_str::used.

{
   if (req->lines == SIP_MAX_LINES)  {
      ast_log(LOG_WARNING, "Out of SIP line space\n");
      return -1;
   }
   if (!req->lines)
      /* Add extra empty return */
      req->len += ast_str_append(&req->data, 0, "\r\n");
   req->line[req->lines] = req->len;
   ast_str_append(&req->data, 0, "%s", line);
   req->len = req->data->used;
   req->lines++;
   return 0;   
}

static void add_noncodec_to_sdp	(	const struct sip_pvt *	p,
		int	format,
		int	sample_rate,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug
	)			[static]

Add RFC 2833 DTMF offer to SDP.

Definition at line 8536 of file chan_sip.c.

References AST_RTP_DTMF, ast_rtp_lookup_code(), ast_rtp_lookup_mime_subtype(), ast_str_append(), ast_verbose, and sip_pvt::rtp.

Referenced by add_sdp().

{
   int rtp_code;

   if (debug)
      ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format, 0));
   if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
          ast_rtp_lookup_mime_subtype(0, format, 0),
          sample_rate);
   if (format == AST_RTP_DTMF)   /* Indicate we support DTMF and FLASH... */
      ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
}

static void add_peer_mailboxes	(	struct sip_peer *	peer,
		const char *	value
	)			[static]

Todo:

document this function

Definition at line 21547 of file chan_sip.c.

References ast_calloc, ast_free, AST_LIST_INSERT_TAIL, ast_strdup, ast_strdupa, ast_strlen_zero(), context, sip_mailbox::entry, mailbox, sip_peer::mailboxes, mbox(), and strsep().

Referenced by build_peer().

{
   char *next, *mbox, *context;

   next = ast_strdupa(value);

   while ((mbox = context = strsep(&next, ","))) {
      struct sip_mailbox *mailbox;

      if (!(mailbox = ast_calloc(1, sizeof(*mailbox))))
         continue;

      strsep(&context, "@");
      if (ast_strlen_zero(mbox)) {
         ast_free(mailbox);
         continue;
      }
      mailbox->mailbox = ast_strdup(mbox);
      mailbox->context = ast_strdup(context);

      AST_LIST_INSERT_TAIL(&peer->mailboxes, mailbox, entry);
   }
}

static void add_peer_mwi_subs

(

struct sip_peer *

peer

)

[static]

Definition at line 19166 of file chan_sip.c.

References AST_EVENT_IE_CONTEXT, AST_EVENT_IE_END, AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, AST_EVENT_MWI, ast_event_subscribe(), AST_LIST_TRAVERSE, sip_mailbox::entry, mailbox, sip_peer::mailboxes, mwi_event_cb(), and S_OR.

Referenced by build_peer(), and handle_request_subscribe().

{
   struct sip_mailbox *mailbox;

   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      mailbox->event_sub = ast_event_subscribe(AST_EVENT_MWI, mwi_event_cb, peer,
         AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox->mailbox,
         AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, S_OR(mailbox->context, "default"),
         AST_EVENT_IE_END);
   }
}

static struct sip_auth * add_realm_authentication	(	struct sip_auth *	authlist,
		const char *	configuration,
		int	lineno
	)			[static]

Add realm authentication in list.

Definition at line 21369 of file chan_sip.c.

References ast_calloc, ast_copy_string(), ast_debug, ast_log(), ast_strlen_zero(), ast_verb, LOG_WARNING, sip_auth::md5secret, sip_auth::next, sip_auth::realm, secret, strsep(), and sip_auth::username.

Referenced by build_peer().

{
   char authcopy[256];
   char *username=NULL, *realm=NULL, *secret=NULL, *md5secret=NULL;
   char *stringp;
   struct sip_auth *a, *b, *auth;

   if (ast_strlen_zero(configuration))
      return authlist;

   ast_debug(1, "Auth config ::  %s\n", configuration);

   ast_copy_string(authcopy, configuration, sizeof(authcopy));
   stringp = authcopy;

   username = stringp;
   realm = strrchr(stringp, '@');
   if (realm)
      *realm++ = '\0';
   if (ast_strlen_zero(username) || ast_strlen_zero(realm)) {
      ast_log(LOG_WARNING, "Format for authentication entry is user[:secret]@realm at line %d\n", lineno);
      return authlist;
   }
   stringp = username;
   username = strsep(&stringp, ":");
   if (username) {
      secret = strsep(&stringp, ":");
      if (!secret) {
         stringp = username;
         md5secret = strsep(&stringp, "#");
      }
   }
   if (!(auth = ast_calloc(1, sizeof(*auth))))
      return authlist;

   ast_copy_string(auth->realm, realm, sizeof(auth->realm));
   ast_copy_string(auth->username, username, sizeof(auth->username));
   if (secret)
      ast_copy_string(auth->secret, secret, sizeof(auth->secret));
   if (md5secret)
      ast_copy_string(auth->md5secret, md5secret, sizeof(auth->md5secret));

   /* find the end of the list */
   for (b = NULL, a = authlist; a ; b = a, a = a->next)
      ;
   if (b)
      b->next = auth;   /* Add structure add end of list */
   else
      authlist = auth;

   ast_verb(3, "Added authentication for realm %s\n", realm);

   return authlist;

}

static void add_route	(	struct sip_request *	req,
		struct sip_route *	route
	)			[static]

Add route header into request per learned route.

Definition at line 7782 of file chan_sip.c.

References add_header(), ast_copy_string(), sip_route::hop, sip_route::next, and SIPBUFSIZE.

Referenced by reqprep().

{
   char r[SIPBUFSIZE*2], *p;
   int n, rem = sizeof(r);

   if (!route)
      return;

   p = r;
   for (;route ; route = route->next) {
      n = strlen(route->hop);
      if (rem < n+3) /* we need room for ",<route>" */
         break;
      if (p != r) {  /* add a separator after fist route */
         *p++ = ',';
         --rem;
      }
      *p++ = '<';
      ast_copy_string(p, route->hop, rem); /* cannot fail */
      p += n;
      *p++ = '>';
      rem -= (n+2);
   }
   *p = '\0';
   add_header(req, "Route", r);
}

static enum sip_result add_sdp	(	struct sip_request *	resp,
		struct sip_pvt *	p,
		int	oldsdp,
		int	add_audio,
		int	add_t38
	)			[static]

Add Session Description Protocol message.

If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism is used in Session-Timers where RE-INVITEs are used for refreshing SIP sessions without modifying the media session in any way.

Definition at line 8604 of file chan_sip.c.

References add_codec_to_sdp(), add_header(), add_header_contentLength(), add_line(), add_noncodec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ast_codec_pref_index(), ast_debug, AST_FAILURE, AST_FORMAT_AUDIO_MASK, AST_FORMAT_TEXT_MASK, AST_FORMAT_VIDEO_MASK, ast_getformatname_multiple(), ast_inet_ntoa(), ast_internal_timing_enabled(), ast_log(), ast_random(), AST_RTP_MAX, ast_str_alloca, ast_str_append(), ast_strlen_zero(), AST_SUCCESS, ast_test_flag, ast_udptl_get_local_max_datagram(), ast_udptl_get_us(), ast_verbose, buf, t38properties::capability, sip_request::debug, FALSE, sip_pvt::flags, get_our_media_address(), t38properties::jointcapability, sip_pvt::jointcapability, sip_pvt::jointnoncodeccapability, sip_pvt::lastrtprx, sip_pvt::lastrtptx, ast_str::len, sip_request::len, LOG_WARNING, sip_pvt::maxcallbitrate, sip_pvt::notext, sip_pvt::novideo, sip_pvt::ourip, sip_pvt::owner, t38properties::peercapability, sip_pvt::prefcodec, sip_pvt::prefs, sip_pvt::rtp, SDP_SAMPLE_RATE, sip_pvt::sessionid, sip_pvt::sessionversion, sip_debug_test_pvt(), SIP_PAGE2_CALL_ONHOLD, SIP_PAGE2_CALL_ONHOLD_INACTIVE, SIP_PAGE2_CALL_ONHOLD_ONEDIR, SIP_PAGE2_T38SUPPORT_RTP, SIPBUFSIZE, ast_str::str, sip_pvt::t38, t38_get_rate(), T38FAX_FILL_BIT_REMOVAL, T38FAX_RATE_MANAGEMENT_LOCAL_TCF, T38FAX_TRANSCODING_JBIG, T38FAX_TRANSCODING_MMR, T38FAX_UDP_EC_NONE, T38FAX_UDP_EC_REDUNDANCY, T38FAX_VERSION, T38FAX_VERSION_0, T38FAX_VERSION_1, t38properties::t38support, sip_pvt::tredirip, sip_pvt::trtp, TRUE, sip_pvt::udptl, sip_pvt::udptlredirip, ast_str::used, version, and sip_pvt::vrtp.

{
   int len = 0;
   int alreadysent = 0;

   struct sockaddr_in sin;
   struct sockaddr_in vsin;
   struct sockaddr_in tsin;
   struct sockaddr_in dest;
   struct sockaddr_in udptlsin;
   struct sockaddr_in vdest = { 0, };
   struct sockaddr_in tdest = { 0, };
   struct sockaddr_in udptldest = { 0, };

   /* SDP fields */
   char *version =   "v=0\r\n";     /* Protocol version */
   char subject[256];            /* Subject of the session */
   char owner[256];           /* Session owner/creator */
   char connection[256];            /* Connection data */
   char *session_time = "t=0 0\r\n";         /* Time the session is active */
   char bandwidth[256] = "";        /* Max bitrate */
   char *hold = "";
   struct ast_str *m_audio = ast_str_alloca(256);  /* Media declaration line for audio */
   struct ast_str *m_video = ast_str_alloca(256);  /* Media declaration line for video */
   struct ast_str *m_text = ast_str_alloca(256);   /* Media declaration line for text */
   struct ast_str *m_modem = ast_str_alloca(256);  /* Media declaration line for modem */
   struct ast_str *a_audio = ast_str_alloca(1024); /* Attributes for audio */
   struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */
   struct ast_str *a_text = ast_str_alloca(1024);  /* Attributes for text */
   struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */

   int x;
   int capability = 0;
   int needaudio = FALSE;
   int needvideo = FALSE;
   int needtext = FALSE;
   int debug = sip_debug_test_pvt(p);
   int min_audio_packet_size = 0;
   int min_video_packet_size = 0;
   int min_text_packet_size = 0;

   char codecbuf[SIPBUFSIZE];
   char buf[SIPBUFSIZE];

   /* Set the SDP session name */
   snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);

   if (!p->rtp) {
      ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
      return AST_FAILURE;
   }
   /* XXX We should not change properties in the SIP dialog until 
      we have acceptance of the offer if this is a re-invite */

   /* Set RTP Session ID and version */
   if (!p->sessionid) {
      p->sessionid = (int)ast_random();
      p->sessionversion = p->sessionid;
   } else {
      if (oldsdp == FALSE)
         p->sessionversion++;
   }

   get_our_media_address(p, needvideo, &sin, &vsin, &tsin, &dest, &vdest);

   snprintf(owner, sizeof(owner), "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner, p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
   snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));

   if (add_audio) {
      capability = p->jointcapability;

      /* XXX note, Video and Text are negated - 'true' means 'no' */
      ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), 
           p->novideo ? "True" : "False", p->notext ? "True" : "False");
      ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
   
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
      if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP)) {
         ast_str_append(&m_audio, 0, " %d", 191);
         ast_str_append(&a_audio, 0, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000);
      }
#endif

      /* Check if we need audio */
      if (capability & AST_FORMAT_AUDIO_MASK)
         needaudio = TRUE;

      /* Check if we need video in this call */
      if ((capability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
         if (p->vrtp) {
            needvideo = TRUE;
            ast_debug(2, "This call needs video offers!\n");
         } else
            ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
      }

      if (debug) 
         ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(sin.sin_port)); 

      /* Ok, we need video. Let's add what we need for video and set codecs.
         Video is handled differently than audio since we can not transcode. */
      if (needvideo) {
         ast_str_append(&m_video, 0, "m=video %d RTP/AVP", ntohs(vdest.sin_port));

         /* Build max bitrate string */
         if (p->maxcallbitrate)
            snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate);
         if (debug) 
            ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(vsin.sin_port));   
      }

      /* Check if we need text in this call */
      if((capability & AST_FORMAT_TEXT_MASK) && !p->notext) {
         if (sipdebug_text)
            ast_verbose("We think we can do text\n");
         if (p->trtp) {
            if (sipdebug_text)
               ast_verbose("And we have a text rtp object\n");
            needtext = TRUE;
            ast_debug(2, "This call needs text offers! \n");
         } else
            ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n");
      }
      
      /* Ok, we need text. Let's add what we need for text and set codecs.
         Text is handled differently than audio since we can not transcode. */
      if (needtext) {
         if (sipdebug_text)
            ast_verbose("Lets set up the text sdp\n");
         /* Determine text destination */
         if (p->tredirip.sin_addr.s_addr) {
            tdest.sin_addr = p->tredirip.sin_addr;
            tdest.sin_port = p->tredirip.sin_port;
         } else {
            tdest.sin_addr = p->ourip.sin_addr;
            tdest.sin_port = tsin.sin_port;
         }
         ast_str_append(&m_text, 0, "m=text %d RTP/AVP", ntohs(tdest.sin_port));

         if (debug) /* XXX should I use tdest below ? */
            ast_verbose("Text is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(tsin.sin_port)); 

      }

      /* Start building generic SDP headers */

      /* We break with the "recommendation" and send our IP, in order that our
         peer doesn't have to ast_gethostbyname() us */

      ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port));

      if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR)
         hold = "a=recvonly\r\n";
      else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE)
         hold = "a=inactive\r\n";
      else
         hold = "a=sendrecv\r\n";

      /* Now, start adding audio codecs. These are added in this order:
         - First what was requested by the calling channel
         - Then preferences in order from sip.conf device config for this peer/user
         - Then other codecs in capabilities, including video
      */

      /* Prefer the audio codec we were requested to use, first, no matter what 
         Note that p->prefcodec can include video codecs, so mask them out
      */
      if (capability & p->prefcodec) {
         int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;

         add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
                &m_audio, &a_audio,
                debug, &min_audio_packet_size);
         alreadysent |= codec;
      }

      /* Start by sending our preferred audio/video codecs */
      for (x = 0; x < 32; x++) {
         int codec;

         if (!(codec = ast_codec_pref_index(&p->prefs, x)))
            break; 

         if (!(capability & codec))
            continue;

         if (alreadysent & codec)
         continue;

         add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
                &m_audio, &a_audio,
                debug, &min_audio_packet_size);
         alreadysent |= codec;
      }

      /* Now send any other common audio and video codecs, and non-codec formats: */
      for (x = 1; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
         if (!(capability & x))  /* Codec not requested */
            continue;

         if (alreadysent & x) /* Already added to SDP */
            continue;

         if (x & AST_FORMAT_AUDIO_MASK)
            add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
                   &m_audio, &a_audio, debug, &min_audio_packet_size);
         else if (x & AST_FORMAT_VIDEO_MASK) 
            add_vcodec_to_sdp(p, x, 90000,
                    &m_video, &a_video, debug, &min_video_packet_size);
         else if (x & AST_FORMAT_TEXT_MASK)
            add_tcodec_to_sdp(p, x, 1000,
                    &m_text, &a_text, debug, &min_text_packet_size);
      }

      /* Now add DTMF RFC2833 telephony-event as a codec */
      for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
         if (!(p->jointnoncodeccapability & x))
            continue;

         add_noncodec_to_sdp(p, x, 8000, &m_audio, &a_audio, debug);
      }

      ast_debug(3, "-- Done with adding codecs to SDP\n");

      if (!p->owner || !ast_internal_timing_enabled(p->owner))
         ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");

      if (min_audio_packet_size)
         ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);

      /* XXX don't think you can have ptime for video */
      if (min_video_packet_size)
         ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);

      /* XXX don't think you can have ptime for text */
      if (min_text_packet_size)
         ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size);
   }

   if (add_t38) {
      ast_udptl_get_us(p->udptl, &udptlsin);

      /* Determine T.38 UDPTL destination */
      if (p->udptlredirip.sin_addr.s_addr) {
         udptldest.sin_port = p->udptlredirip.sin_port;
         udptldest.sin_addr = p->udptlredirip.sin_addr;
      } else {
         udptldest.sin_addr = p->ourip.sin_addr;
         udptldest.sin_port = udptlsin.sin_port;
      }

      if (debug)
         ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(udptlsin.sin_port));

      /* We break with the "recommendation" and send our IP, in order that our
         peer doesn't have to ast_gethostbyname() us */

      if (debug) {
         ast_debug(1, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n",
              p->t38.capability,
              p->t38.peercapability,
              p->t38.jointcapability);
      }

      ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n", ntohs(udptldest.sin_port));

      if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0)
         ast_str_append(&a_modem, 0, "a=T38FaxVersion:0\r\n");
      if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_1)
         ast_str_append(&a_modem, 0, "a=T38FaxVersion:1\r\n");
      if ((x = t38_get_rate(p->t38.jointcapability)))
         ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n", x);
      if ((p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) == T38FAX_FILL_BIT_REMOVAL)
         ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval\r\n");
      if ((p->t38.jointcapability & T38FAX_TRANSCODING_MMR) == T38FAX_TRANSCODING_MMR)
         ast_str_append(&a_modem, 0, "a=T38FaxTranscodingMMR\r\n");
      if ((p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) == T38FAX_TRANSCODING_JBIG)
         ast_str_append(&a_modem, 0, "a=T38FaxTranscodingJBIG\r\n");
      ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:%s\r\n", (p->t38.jointcapability & T38FAX_RATE_MANAGEMENT_LOCAL_TCF) ? "localTCF" : "transferredTCF");
      x = ast_udptl_get_local_max_datagram(p->udptl);
      ast_str_append(&a_modem, 0, "a=T38FaxMaxBuffer:%d\r\n", x);
      ast_str_append(&a_modem, 0, "a=T38FaxMaxDatagram:%d\r\n", x);
      if (p->t38.jointcapability != T38FAX_UDP_EC_NONE)
         ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC");
   }

   if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used < 2 ||
       m_text->len - m_text->used < 2 || a_text->len - a_text->used < 2 ||
       a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2)
      ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");

   if (needaudio)
      ast_str_append(&m_audio, 0, "\r\n");
   if (needvideo)
      ast_str_append(&m_video, 0, "\r\n");
   if (needtext)
      ast_str_append(&m_text, 0, "\r\n");

   len = strlen(version) + strlen(subject) + strlen(owner) +
      strlen(connection) + strlen(session_time);
   if (needaudio)
      len += m_audio->used + a_audio->used + strlen(hold);
   if (needvideo) /* only if video response is appropriate */
      len += m_video->used + a_video->used + strlen(bandwidth) + strlen(hold);
   if (needtext) /* only if text response is appropriate */
      len += m_text->used + a_text->used + strlen(hold);
   if (add_t38)
      len += m_modem->used + a_modem->used;

   add_header(resp, "Content-Type", "application/sdp");
   add_header_contentLength(resp, len);
   add_line(resp, version);
   add_line(resp, owner);
   add_line(resp, subject);
   add_line(resp, connection);
   if (needvideo)    /* only if video response is appropriate */
      add_line(resp, bandwidth);
   add_line(resp, session_time);
   if (needaudio) {
      add_line(resp, m_audio->str);
      add_line(resp, a_audio->str);
      add_line(resp, hold);
   }
   if (needvideo) { /* only if video response is appropriate */
      add_line(resp, m_video->str);
      add_line(resp, a_video->str);
      add_line(resp, hold);   /* Repeat hold for the video stream */
   }
   if (needtext) { /* only if text response is appropriate */
      add_line(resp, m_text->str);
      add_line(resp, a_text->str);
      add_line(resp, hold);   /* Repeat hold for the text stream */
   }
   if (add_t38) {
      add_line(resp, m_modem->str);
      add_line(resp, a_modem->str);
   }

   /* Update lastrtprx when we send our SDP */
   p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */

   ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, capability));

   return AST_SUCCESS;
}

static int add_sip_domain	(	const char *	domain,
		const enum domain_mode	mode,
		const char *	context
	)			[static]

Add SIP domain to list of domains we are responsible for.

Definition at line 21305 of file chan_sip.c.

References ast_calloc, ast_copy_string(), ast_debug, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_UNLOCK, ast_log(), ast_strlen_zero(), domain::list, and LOG_WARNING.

{
   struct domain *d;

   if (ast_strlen_zero(domain)) {
      ast_log(LOG_WARNING, "Zero length domain.\n");
      return 1;
   }

   if (!(d = ast_calloc(1, sizeof(*d))))
      return 0;

   ast_copy_string(d->domain, domain, sizeof(d->domain));

   if (!ast_strlen_zero(context))
      ast_copy_string(d->context, context, sizeof(d->context));

   d->mode = mode;

   AST_LIST_LOCK(&domain_list);
   AST_LIST_INSERT_TAIL(&domain_list, d, list);
   AST_LIST_UNLOCK(&domain_list);

   if (sipdebug)  
      ast_debug(1, "Added local SIP domain '%s'\n", domain);

   return 1;
}

static void add_tcodec_to_sdp	(	const struct sip_pvt *	p,
		int	codec,
		int	sample_rate,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)			[static]

Add text codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 8476 of file chan_sip.c.

References AST_FORMAT_T140, AST_FORMAT_T140RED, ast_getformatname(), ast_rtp_lookup_code(), ast_rtp_lookup_mime_subtype(), ast_str_append(), ast_verbose, and sip_pvt::trtp.

Referenced by add_sdp().

{
   int rtp_code;

   if (!p->trtp)
      return;

   if (debug)
      ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));

   if ((rtp_code = ast_rtp_lookup_code(p->trtp, 1, codec)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
          ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
   /* Add fmtp code here */

   if (codec == AST_FORMAT_T140RED) {
      ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code, 
          ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
          ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
          ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140));

   }
}

static int add_text	(	struct sip_request *	req,
		const char *	text
	)			[static]

Add text body to SIP message.

Definition at line 8348 of file chan_sip.c.

References add_header(), add_header_contentLength(), and add_line().

Referenced by transmit_message_with_text().

{
   /* XXX Convert \n's to \r\n's XXX */
   add_header(req, "Content-Type", "text/plain;charset=UTF-8");
   add_header_contentLength(req, strlen(text));
   add_line(req, text);
   return 0;
}

static struct ast_variable* add_var	(	const char *	buf,
		struct ast_variable *	list
	)			[static]

implement the setvar config line

Definition at line 21456 of file chan_sip.c.

References ast_strdupa, ast_variable_new(), and ast_variable::next.

Referenced by build_device(), and build_peer().

{
   struct ast_variable *tmpvar = NULL;
   char *varname = ast_strdupa(buf), *varval = NULL;
   
   if ((varval = strchr(varname, '='))) {
      *varval++ = '\0';
      if ((tmpvar = ast_variable_new(varname, varval, ""))) {
         tmpvar->next = list;
         list = tmpvar;
      }
   }
   return list;
}

static void add_vcodec_to_sdp	(	const struct sip_pvt *	p,
		int	codec,
		int	sample_rate,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)			[static]

Add video codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 8454 of file chan_sip.c.

References ast_getformatname(), ast_rtp_lookup_code(), ast_rtp_lookup_mime_subtype(), ast_str_append(), ast_verbose, and sip_pvt::vrtp.

Referenced by add_sdp().

{
   int rtp_code;

   if (!p->vrtp)
      return;

   if (debug)
      ast_verbose("Adding video codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));

   if ((rtp_code = ast_rtp_lookup_code(p->vrtp, 1, codec)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
          ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
   /* Add fmtp code here */
}

static int add_vidupdate

(

struct sip_request *

req

)

[static]

add XML encoded media control with update

Note:

XML: The only way to turn 0 bits of information into a few hundred. (markster)

Definition at line 8391 of file chan_sip.c.

References add_header(), add_header_contentLength(), and add_line().

Referenced by transmit_info_with_vidupdate().

{
   const char *xml_is_a_huge_waste_of_space =
      "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
      " <media_control>\r\n"
      "  <vc_primitive>\r\n"
      "   <to_encoder>\r\n"
      "    <picture_fast_update>\r\n"
      "    </picture_fast_update>\r\n"
      "   </to_encoder>\r\n"
      "  </vc_primitive>\r\n"
      " </media_control>\r\n";
   add_header(req, "Content-Type", "application/media_control+xml");
   add_header_contentLength(req, strlen(xml_is_a_huge_waste_of_space));
   add_line(req, xml_is_a_huge_waste_of_space);
   return 0;
}

static void append_date

(

struct sip_request *

req

)

[static]

Append date to SIP message.

Definition at line 8295 of file chan_sip.c.

References add_header().

Referenced by build_csv_record(), transmit_invite(), transmit_response_with_date(), transmit_response_with_minse(), and transmit_response_with_unsupported().

{
   char tmpdat[256];
   struct tm tm;
   time_t t = time(NULL);

   gmtime_r(&t, &tm);
   strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T GMT", &tm);
   add_header(req, "Date", tmpdat);
}

static void append_history_full	(	struct sip_pvt *	p,
		const char *	fmt,
			...
	)			[static]

Append to SIP dialog history with arg list.

Definition at line 3004 of file chan_sip.c.

References append_history_va(), and sip_pvt::do_history.

{
   va_list ap;

   if (!p)
      return;

   if (!p->do_history && !recordhistory && !dumphistory)
      return;

   va_start(ap, fmt);
   append_history_va(p, fmt, ap);
   va_end(ap);

   return;
}

static void append_history_va	(	struct sip_pvt *	p,
		const char *	fmt,
		va_list	ap
	)			[static]

Append to SIP dialog history with arg list.

Definition at line 2977 of file chan_sip.c.

References ast_calloc, ast_free, AST_LIST_INSERT_TAIL, AST_LIST_REMOVE_HEAD, buf, sip_history::list, MAX_HISTORY_ENTRIES, and strsep().

Referenced by append_history_full().

{
   char buf[80], *c = buf; /* max history length */
   struct sip_history *hist;
   int l;

   vsnprintf(buf, sizeof(buf), fmt, ap);
   strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
   l = strlen(buf) + 1;
   if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
      return;
   if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
      ast_free(hist);
      return;
   }
   memcpy(hist->event, buf, l);
   if (p->history_entries == MAX_HISTORY_ENTRIES) {
      struct sip_history *oldest;
      oldest = AST_LIST_REMOVE_HEAD(p->history, list);
      p->history_entries--;
      ast_free(oldest);
   }
   AST_LIST_INSERT_TAIL(p->history, hist, list);
   p->history_entries++;
}

static void ast_quiet_chan

(

struct ast_channel *

chan

)

[static]

Turn off generator data XXX Does this function belong in the SIP channel?

Definition at line 17075 of file chan_sip.c.

References ast_channel::_state, ast_deactivate_generator(), AST_FLAG_MOH, ast_moh_stop(), AST_STATE_UP, ast_test_flag, and chan.

Referenced by attempt_transfer(), and handle_invite_replaces().

{
   if (chan && chan->_state == AST_STATE_UP) {
      if (ast_test_flag(chan, AST_FLAG_MOH))
         ast_moh_stop(chan);
      else if (chan->generatordata)
         ast_deactivate_generator(chan);
   }
}

static void ast_sip_ouraddrfor	(	struct in_addr *	them,
		struct sockaddr_in *	us
	)			[static]

NAT fix - decide which IP address to use for Asterisk server?

Using the localaddr structure built up with localnet statements in sip.conf apply it to their address to see if we need to substitute our externip or can get away with our internal bindaddr 'us' is always overwritten.

Definition at line 2926 of file chan_sip.c.

References ast_apply_ha(), ast_debug, ast_inet_ntoa(), ast_log(), ast_ouraddrfor(), ast_parse_arg(), AST_SENSE_ALLOW, ast_stun_request(), externexpire, externhost, externrefresh, localaddr, LOG_NOTICE, LOG_WARNING, PARSE_INADDR, and sipsock.

Referenced by manager_sipnotify(), sip_alloc(), sip_cli_notify(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), transmit_register(), and transmit_response_using_temp().

{
   struct sockaddr_in theirs;
   /* Set want_remap to non-zero if we want to remap 'us' to an externally
    * reachable IP address and port. This is done if:
    * 1. we have a localaddr list (containing 'internal' addresses marked
    *    as 'deny', so ast_apply_ha() will return AST_SENSE_DENY on them,
    *    and AST_SENSE_ALLOW on 'external' ones);
    * 2. either stunaddr or externip is set, so we know what to use as the
    *    externally visible address;
    * 3. the remote address, 'them', is external;
    * 4. the address returned by ast_ouraddrfor() is 'internal' (AST_SENSE_DENY
    *    when passed to ast_apply_ha() so it does need to be remapped.
    *    This fourth condition is checked later.
    */
   int want_remap;

   *us = internip;      /* starting guess for the internal address */
   /* now ask the system what would it use to talk to 'them' */
   ast_ouraddrfor(them, &us->sin_addr);
   theirs.sin_addr = *them;

   want_remap = localaddr &&
      (externip.sin_addr.s_addr || stunaddr.sin_addr.s_addr) &&
      ast_apply_ha(localaddr, &theirs) == AST_SENSE_ALLOW ;

   if (want_remap &&
       (!global_matchexterniplocally || !ast_apply_ha(localaddr, us)) ) {
      /* if we used externhost or stun, see if it is time to refresh the info */
      if (externexpire && time(NULL) >= externexpire) {
         if (stunaddr.sin_addr.s_addr) {
            ast_stun_request(sipsock, &stunaddr, NULL, &externip);
         } else {
            if (ast_parse_arg(externhost, PARSE_INADDR, &externip))
               ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
         }
         externexpire = time(NULL) + externrefresh;
      }
      if (externip.sin_addr.s_addr)
         *us = externip;
      else
         ast_log(LOG_WARNING, "stun failed\n");
      ast_debug(1, "Target address %s is not local, substituting externip\n", 
         ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
   } else if (bindaddr.sin_addr.s_addr) {
      /* no remapping, but we bind to a specific address, so use it. */
      *us = bindaddr;
   }
}

static int attempt_transfer	(	struct sip_dual *	transferer,
		struct sip_dual *	target
	)			[static]

Attempt transfer of SIP call This fix for attended transfers on a local PBX.

Definition at line 17087 of file chan_sip.c.

References ast_channel::_state, ast_channel_masquerade(), ast_debug, ast_log(), ast_quiet_chan(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), ast_state2str(), sip_dual::chan1, sip_dual::chan2, LOG_NOTICE, LOG_WARNING, and ast_channel::name.

{
   int res = 0;
   struct ast_channel *peera = NULL,   
      *peerb = NULL,
      *peerc = NULL,
      *peerd = NULL;


   /* We will try to connect the transferee with the target and hangup
      all channels to the transferer */   
   ast_debug(4, "Sip transfer:--------------------\n");
   if (transferer->chan1)
      ast_debug(4, "-- Transferer to PBX channel: %s State %s\n", transferer->chan1->name, ast_state2str(transferer->chan1->_state));
   else
      ast_debug(4, "-- No transferer first channel - odd??? \n");
   if (target->chan1)
      ast_debug(4, "-- Transferer to PBX second channel (target): %s State %s\n", target->chan1->name, ast_state2str(target->chan1->_state));
   else
      ast_debug(4, "-- No target first channel ---\n");
   if (transferer->chan2)
      ast_debug(4, "-- Bridged call to transferee: %s State %s\n", transferer->chan2->name, ast_state2str(transferer->chan2->_state));
   else
      ast_debug(4, "-- No bridged call to transferee\n");
   if (target->chan2)
      ast_debug(4, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? target->chan2->name : "<none>", target->chan2 ? ast_state2str(target->chan2->_state) : "(none)");
   else
      ast_debug(4, "-- No target second channel ---\n");
   ast_debug(4, "-- END Sip transfer:--------------------\n");
   if (transferer->chan2) { /* We have a bridge on the transferer's channel */
      peera = transferer->chan1; /* Transferer - PBX -> transferee channel * the one we hangup */
      peerb = target->chan1;     /* Transferer - PBX -> target channel - This will get lost in masq */
      peerc = transferer->chan2; /* Asterisk to Transferee */
      peerd = target->chan2;     /* Asterisk to Target */
      ast_debug(3, "SIP transfer: Four channels to handle\n");
   } else if (target->chan2) {   /* Transferer has no bridge (IVR), but transferee */
      peera = target->chan1;     /* Transferer to PBX -> target channel */
      peerb = transferer->chan1; /* Transferer to IVR*/
      peerc = target->chan2;     /* Asterisk to Target */
      peerd = transferer->chan2; /* Nothing */
      ast_debug(3, "SIP transfer: Three channels to handle\n");
   }

   if (peera && peerb && peerc && (peerb != peerc)) {
      ast_quiet_chan(peera);     /* Stop generators */
      ast_quiet_chan(peerb);  
      ast_quiet_chan(peerc);
      if (peerd)
         ast_quiet_chan(peerd);

      ast_debug(4, "SIP transfer: trying to masquerade %s into %s\n", peerc->name, peerb->name);
      if (ast_channel_masquerade(peerb, peerc)) {
         ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name, peerc->name);
         res = -1;
      } else
         ast_debug(4, "SIP transfer: Succeeded to masquerade channels.\n");
      return res;
   } else {
      ast_log(LOG_NOTICE, "SIP Transfer attempted with no appropriate bridged calls to transfer\n");
      if (transferer->chan1)
         ast_softhangup_nolock(transferer->chan1, AST_SOFTHANGUP_DEV);
      if (target->chan1)
         ast_softhangup_nolock(target->chan1, AST_SOFTHANGUP_DEV);
      return -1;
   }
   return 0;
}

static void auth_headers	(	enum sip_auth_type	code,
		char **	header,
		char **	respheader
	)			[static]

return the request and response heade for a 401 or 407 code

Definition at line 10330 of file chan_sip.c.

References ast_verbose, PROXY_AUTH, and WWW_AUTH.

Referenced by check_auth(), do_proxy_auth(), do_register_auth(), and transmit_request_with_auth().

{
   if (code == WWW_AUTH) {       /* 401 */
      *header = "WWW-Authenticate";
      *respheader = "Authorization";
   } else if (code == PROXY_AUTH) { /* 407 */
      *header = "Proxy-Authenticate";
      *respheader = "Proxy-Authorization";
   } else {
      ast_verbose("-- wrong response code %d\n", code);
      *header = *respheader = "Invalid";
   }
}

static int auto_congest

(

const void *

arg

)

[static]

Scheduled congestion on a call. Only called by the scheduler, must return the reference when done.

Definition at line 4544 of file chan_sip.c.

References append_history, ast_channel_trylock, ast_channel_unlock, AST_CONTROL_CONGESTION, ast_queue_control(), dialog_unref(), sip_pvt_lock, and sip_pvt_unlock.

{
   struct sip_pvt *p = (struct sip_pvt *)arg;

   sip_pvt_lock(p);
   p->initid = -1;   /* event gone, will not be rescheduled */
   if (p->owner) {
      /* XXX fails on possible deadlock */
      if (!ast_channel_trylock(p->owner)) {
         append_history(p, "Cong", "Auto-congesting (timer)");
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         ast_channel_unlock(p->owner);
      }
   }
   sip_pvt_unlock(p);
   dialog_unref(p, "unreffing arg passed into auto_congest callback (p->initid)");
   return 0;
}

static void build_callid_pvt

(

struct sip_pvt *

pvt

)

[static]

Build SIP Call-ID value for a non-REGISTER transaction.

Definition at line 6163 of file chan_sip.c.

References ast_inet_ntoa(), ast_string_field_build, buf, sip_pvt::callid, sip_pvt::fromdomain, generate_random_string(), sip_pvt::ourip, and S_OR.

Referenced by manager_sipnotify(), sip_alloc(), sip_cli_notify(), sip_poke_peer(), sip_request_call(), and sip_send_mwi_to_peer().

{
   char buf[33];

   const char *host = S_OR(pvt->fromdomain, ast_inet_ntoa(pvt->ourip.sin_addr));
   
   ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);

}

static void build_callid_registry	(	struct sip_registry *	reg,
		struct in_addr	ourip,
		const char *	fromdomain
	)			[static]

Build SIP Call-ID value for a REGISTER transaction.

Definition at line 6174 of file chan_sip.c.

References ast_inet_ntoa(), ast_string_field_build, buf, sip_pvt::callid, generate_random_string(), and S_OR.

Referenced by transmit_register().

{
   char buf[33];

   const char *host = S_OR(fromdomain, ast_inet_ntoa(ourip));

   ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
}

static void build_contact

(

struct sip_pvt *

p

)

[static]

Build contact header - the contact header we send out.

Definition at line 9140 of file chan_sip.c.

References ast_inet_ntoa(), ast_string_field_build, ast_strlen_zero(), sip_pvt::exten, get_transport(), sip_pvt::ourip, ourport, sip_standard_port(), SIP_TRANSPORT_UDP, sip_pvt::socket, and sip_socket::type.

Referenced by check_user_full(), handle_request_invite(), handle_request_options(), handle_request_subscribe(), initreqprep(), register_verify(), and transmit_register().

{

   int ourport = ntohs(p->ourip.sin_port);

   if (p->socket.type & SIP_TRANSPORT_UDP) {
      if (!sip_standard_port(p->socket.type, ourport))
         ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ourport);
      else
         ast_string_field_build(p, our_contact, "<sip:%s%s%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr));
   } else 
      ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d;transport=%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ourport, get_transport(p->socket.type));
}

static struct sip_peer * build_peer	(	const char *	name,
		struct ast_variable *	v,
		struct ast_variable *	alt,
		int	realtime
	)			[static]

Build peer from configuration (file or realtime static/dynamic).

< The first transport listed should be used for outgoing

Definition at line 21572 of file chan_sip.c.

References __set_address_from_contact(), sip_peer::accountcode, add_peer_mailboxes(), add_peer_mwi_subs(), add_realm_authentication(), add_var(), sip_peer::addr, sip_peer::allowtransfer, sip_peer::amaflags, ao2_t_alloc, ao2_t_find, asprintf, ast_append_ha(), ast_atomic_fetchadd_int(), ast_callerid_split(), ast_cdr_amaflags2int(), ast_copy_flags, ast_copy_string(), ast_debug, ast_dnsmgr_lookup(), ast_free, ast_free_ha(), ast_get_group(), ast_get_ip(), ast_get_time_t(), ast_inet_ntoa(), AST_LIST_EMPTY, ast_log(), ast_parse_allow_disallow(), ast_parse_caller_presentation(), AST_SCHED_DEL_UNREF, ast_set2_flag, ast_skip_blanks(), ast_str_alloca, ast_str_append(), ast_str_set(), ast_strdupa, ast_strlen_zero(), ast_test_flag, ast_true(), ast_variables_destroy(), sip_peer::auth, sip_peer::autoframing, sip_peer::busy_level, sip_peer::call_limit, sip_peer::callgroup, sip_peer::callingpres, sip_peer::capability, sip_peer::chanvars, sip_peer::cid_name, sip_peer::cid_num, clear_realm_authentication(), sip_peer::contactha, sip_peer::context, sip_peer::defaddr, DEFAULT_MAXMS, sip_peer::deprecated_username, destroy_association(), sip_peer::dnsmgr, errno, sip_peer::expire, FALSE, ast_flags::flags, sip_peer::flags, format, sip_peer::fromdomain, sip_peer::fromuser, sip_peer::fullcontact, get_transport(), global_contact_ha, global_flags, sip_peer::ha, handle_common_options(), sip_peer::host_dynamic, sip_settings::ignore_regexpire, inet_aton(), sip_peer::is_realtime, sip_peer::language, sip_peer::lastms, sip_peer::lastmsgssent, ast_variable::lineno, LOG_ERROR, LOG_NOTICE, LOG_WARNING, sip_peer::mailboxes, sip_peer::maxcallbitrate, MAXHOSTNAMELEN, sip_peer::maxms, sip_peer::md5secret, sip_peer::mohinterpret, sip_peer::mohsuggest, ast_variable::name, sip_peer::name, ast_variable::next, OBJ_POINTER, OBJ_UNLINK, sip_peer::outboundproxy, sip_peer::parkinglot, sip_peer::pickupgroup, sip_socket::port, sip_peer::prefs, proxy_allocate(), sip_peer::qualifyfreq, ref_peer(), reg_source_db(), sip_peer::regexten, sip_peer::rt_fromcontact, sip_peer::rtpholdtimeout, sip_peer::rtpkeepalive, sip_peer::rtptimeout, sched, sip_peer::secret, set_peer_defaults(), sip_cfg, sip_destroy_peer_fn(), SIP_PAGE2_ALLOWSUBSCRIBE, SIP_PAGE2_REGISTERTRYING, SIP_PAGE2_RTCACHEFRIENDS, SIP_PAGE2_SUBSCRIBEMWIONLY, sip_poke_peer(), sip_register(), sip_send_mwi_to_peer(), SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, SIP_TRANSPORT_UDP, SIP_TYPE_PEER, SIP_TYPE_USER, SIP_USEREQPHONE, sip_peer::socket, srvlookup, sip_st_cfg::st_max_se, sip_st_cfg::st_min_se, sip_st_cfg::st_mode_oper, sip_st_cfg::st_ref, STANDARD_SIP_PORT, STANDARD_TLS_PORT, sip_peer::stimer, ast_str::str, str2stmode(), str2strefresher(), strsep(), sip_peer::subscribecontext, sip_peer::the_mark, sip_peer::timer_b, sip_peer::timer_t1, sip_peer::tohost, TRANSFER_CLOSED, TRANSFER_OPENFORALL, sip_peer::transports, TRUE, sip_socket::type, sip_peer::type, unref_peer(), ast_str::used, sip_peer::username, ast_variable::value, and sip_peer::vmexten.

{
   struct sip_peer *peer = NULL;
   struct ast_ha *oldha = NULL;
   int found=0;
   int firstpass=1;
   int format=0;     /* Ama flags */
   time_t regseconds = 0;
   struct ast_flags peerflags[2] = {{(0)}};
   struct ast_flags mask[2] = {{(0)}};
   char callback[256] = "";
   struct sip_peer tmp_peer;
   const char *srvlookup = NULL;
   static int deprecation_warning = 1;
   struct ast_str *fullcontact = ast_str_alloca(512);

   if (!realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
      /* Note we do NOT use find_peer here, to avoid realtime recursion */
      /* We also use a case-sensitive comparison (unlike find_peer) so
         that case changes made to the peer name will be properly handled
         during reload
      */
      ast_copy_string(tmp_peer.name, name, sizeof(tmp_peer.name));
      peer = ao2_t_find(peers, &tmp_peer, OBJ_POINTER | OBJ_UNLINK, "find and unlink peer from peers table");
   }

   if (peer) {
      /* Already in the list, remove it and it will be added back (or FREE'd)  */
      found++;
      if (!(peer->the_mark))
         firstpass = 0;
   } else {
      if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct")))
         return NULL;

      if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
         ast_atomic_fetchadd_int(&rpeerobjs, 1);
         ast_debug(3, "-REALTIME- peer built. Name: %s. Peer objects: %d\n", name, rpeerobjs);
      } else
         ast_atomic_fetchadd_int(&speerobjs, 1);
   }

   /* Note that our peer HAS had its reference count increased */
   if (firstpass) {
      peer->lastmsgssent = -1;
      oldha = peer->ha;
      peer->ha = NULL;
      set_peer_defaults(peer);   /* Set peer defaults */
      peer->type = 0;
   }
   if (!found && name)
      ast_copy_string(peer->name, name, sizeof(peer->name));

   /* If we have channel variables, remove them (reload) */
   if (peer->chanvars) {
      ast_variables_destroy(peer->chanvars);
      peer->chanvars = NULL;
      /* XXX should unregister ? */
   }

   /* If we have realm authentication information, remove them (reload) */
   clear_realm_authentication(peer->auth);
   peer->auth = NULL;
   peer->transports = 0;
   peer->socket.type = 0;

   for (; v || ((v = alt) && !(alt=NULL)); v = v->next) {
      if (handle_common_options(&peerflags[0], &mask[0], v))
         continue;
      if (!strcasecmp(v->name, "transport") && !ast_strlen_zero(v->value)) {
         char *val = ast_strdupa(v->value);
         char *trans;

         while ((trans = strsep(&val, ","))) {
            trans = ast_skip_blanks(trans);

            if (!strncasecmp(trans, "udp", 3)) 
               peer->transports |= SIP_TRANSPORT_UDP;
            else if (!strncasecmp(trans, "tcp", 3))
               peer->transports |= SIP_TRANSPORT_TCP;
            else if (!strncasecmp(trans, "tls", 3))
               peer->transports |= SIP_TRANSPORT_TLS;
            else
               ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);

            if (!peer->socket.type) /*!< The first transport listed should be used for outgoing */
               peer->socket.type = peer->transports;
         }
      } else if (realtime && !strcasecmp(v->name, "regseconds")) {
         ast_get_time_t(v->value, &regseconds, 0, NULL);
      } else if (realtime && !strcasecmp(v->name, "lastms")) {
         sscanf(v->value, "%d", &peer->lastms);
      } else if (realtime && !strcasecmp(v->name, "ipaddr") && !ast_strlen_zero(v->value) ) {
         inet_aton(v->value, &(peer->addr.sin_addr));
      } else if (realtime && !strcasecmp(v->name, "name"))
         ast_copy_string(peer->name, v->value, sizeof(peer->name));
      else if (realtime && !strcasecmp(v->name, "fullcontact")) {
         /* Reconstruct field, because realtime separates our value at the ';' */
         if (fullcontact->used > 0) {
            ast_str_append(&fullcontact, 0, ";%s", v->value);
         } else {
            ast_str_set(&fullcontact, 0, "%s", v->value);
         }
      } else if (!strcasecmp(v->name, "type")) {
         if (!strcasecmp(v->value, "peer")) {
            peer->type |= SIP_TYPE_PEER;
         } else if (!strcasecmp(v->value, "user")) {
            peer->type |= SIP_TYPE_USER;
         } else if (!strcasecmp(v->value, "friend")) {
            peer->type = SIP_TYPE_USER | SIP_TYPE_PEER;
         }
      } else if (!strcasecmp(v->name, "secret")) 
         ast_copy_string(peer->secret, v->value, sizeof(peer->secret));
      else if (!strcasecmp(v->name, "md5secret")) 
         ast_copy_string(peer->md5secret, v->value, sizeof(peer->md5secret));
      else if (!strcasecmp(v->name, "auth"))
         peer->auth = add_realm_authentication(peer->auth, v->value, v->lineno);
      else if (!strcasecmp(v->name, "callerid")) {
         ast_callerid_split(v->value, peer->cid_name, sizeof(peer->cid_name), peer->cid_num, sizeof(peer->cid_num));
      } else if (!strcasecmp(v->name, "fullname")) {
         ast_copy_string(peer->cid_name, v->value, sizeof(peer->cid_name));
      } else if (!strcasecmp(v->name, "cid_number")) {
         ast_copy_string(peer->cid_num, v->value, sizeof(peer->cid_num));
      } else if (!strcasecmp(v->name, "context")) {
         ast_copy_string(peer->context, v->value, sizeof(peer->context));
      } else if (!strcasecmp(v->name, "subscribecontext")) {
         ast_copy_string(peer->subscribecontext, v->value, sizeof(peer->subscribecontext));
      } else if (!strcasecmp(v->name, "fromdomain")) {
         ast_copy_string(peer->fromdomain, v->value, sizeof(peer->fromdomain));
      } else if (!strcasecmp(v->name, "usereqphone")) {
         ast_set2_flag(&peer->flags[0], ast_true(v->value), SIP_USEREQPHONE);
      } else if (!strcasecmp(v->name, "fromuser")) {
         ast_copy_string(peer->fromuser, v->value, sizeof(peer->fromuser));
      } else if (!strcasecmp(v->name, "outboundproxy")) {
         char *port, *next, *force, *proxyname;
         int forceopt = FALSE;
         /* Set peer channel variable */
         next = proxyname = ast_strdupa(v->value);
         if ((port = strchr(proxyname, ':'))) {
            *port++ = '\0';
            next = port;
         }
         if ((force = strchr(next, ','))) {
            *force++ = '\0';
            forceopt = strcmp(force, "force");
         }
         /* Allocate proxy object */
         peer->outboundproxy = proxy_allocate(proxyname, port, forceopt);
      } else if (!strcasecmp(v->name, "host")) {
         if (!strcasecmp(v->value, "dynamic")) {
            /* They'll register with us */
            if (!found || !peer->host_dynamic) {
               /* Initialize stuff if this is a new peer, or if it used to
                * not be dynamic before the reload. */
               memset(&peer->addr.sin_addr, 0, 4);
               if (peer->addr.sin_port) {
                  /* If we've already got a port, make it the default rather than absolute */
                  peer->defaddr.sin_port = peer->addr.sin_port;
                  peer->addr.sin_port = 0;
               }
            }
            peer->host_dynamic = TRUE;
         } else {
            /* Non-dynamic.  Make sure we become that way if we're not */
            AST_SCHED_DEL_UNREF(sched, peer->expire,
                  unref_peer(peer, "removing register expire ref"));
            peer->host_dynamic = FALSE;
            srvlookup = v->value;
            if (global_dynamic_exclude_static) {
               int err = 0;
               global_contact_ha = ast_append_ha("deny", (char *)ast_inet_ntoa(peer->addr.sin_addr), global_contact_ha, &err);
               if (err) {
                  ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
               }
            }
         }
      } else if (!strcasecmp(v->name, "defaultip")) {
         if (ast_get_ip(&peer->defaddr, v->value)) {
            unref_peer(peer, "unref_peer: from build_peer defaultip");
            return NULL;
         }
      } else if (!strcasecmp(v->name, "permit") || !strcasecmp(v->name, "deny")) {
         int ha_error = 0;

         peer->ha = ast_append_ha(v->name, v->value, peer->ha, &ha_error);
         if (ha_error)
            ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
      } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) {
         int ha_error = 0;
         peer->contactha = ast_append_ha(v->name + 7, v->value, peer->contactha, &ha_error);
         if (ha_error) {
            ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
         }
      } else if (!strcasecmp(v->name, "port")) {
         if (!realtime && peer->host_dynamic)
            peer->defaddr.sin_port = htons(atoi(v->value));
         else
            peer->addr.sin_port = htons(atoi(v->value));
      } else if (!strcasecmp(v->name, "callingpres")) {
         peer->callingpres = ast_parse_caller_presentation(v->value);
         if (peer->callingpres == -1)
            peer->callingpres = atoi(v->value);
      } else if (!strcasecmp(v->name, "username") || !strcmp(v->name, "defaultuser")) {   /* "username" is deprecated */
         ast_copy_string(peer->username, v->value, sizeof(peer->username));
         if (!strcasecmp(v->name, "username")) {
            if (deprecation_warning) {
               ast_log(LOG_NOTICE, "The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'\n");
               deprecation_warning = 0;
            }
            peer->deprecated_username = 1;
         }
      } else if (!strcasecmp(v->name, "language")) {
         ast_copy_string(peer->language, v->value, sizeof(peer->language));
      } else if (!strcasecmp(v->name, "regexten")) {
         ast_copy_string(peer->regexten, v->value, sizeof(peer->regexten));
      } else if (!strcasecmp(v->name, "callbackextension")) {
         ast_copy_string(callback, v->value, sizeof(callback));
      } else if (!strcasecmp(v->name, "callcounter")) {
         peer->call_limit = ast_true(v->value) ? 999 : 0;
      } else if (!strcasecmp(v->name, "call-limit")) {
         peer->call_limit = atoi(v->value);
         if (peer->call_limit < 0)
            peer->call_limit = 0;
      } else if (!strcasecmp(v->name, "busylevel")) {
         peer->busy_level = atoi(v->value);
         if (peer->busy_level < 0)
            peer->busy_level = 0;
      } else if (!strcasecmp(v->name, "amaflags")) {
         format = ast_cdr_amaflags2int(v->value);
         if (format < 0) {
            ast_log(LOG_WARNING, "Invalid AMA Flags for peer: %s at line %d\n", v->value, v->lineno);
         } else {
            peer->amaflags = format;
         }
      } else if (!strcasecmp(v->name, "accountcode")) {
         ast_copy_string(peer->accountcode, v->value, sizeof(peer->accountcode));
      } else if (!strcasecmp(v->name, "mohinterpret")) {
         ast_copy_string(peer->mohinterpret, v->value, sizeof(peer->mohinterpret));
      } else if (!strcasecmp(v->name, "mohsuggest")) {
         ast_copy_string(peer->mohsuggest, v->value, sizeof(peer->mohsuggest));
      } else if (!strcasecmp(v->name, "parkinglot")) {
         ast_copy_string(peer->parkinglot, v->value, sizeof(peer->parkinglot));
      } else if (!strcasecmp(v->name, "mailbox")) {
         add_peer_mailboxes(peer, v->value);
      } else if (!strcasecmp(v->name, "hasvoicemail")) {
         /* People expect that if 'hasvoicemail' is set, that the mailbox will
          * be also set, even if not explicitly specified. */
         if (ast_true(v->value) && AST_LIST_EMPTY(&peer->mailboxes)) {
            add_peer_mailboxes(peer, name);
         }
      } else if (!strcasecmp(v->name, "subscribemwi")) {
         ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_SUBSCRIBEMWIONLY);
      } else if (!strcasecmp(v->name, "vmexten")) {
         ast_copy_string(peer->vmexten, v->value, sizeof(peer->vmexten));
      } else if (!strcasecmp(v->name, "callgroup")) {
         peer->callgroup = ast_get_group(v->value);
      } else if (!strcasecmp(v->name, "allowtransfer")) {
         peer->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
      } else if (!strcasecmp(v->name, "pickupgroup")) {
         peer->pickupgroup = ast_get_group(v->value);
      } else if (!strcasecmp(v->name, "allow")) {
         int error =  ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, TRUE);
         if (error)
            ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
      } else if (!strcasecmp(v->name, "disallow")) {
         int error =  ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, FALSE);
         if (error)
            ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
      } else if (!strcasecmp(v->name, "registertrying")) {
         ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_REGISTERTRYING);
      } else if (!strcasecmp(v->name, "autoframing")) {
         peer->autoframing = ast_true(v->value);
      } else if (!strcasecmp(v->name, "rtptimeout")) {
         if ((sscanf(v->value, "%d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
            peer->rtptimeout = global_rtptimeout;
         }
      } else if (!strcasecmp(v->name, "rtpholdtimeout")) {
         if ((sscanf(v->value, "%d", &peer->rtpholdtimeout) != 1) || (peer->rtpholdtimeout < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
            peer->rtpholdtimeout = global_rtpholdtimeout;
         }
      } else if (!strcasecmp(v->name, "rtpkeepalive")) {
         if ((sscanf(v->value, "%d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d.  Using default.\n", v->value, v->lineno);
            peer->rtpkeepalive = global_rtpkeepalive;
         }
      } else if (!strcasecmp(v->name, "timert1")) {
         if ((sscanf(v->value, "%d", &peer->timer_t1) != 1) || (peer->timer_t1 < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid T1 time at line %d.  Using default.\n", v->value, v->lineno);
            peer->timer_t1 = global_t1;
         }
         /* Note that Timer B is dependent upon T1 and MUST NOT be lower
          * than T1 * 64, according to RFC 3261, Section 17.1.1.2 */
         if (peer->timer_b < peer->timer_t1 * 64) {
            peer->timer_b = peer->timer_t1 * 64;
         }
      } else if (!strcasecmp(v->name, "timerb")) {
         if ((sscanf(v->value, "%d", &peer->timer_b) != 1) || (peer->timer_b < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d.  Using default.\n", v->value, v->lineno);
            peer->timer_b = global_timer_b;
         }
         if (peer->timer_b < peer->timer_t1 * 64) {
            static int warning = 0;
            if (warning++ % 20 == 0) {
               ast_log(LOG_WARNING, "Timer B has been set lower than recommended. (RFC 3261, 17.1.1.2)\n");
            }
         }
      } else if (!strcasecmp(v->name, "setvar")) {
         peer->chanvars = add_var(v->value, peer->chanvars);
      } else if (!strcasecmp(v->name, "qualify")) {
         if (!strcasecmp(v->value, "no")) {
            peer->maxms = 0;
         } else if (!strcasecmp(v->value, "yes")) {
            peer->maxms = default_qualify ? default_qualify : DEFAULT_MAXMS;
         } else if (sscanf(v->value, "%d", &peer->maxms) != 1) {
            ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
            peer->maxms = 0;
         }
         if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->maxms > 0) {
            /* This would otherwise cause a network storm, where the
             * qualify response refreshes the peer from the database,
             * which in turn causes another qualify to be sent, ad
             * infinitum. */
            ast_log(LOG_WARNING, "Qualify is incompatible with dynamic uncached realtime.  Please either turn rtcachefriends on or turn qualify off on peer '%s'\n", peer->name);
            peer->maxms = 0;
         }
      } else if (!strcasecmp(v->name, "qualifyfreq")) {
         int i;
         if (sscanf(v->value, "%d", &i) == 1)
            peer->qualifyfreq = i * 1000;  
         else {
            ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
            peer->qualifyfreq = global_qualifyfreq;
         }
      } else if (!strcasecmp(v->name, "maxcallbitrate")) {
         peer->maxcallbitrate = atoi(v->value);
         if (peer->maxcallbitrate < 0)
            peer->maxcallbitrate = default_maxcallbitrate;
      } else if (!strcasecmp(v->name, "session-timers")) {
         int i = (int) str2stmode(v->value); 
         if (i < 0) {
            ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
            peer->stimer.st_mode_oper = global_st_mode;
         } else {
            peer->stimer.st_mode_oper = i;
         }
      } else if (!strcasecmp(v->name, "session-expires")) {
         if (sscanf(v->value, "%d", &peer->stimer.st_max_se) != 1) {
            ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
            peer->stimer.st_max_se = global_max_se;
         } 
      } else if (!strcasecmp(v->name, "session-minse")) {
         if (sscanf(v->value, "%d", &peer->stimer.st_min_se) != 1) {
            ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
            peer->stimer.st_min_se = global_min_se;
         } 
         if (peer->stimer.st_min_se < 90) {
            ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < 90 secs\n", v->value, v->lineno, config);
            peer->stimer.st_min_se = global_min_se;
         } 
      } else if (!strcasecmp(v->name, "session-refresher")) {
         int i = (int) str2strefresher(v->value); 
         if (i < 0) {
            ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
            peer->stimer.st_ref = global_st_refresher;
         } else {
            peer->stimer.st_ref = i;
         }
      }
   }

   if (!peer->socket.type) {
      peer->transports  = SIP_TRANSPORT_UDP;
      peer->socket.type = SIP_TRANSPORT_UDP;
   }

   if (fullcontact->used > 0) {
      ast_copy_string(peer->fullcontact, fullcontact->str, sizeof(peer->fullcontact));
      peer->rt_fromcontact = TRUE;
      /* We have a hostname in the fullcontact, but if we don't have an
       * address listed on the entry (or if it's 'dynamic'), then we need to
       * parse the entry to obtain the IP address, so a dynamic host can be
       * contacted immediately after reload (as opposed to waiting for it to
       * register once again). */
      /* XXX May need to revisit the final argument; does the realtime DB store whether
       * the original contact was over TLS or not? XXX */
      __set_address_from_contact(fullcontact->str, &peer->addr, 0);
   }

   if (srvlookup && peer->dnsmgr == NULL) {
      char transport[MAXHOSTNAMELEN];
      char _srvlookup[MAXHOSTNAMELEN];
      char *params;

      ast_copy_string(_srvlookup, srvlookup, sizeof(_srvlookup));
      if ((params = strchr(_srvlookup, ';'))) {
         *params++ = '\0';
      }
      
      snprintf(transport, sizeof(transport), "_sip._%s", get_transport(peer->socket.type));

      if (ast_dnsmgr_lookup(_srvlookup, &peer->addr, &peer->dnsmgr, global_srvlookup ? transport : NULL)) {
         unref_peer(peer, "getting rid of a peer pointer");
         return NULL;
      }

      ast_copy_string(peer->tohost, srvlookup, sizeof(peer->tohost));
   }

   if (!peer->addr.sin_port)
      peer->addr.sin_port = htons(((peer->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT));

   if (!peer->socket.port)
      peer->socket.port = htons(((peer->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT));

   if (!sip_cfg.ignore_regexpire && peer->host_dynamic && realtime) {
      time_t nowtime = time(NULL);

      if ((nowtime - regseconds) > 0) {
         destroy_association(peer);
         memset(&peer->addr, 0, sizeof(peer->addr));
         peer->lastms = -1;
         ast_debug(1, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime);
      }
   }

   /* Startup regular pokes */
   if (realtime && peer->lastms > 0) {
      ref_peer(peer, "schedule qualify");
      sip_poke_peer(peer, 0);
   }

   ast_copy_flags(&peer->flags[0], &peerflags[0], mask[0].flags);
   ast_copy_flags(&peer->flags[1], &peerflags[1], mask[1].flags);
   if (ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE))
      global_allowsubscribe = TRUE; /* No global ban any more */
   if (!found && peer->host_dynamic && !peer->is_realtime)
      reg_source_db(peer);

   /* If they didn't request that MWI is sent *only* on subscribe, go ahead and
    * subscribe to it now. */
   if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY) && 
      !AST_LIST_EMPTY(&peer->mailboxes)) {
      add_peer_mwi_subs(peer);
      /* Send MWI from the event cache only.  This is so we can send initial
       * MWI if app_voicemail got loaded before chan_sip.  If it is the other
       * way, then we will get events when app_voicemail gets loaded. */
      sip_send_mwi_to_peer(peer, NULL, 1);
   }
   peer->the_mark = 0;

   ast_free_ha(oldha);
   if (!ast_strlen_zero(callback)) { /* build string from peer info */
      char *reg_string;

      if (asprintf(&reg_string, "%s:%s@%s/%s", peer->username, peer->secret, peer->tohost, callback) < 0) {
         ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
      } else   if (reg_string) {
         sip_register(reg_string, 0); /* XXX TODO: count in registry_count */
         ast_free(reg_string);
      }
   }
   return peer;
}

static int build_reply_digest	(	struct sip_pvt *	p,
		int	method,
		char *	digest,
		int	digest_len
	)			[static]

Build reply digest.

Returns:

Returns -1 if we have no auth

Note:

Build digest challenge for authentication of registrations and calls Also used for authentication of BYE

Definition at line 15362 of file chan_sip.c.

References append_history, ast_copy_string(), ast_debug, ast_inet_ntoa(), ast_log(), ast_md5_hash(), ast_random(), ast_strlen_zero(), authl, sip_pvt::authname, sip_pvt::callid, sip_pvt::domain, find_realm_authentication(), LOG_DEBUG, sip_auth::md5secret, sip_registry::md5secret, sip_pvt::nonce, sip_pvt::noncecount, sip_pvt::opaque, sip_registry::opaque, sip_pvt::peerauth, sip_pvt::peermd5secret, sip_pvt::peername, sip_pvt::peersecret, sip_pvt::qop, sip_pvt::realm, sip_pvt::sa, sip_auth::secret, secret, sip_methods, text, sip_pvt::uri, sip_auth::username, sip_pvt::username, and sip_registry::username.

Referenced by reply_digest(), transmit_register(), and transmit_request_with_auth().

{
   char a1[256];
   char a2[256];
   char a1_hash[256];
   char a2_hash[256];
   char resp[256];
   char resp_hash[256];
   char uri[256];
   char opaque[256] = "";
   char cnonce[80];
   const char *username;
   const char *secret;
   const char *md5secret;
   struct sip_auth *auth = NULL; /* Realm authentication */

   if (!ast_strlen_zero(p->domain))
      ast_copy_string(uri, p->domain, sizeof(uri));
   else if (!ast_strlen_zero(p->uri))
      ast_copy_string(uri, p->uri, sizeof(uri));
   else
      snprintf(uri, sizeof(uri), "sip:%s@%s", p->username, ast_inet_ntoa(p->sa.sin_addr));

   snprintf(cnonce, sizeof(cnonce), "%08lx", ast_random());

   /* Check if we have separate auth credentials */
   if(!(auth = find_realm_authentication(p->peerauth, p->realm))) /* Start with peer list */
      auth = find_realm_authentication(authl, p->realm); /* If not, global list */

   if (auth) {
      ast_log(LOG_DEBUG, "use realm [%s] from peer [%s][%s]\n", auth->username, p->peername, p->username);
      username = auth->username;
      secret = auth->secret;
      md5secret = auth->md5secret;
      if (sipdebug)
         ast_debug(1, "Using realm %s authentication for call %s\n", p->realm, p->callid);
   } else {
      /* No authentication, use peer or register= config */
      username = p->authname;
      secret =  p->peersecret;
      md5secret = p->peermd5secret;
   }
   if (ast_strlen_zero(username))   /* We have no authentication */
      return -1;

   /* Calculate SIP digest response */
   snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
   snprintf(a2, sizeof(a2), "%s:%s", sip_methods[method].text, uri);
   if (!ast_strlen_zero(md5secret))
      ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
   else
      ast_md5_hash(a1_hash, a1);
   ast_md5_hash(a2_hash, a2);

   p->noncecount++;
   if (!ast_strlen_zero(p->qop))
      snprintf(resp, sizeof(resp), "%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, p->noncecount, cnonce, "auth", a2_hash);
   else
      snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, p->nonce, a2_hash);
   ast_md5_hash(resp_hash, resp);

   /* only include the opaque string if it's set */
   if (!ast_strlen_zero(p->opaque)) {
     snprintf(opaque, sizeof(opaque), ", opaque=\"%s\"", p->opaque);
   }

   /* XXX We hard code our qop to "auth" for now.  XXX */
   if (!ast_strlen_zero(p->qop))
      snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s, qop=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, opaque, cnonce, p->noncecount);
   else
      snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s", username, p->realm, uri, p->nonce, resp_hash, opaque);

   append_history(p, "AuthResp", "Auth response sent for %s in realm %s - nc %d", username, p->realm, p->noncecount);

   return 0;
}

static void build_route	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	backwards
	)			[static]

Build route list from Record-Route header.

Definition at line 10780 of file chan_sip.c.

References __get_header(), ast_copy_string(), ast_debug, ast_malloc, ast_strlen_zero(), free_old_route(), get_header(), sip_route::hop, sip_request::len, list_route(), sip_route::next, sip_pvt::route, sip_pvt::route_persistant, and sip_debug_test_pvt().

Referenced by build_user_routes(), handle_request_invite(), and handle_response_invite().

{
   struct sip_route *thishop, *head, *tail;
   int start = 0;
   int len;
   const char *rr, *contact, *c;

   /* Once a persistant route is set, don't fool with it */
   if (p->route && p->route_persistant) {
      ast_debug(1, "build_route: Retaining previous route: <%s>\n", p->route->hop);
      return;
   }

   if (p->route) {
      free_old_route(p->route);
      p->route = NULL;
   }

   /* We only want to create the route set the first time this is called */
   p->route_persistant = 1;
   
   /* Build a tailq, then assign it to p->route when done.
    * If backwards, we add entries from the head so they end up
    * in reverse order. However, we do need to maintain a correct
    * tail pointer because the contact is always at the end.
    */
   head = NULL;
   tail = head;
   /* 1st we pass through all the hops in any Record-Route headers */
   for (;;) {
      /* Each Record-Route header */
      rr = __get_header(req, "Record-Route", &start);
      if (*rr == '\0')
         break;
      for (; (rr = strchr(rr, '<')) ; rr += len) { /* Each route entry */
         ++rr;
         len = strcspn(rr, ">") + 1;
         /* Make a struct route */
         if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
            /* ast_calloc is not needed because all fields are initialized in this block */
            ast_copy_string(thishop->hop, rr, len);
            ast_debug(2, "build_route: Record-Route hop: <%s>\n", thishop->hop);
            /* Link in */
            if (backwards) {
               /* Link in at head so they end up in reverse order */
               thishop->next = head;
               head = thishop;
               /* If this was the first then it'll be the tail */
               if (!tail)
                  tail = thishop;
            } else {
               thishop->next = NULL;
               /* Link in at the end */
               if (tail)
                  tail->next = thishop;
               else
                  head = thishop;
               tail = thishop;
            }
         }
      }
   }

   /* Only append the contact if we are dealing with a strict router */
   if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop, ";lr") == NULL) ) {
      /* 2nd append the Contact: if there is one */
      /* Can be multiple Contact headers, comma separated values - we just take the first */
      contact = get_header(req, "Contact");
      if (!ast_strlen_zero(contact)) {
         ast_debug(2, "build_route: Contact hop: %s\n", contact);
         /* Look for <: delimited address */
         c = strchr(contact, '<');
         if (c) {
            /* Take to > */
            ++c;
            len = strcspn(c, ">") + 1;
         } else {
            /* No <> - just take the lot */
            c = contact;
            len = strlen(contact) + 1;
         }
         if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
            /* ast_calloc is not needed because all fields are initialized in this block */
            ast_copy_string(thishop->hop, c, len);
            thishop->next = NULL;
            /* Goes at the end */
            if (tail)
               tail->next = thishop;
            else
               head = thishop;
         }
      }
   }

   /* Store as new route */
   p->route = head;

   /* For debugging dump what we ended up with */
   if (sip_debug_test_pvt(p))
      list_route(p->route);
}

static void build_rpid

(

struct sip_pvt *

p

)

[static]

Build the Remote Party-ID & From using callingpres options.

Definition at line 9155 of file chan_sip.c.

References ast_inet_ntoa(), ast_log(), AST_PRES_ALLOWED, AST_PRES_ALLOWED_NETWORK_NUMBER, AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN, AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, AST_PRES_NUMBER_NOT_AVAILABLE, AST_PRES_PROHIB_NETWORK_NUMBER, AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN, AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED, AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN, AST_PRES_RESTRICTION, ast_string_field_build, ast_string_field_set, ast_strlen_zero(), buf, sip_pvt::callingpres, ast_channel::cid, ast_callerid::cid_name, ast_callerid::cid_num, FALSE, sip_pvt::fromdomain, sip_pvt::fromuser, LOG_WARNING, sip_pvt::ourip, sip_pvt::owner, sip_pvt::rpid, sip_pvt::rpid_from, S_OR, sip_pvt::tag, and TRUE.

Referenced by initreqprep().

{
   int send_pres_tags = TRUE;
   const char *privacy=NULL;
   const char *screen=NULL;
   char buf[256];
   const char *clid = default_callerid;
   const char *clin = NULL;
   const char *fromdomain;

   if (!ast_strlen_zero(p->rpid) || !ast_strlen_zero(p->rpid_from))  
      return;

   if (p->owner && p->owner->cid.cid_num)
      clid = p->owner->cid.cid_num;
   if (p->owner && p->owner->cid.cid_name)
      clin = p->owner->cid.cid_name;
   if (ast_strlen_zero(clin))
      clin = clid;

   switch (p->callingpres) {
   case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED:
      privacy = "off";
      screen = "no";
      break;
   case AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN:
      privacy = "off";
      screen = "yes";
      break;
   case AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN:
      privacy = "off";
      screen = "no";
      break;
   case AST_PRES_ALLOWED_NETWORK_NUMBER:
      privacy = "off";
      screen = "yes";
      break;
   case AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED:
      privacy = "full";
      screen = "no";
      break;
   case AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN:
      privacy = "full";
      screen = "yes";
      break;
   case AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN:
      privacy = "full";
      screen = "no";
      break;
   case AST_PRES_PROHIB_NETWORK_NUMBER:
      privacy = "full";
      screen = "yes";
      break;
   case AST_PRES_NUMBER_NOT_AVAILABLE:
      send_pres_tags = FALSE;
      break;
   default:
      ast_log(LOG_WARNING, "Unsupported callingpres (%d)\n", p->callingpres);
      if ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED)
         privacy = "full";
      else
         privacy = "off";
      screen = "no";
      break;
   }
   
   fromdomain = S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr));

   snprintf(buf, sizeof(buf), "\"%s\" <sip:%s@%s>", clin, clid, fromdomain);
   if (send_pres_tags)
      snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), ";privacy=%s;screen=%s", privacy, screen);
   ast_string_field_set(p, rpid, buf);

   ast_string_field_build(p, rpid_from, "\"%s\" <sip:%s@%s>;tag=%s", clin,
                S_OR(p->fromuser, clid),
                fromdomain, p->tag);
}

static void build_via

(

struct sip_pvt *

p

)

[static]

Build a Via header for a request.

Definition at line 2907 of file chan_sip.c.

References ast_inet_ntoa(), ast_test_flag, sip_pvt::branch, sip_pvt::flags, get_transport_pvt(), sip_pvt::ourip, SIP_NAT, SIP_NAT_RFC3581, and sip_pvt::via.

Referenced by manager_sipnotify(), reqprep(), sip_alloc(), sip_cli_notify(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), transmit_invite(), transmit_register(), and transmit_response_using_temp().

{
   /* Work around buggy UNIDEN UIP200 firmware */
   const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";

   /* z9hG4bK is a magic cookie.  See RFC 3261 section 8.1.1.7 */
   snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s:%d;branch=z9hG4bK%08x%s",
       get_transport_pvt(p),
       ast_inet_ntoa(p->ourip.sin_addr),
       ntohs(p->ourip.sin_port), (int) p->branch, rport);
}

static int cb_extensionstate	(	char *	context,
		char *	exten,
		int	state,
		void *	data
	)			[static]

Callback for the devicestate notification (SUBSCRIBE) support subsystem.

Note:

If you add an "hint" priority to the extension in the dial plan, you will get notifications on device state changes

Definition at line 11095 of file chan_sip.c.

References append_history, AST_EXTENSION_DEACTIVATED, AST_EXTENSION_REMOVED, ast_extension_state2str(), ast_log(), ast_set_flag, ast_test_flag, ast_verb, sip_pvt::autokillid, DEFAULT_TRANS_TIMEOUT, FALSE, sip_pvt::flags, sip_pvt::laststate, LOG_WARNING, NONE, sip_pvt::pendinginvite, sip_cancel_destroy(), SIP_PAGE2_STATECHANGEQUEUE, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), sip_pvt::stateid, sip_pvt::subscribed, transmit_state_notify(), and sip_pvt::username.

Referenced by handle_request_subscribe(), handle_response(), and handle_response_notify().

{
   struct sip_pvt *p = data;

   sip_pvt_lock(p);

   switch(state) {
   case AST_EXTENSION_DEACTIVATED:  /* Retry after a while */
   case AST_EXTENSION_REMOVED:   /* Extension is gone */
      if (p->autokillid > -1 && sip_cancel_destroy(p))   /* Remove subscription expiry for renewals */
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);  /* Delete subscription in 32 secs */
      ast_verb(2, "Extension state: Watcher for hint %s %s. Notify User %s\n", exten, state == AST_EXTENSION_DEACTIVATED ? "deactivated" : "removed", p->username);
      p->stateid = -1;
      p->subscribed = NONE;
      append_history(p, "Subscribestatus", "%s", state == AST_EXTENSION_REMOVED ? "HintRemoved" : "Deactivated");
      break;
   default: /* Tell user */
      p->laststate = state;
      break;
   }
   if (p->subscribed != NONE) {  /* Only send state NOTIFY if we know the format */
      if (!p->pendinginvite) {
         transmit_state_notify(p, state, 1, FALSE);
      } else {
         /* We already have a NOTIFY sent that is not answered. Queue the state up.
            if many state changes happen meanwhile, we will only send a notification of the last one */
         ast_set_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
      }
   }
   ast_verb(2, "Extension Changed %s[%s] new state %s for Notify User %s %s\n", exten, context, ast_extension_state2str(state), p->username,
         ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE) ? "(queued)" : "");

   sip_pvt_unlock(p);

   return 0;
}

static void change_t38_state	(	struct sip_pvt *	p,
		int	state
	)			[static]

Change the T38 state on a SIP dialog.

Definition at line 4236 of file chan_sip.c.

References AST_CONTROL_T38, ast_debug, ast_queue_control_data(), AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_TERMINATED, chan, t38properties::direct, ast_channel::name, sip_pvt::owner, t38properties::state, sip_pvt::t38, T38_DISABLED, T38_ENABLED, T38_LOCAL_REINVITE, T38_PEER_DIRECT, and T38_PEER_REINVITE.

Referenced by handle_response_invite(), process_sdp(), sip_answer(), sip_indicate(), sip_read(), sip_t38_abort(), and sip_write().

{
   int old = p->t38.state;
   struct ast_channel *chan = p->owner;
   enum ast_control_t38 message = 0;

   /* Don't bother changing if we are already in the state wanted */
   if (old == state)
      return;

   if (state == T38_PEER_DIRECT) {
      p->t38.direct = 1;
   }

   p->t38.state = state;
   ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>");

   /* If no channel was provided we can't send off a control frame */
   if (!chan)
      return;

   /* Given the state requested and old state determine what control frame we want to queue up */
   if (state == T38_PEER_REINVITE)
      message = AST_T38_REQUEST_NEGOTIATE;
   else if (state == T38_ENABLED)
      message = AST_T38_NEGOTIATED;
   else if (state == T38_DISABLED && old == T38_ENABLED)
      message = AST_T38_TERMINATED;
   else if (state == T38_DISABLED && old == T38_LOCAL_REINVITE)
      message = AST_T38_REFUSED;

   /* Woot we got a message, create a control frame and send it on! */
   if (message)
      ast_queue_control_data(chan, AST_CONTROL_T38, &message, sizeof(message));
}

static enum check_auth_result check_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		const char *	username,
		const char *	secret,
		const char *	md5secret,
		int	sipmethod,
		char *	uri,
		enum xmittype	reliable,
		int	ignore
	)			[static]

Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set).

Returns:

0 on success, non-zero on error

XXX

Todo:

need a better return code here

XXX

Todo:

need a better return code here

Definition at line 10890 of file chan_sip.c.

References append_history, ast_copy_string(), AST_DYNSTR_BUILD_FAILED, ast_log(), ast_md5_hash(), ast_random(), ast_skip_blanks(), ast_str_set(), ast_str_thread_get(), ast_string_field_build, ast_strlen_zero(), AUTH_CHALLENGE_SENT, auth_headers(), AUTH_SECRET_FAILED, AUTH_SUCCESSFUL, AUTH_USERNAME_MISMATCH, buf, check_auth_buf, CHECK_AUTH_BUF_INITLEN, DEFAULT_TRANS_TIMEOUT, FALSE, get_header(), sip_request::ignore, LOG_NOTICE, LOG_WARNING, sip_pvt::randdata, s, S_OR, sip_methods, sip_scheddestroy(), strsep(), text, transmit_response_with_auth(), TRUE, and WWW_AUTH.

Referenced by check_peer_ok(), and register_verify().

{
   const char *response;
   char *reqheader, *respheader;
   const char *authtoken;
   char a1_hash[256];
   char resp_hash[256]="";
   char *c;
   int  wrongnonce = FALSE;
   int  good_response;
   const char *usednonce = p->randdata;
   struct ast_str *buf;
   int res;

   /* table of recognised keywords, and their value in the digest */
   enum keys { K_RESP, K_URI, K_USER, K_NONCE, K_LAST };
   struct x {
      const char *key;
      const char *s;
   } *i, keys[] = {
      [K_RESP] = { "response=", "" },
      [K_URI] = { "uri=", "" },
      [K_USER] = { "username=", "" },
      [K_NONCE] = { "nonce=", "" },
      [K_LAST] = { NULL, NULL}
   };

   /* Always OK if no secret */
   if (ast_strlen_zero(secret) && ast_strlen_zero(md5secret))
      return AUTH_SUCCESSFUL;

   /* Always auth with WWW-auth since we're NOT a proxy */
   /* Using proxy-auth in a B2BUA may block proxy authorization in the same transaction */
   response = "401 Unauthorized";

   /*
    * Note the apparent swap of arguments below, compared to other
    * usages of auth_headers().
    */
   auth_headers(WWW_AUTH, &respheader, &reqheader);

   authtoken =  get_header(req, reqheader);  
   if (ignore && !ast_strlen_zero(p->randdata) && ast_strlen_zero(authtoken)) {
      /* This is a retransmitted invite/register/etc, don't reconstruct authentication
         information */
      if (!reliable) {
         /* Resend message if this was NOT a reliable delivery.   Otherwise the
            retransmission should get it */
         transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
         /* Schedule auto destroy in 32 seconds (according to RFC 3261) */
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      }
      return AUTH_CHALLENGE_SENT;
   } else if (ast_strlen_zero(p->randdata) || ast_strlen_zero(authtoken)) {
      /* We have no auth, so issue challenge and request authentication */
      ast_string_field_build(p, randdata, "%08lx", ast_random()); /* Create nonce for challenge */
      transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return AUTH_CHALLENGE_SENT;
   } 

   /* --- We have auth, so check it */

   /* Whoever came up with the authentication section of SIP can suck my %&#$&* for not putting
      an example in the spec of just what it is you're doing a hash on. */

   if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN)))
      return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */

   /* Make a copy of the response and parse it */
   res = ast_str_set(&buf, 0, "%s", authtoken);

   if (res == AST_DYNSTR_BUILD_FAILED)
      return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */

   c = buf->str;

   while(c && *(c = ast_skip_blanks(c)) ) { /* lookup for keys */
      for (i = keys; i->key != NULL; i++) {
         const char *separator = ",";  /* default */

         if (strncasecmp(c, i->key, strlen(i->key)) != 0)
            continue;
         /* Found. Skip keyword, take text in quotes or up to the separator. */
         c += strlen(i->key);
         if (*c == '"') { /* in quotes. Skip first and look for last */
            c++;
            separator = "\"";
         }
         i->s = c;
         strsep(&c, separator);
         break;
      }
      if (i->key == NULL) /* not found, jump after space or comma */
         strsep(&c, " ,");
   }

   /* Verify that digest username matches  the username we auth as */
   if (strcmp(username, keys[K_USER].s)) {
      ast_log(LOG_WARNING, "username mismatch, have <%s>, digest has <%s>\n",
         username, keys[K_USER].s);
      /* Oops, we're trying something here */
      return AUTH_USERNAME_MISMATCH;
   }

   /* Verify nonce from request matches our nonce.  If not, send 401 with new nonce */
   if (strcasecmp(p->randdata, keys[K_NONCE].s)) { /* XXX it was 'n'casecmp ? */
      wrongnonce = TRUE;
      usednonce = keys[K_NONCE].s;
   }

   if (!ast_strlen_zero(md5secret))
      ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
   else {
      char a1[256];
      snprintf(a1, sizeof(a1), "%s:%s:%s", username, global_realm, secret);
      ast_md5_hash(a1_hash, a1);
   }

   /* compute the expected response to compare with what we received */
   {
      char a2[256];
      char a2_hash[256];
      char resp[256];

      snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text,
            S_OR(keys[K_URI].s, uri));
      ast_md5_hash(a2_hash, a2);
      snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash);
      ast_md5_hash(resp_hash, resp);
   }

   good_response = keys[K_RESP].s &&
         !strncasecmp(keys[K_RESP].s, resp_hash, strlen(resp_hash));
   if (wrongnonce) {
      if (good_response) {
         if (sipdebug)
            ast_log(LOG_NOTICE, "Correct auth, but based on stale nonce received from '%s'\n", get_header(req, "To"));
         /* We got working auth token, based on stale nonce . */
         ast_string_field_build(p, randdata, "%08lx", ast_random());
         transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, TRUE);
      } else {
         /* Everything was wrong, so give the device one more try with a new challenge */
         if (!req->ignore) {
            if (sipdebug)
               ast_log(LOG_NOTICE, "Bad authentication received from '%s'\n", get_header(req, "To"));
            ast_string_field_build(p, randdata, "%08lx", ast_random());
         } else {
            if (sipdebug)
               ast_log(LOG_NOTICE, "Duplicate authentication received from '%s'\n", get_header(req, "To"));
         }
         transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, FALSE);
      }

      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return AUTH_CHALLENGE_SENT;
   } 
   if (good_response) {
      append_history(p, "AuthOK", "Auth challenge succesful for %s", username);
      return AUTH_SUCCESSFUL;
   }

   /* Ok, we have a bad username/secret pair */
   /* Tell the UAS not to re-send this authentication data, because
      it will continue to fail
   */

   return AUTH_SECRET_FAILED;
}

static enum check_auth_result check_peer_ok	(	struct sip_pvt *	p,
		char *	of,
		struct sip_request *	req,
		int	sipmethod,
		struct sockaddr_in *	sin,
		struct sip_peer **	authpeer,
		enum xmittype	reliable,
		char *	rpid_num,
		char *	calleridname,
		char *	uri2
	)			[static]

Validate device authentication.

Definition at line 12206 of file chan_sip.c.

References sip_peer::accountcode, accountcode, sip_peer::amaflags, sip_pvt::amaflags, ao2_t_ref, ast_copy_flags, AST_FORMAT_TEXT_MASK, AST_FORMAT_VIDEO_MASK, ast_inet_ntoa(), ast_is_shrinkable_phonenumber(), ast_rtp_codec_setpref(), ast_rtp_destroy(), AST_RTP_DTMF, ast_set_flag, ast_shrink_phone_number(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verbose, AUTH_DONT_KNOW, sip_peer::autoframing, sip_pvt::autoframing, sip_peer::call_limit, sip_peer::callgroup, sip_pvt::callgroup, sip_pvt::callingpres, sip_peer::callingpres, sip_peer::capability, sip_pvt::capability, sip_peer::chanvars, sip_pvt::chanvars, check_auth(), cid_name, sip_peer::cid_name, cid_num, sip_peer::cid_num, context, sip_peer::context, copy_vars(), debug, do_setnat(), FALSE, find_peer(), FINDALLDEVICES, FINDPEERS, FINDUSERS, sip_peer::flags, sip_pvt::flags, sip_peer::fullcontact, sip_request::ignore, t38properties::jointcapability, sip_pvt::jointcapability, sip_pvt::jointnoncodeccapability, sip_peer::language, language, sip_peer::lastms, sip_peer::maxcallbitrate, sip_pvt::maxcallbitrate, sip_peer::maxms, sip_peer::md5secret, sip_peer::mohinterpret, mohinterpret, sip_peer::mohsuggest, mohsuggest, sip_peer::name, sip_pvt::noncodeccapability, sip_peer::parkinglot, parkinglot, t38properties::peercapability, sip_pvt::peercapability, sip_pvt::peermd5secret, sip_pvt::peersecret, sip_peer::pickupgroup, sip_pvt::pickupgroup, sip_pvt::prefs, sip_peer::prefs, sip_pvt::recv, replace_cid(), sip_pvt::rtp, sip_peer::secret, SIP_CALL_LIMIT, sip_debug_test_addr(), SIP_DTMF, SIP_DTMF_AUTO, SIP_DTMF_RFC2833, SIP_FLAGS_TO_COPY, SIP_INSECURE_INVITE, SIP_INVITE, SIP_NAT_ROUTE, SIP_PAGE2_FLAGS_TO_COPY, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_VIDEOSUPPORT, SIP_PAGE2_VIDEOSUPPORT_ALWAYS, SIP_SUBSCRIBE, sip_peer::sipoptions, sip_pvt::sipoptions, sip_peer::subscribecontext, sip_pvt::t38, sip_pvt::timer_b, sip_peer::timer_b, sip_peer::timer_t1, sip_pvt::timer_t1, sip_pvt::trtp, TRUE, unref_peer(), sip_peer::username, and sip_pvt::vrtp.

Referenced by check_user_full().

{
   enum check_auth_result res;
   int debug=sip_debug_test_addr(sin);
   struct sip_peer *peer;

   if (sipmethod == SIP_SUBSCRIBE) {
      /* For subscribes, match on device name only; for other methods,
      * match on IP address-port of the incoming request.
      */
      peer = find_peer(of, NULL, TRUE, FINDALLDEVICES, FALSE);
   } else {
      /* First find devices based on username (avoid all type=peer's) */
      peer = find_peer(of, NULL, TRUE, FINDUSERS, FALSE);

      /* Then find devices based on IP */
      if (!peer) {
         peer = find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE);
      }
   }

   if (!peer) {
      if (debug)
         ast_verbose("No matching peer for '%s' from '%s:%d'\n",
            of, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
      return AUTH_DONT_KNOW;
   }

   if (debug)
      ast_verbose("Found peer '%s' for '%s' from %s:%d\n",
         peer->name, of, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));

   /* XXX what about p->prefs = peer->prefs; ? */
   /* Set Frame packetization */
   if (p->rtp) {
      ast_rtp_codec_setpref(p->rtp, &peer->prefs);
      p->autoframing = peer->autoframing;
   }

   /* Take the peer */
   ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);

   /* Copy SIP extensions profile to peer */
   /* XXX is this correct before a successful auth ? */
   if (p->sipoptions)
      peer->sipoptions = p->sipoptions;

   replace_cid(p, rpid_num, calleridname);
   do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE));

   ast_string_field_set(p, peersecret, peer->secret);
   ast_string_field_set(p, peermd5secret, peer->md5secret);
   ast_string_field_set(p, subscribecontext, peer->subscribecontext);
   ast_string_field_set(p, mohinterpret, peer->mohinterpret);
   ast_string_field_set(p, mohsuggest, peer->mohsuggest);
   ast_string_field_set(p, parkinglot, peer->parkinglot);
   if (peer->callingpres)  /* Peer calling pres setting will override RPID */
      p->callingpres = peer->callingpres;
   if (peer->maxms && peer->lastms)
      p->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
   else
      p->timer_t1 = peer->timer_t1;
 
   /* Set timer B to control transaction timeouts */
   if (peer->timer_b)
      p->timer_b = peer->timer_b;
   else
      p->timer_b = 64 * p->timer_t1;
 
   if (ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)) {
      /* Pretend there is no required authentication */
      ast_string_field_set(p, peersecret, NULL);
      ast_string_field_set(p, peermd5secret, NULL);
   }
   if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable, req->ignore))) {
      ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
      ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
      /* If we have a call limit, set flag */
      if (peer->call_limit)
         ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
      ast_string_field_set(p, peername, peer->name);
      ast_string_field_set(p, authname, peer->name);

      if (sipmethod == SIP_INVITE) {
         /* copy channel vars */
         p->chanvars = copy_vars(peer->chanvars);
      }

      if (authpeer) {
         ao2_t_ref(peer, 1, "copy pointer into (*authpeer)");
         (*authpeer) = peer;  /* Add a ref to the object here, to keep it in memory a bit longer if it is realtime */
      }

      if (!ast_strlen_zero(peer->username)) {
         ast_string_field_set(p, username, peer->username);
         /* Use the default username for authentication on outbound calls */
         /* XXX this takes the name from the caller... can we override ? */
         ast_string_field_set(p, authname, peer->username);
      }
      if (!ast_strlen_zero(peer->cid_num)) {
         char *tmp = ast_strdupa(peer->cid_num);
         if (ast_is_shrinkable_phonenumber(tmp))
            ast_shrink_phone_number(tmp);
         ast_string_field_set(p, cid_num, tmp);
      }
      if (!ast_strlen_zero(peer->cid_name)) 
         ast_string_field_set(p, cid_name, peer->cid_name);
      ast_string_field_set(p, fullcontact, peer->fullcontact);
      if (!ast_strlen_zero(peer->context))
         ast_string_field_set(p, context, peer->context);
      ast_string_field_set(p, peersecret, peer->secret);
      ast_string_field_set(p, peermd5secret, peer->md5secret);
      ast_string_field_set(p, language, peer->language);
      ast_string_field_set(p, accountcode, peer->accountcode);
      p->amaflags = peer->amaflags;
      p->callgroup = peer->callgroup;
      p->pickupgroup = peer->pickupgroup;
      p->capability = peer->capability;
      p->prefs = peer->prefs;
      p->jointcapability = peer->capability;
      if (p->peercapability)
         p->jointcapability &= p->peercapability;
      p->maxcallbitrate = peer->maxcallbitrate;
      if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) &&
            (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ||
               !(p->capability & AST_FORMAT_VIDEO_MASK)) &&
            p->vrtp) {
         ast_rtp_destroy(p->vrtp);
         p->vrtp = NULL;
      }
      if ((!ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) || !(p->capability & AST_FORMAT_TEXT_MASK)) && p->trtp) {
         ast_rtp_destroy(p->trtp);
         p->trtp = NULL;
      }
      if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
          (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
         p->noncodeccapability |= AST_RTP_DTMF;
      else
         p->noncodeccapability &= ~AST_RTP_DTMF;
      p->jointnoncodeccapability = p->noncodeccapability;
      if (p->t38.peercapability)
         p->t38.jointcapability &= p->t38.peercapability;
   }
   unref_peer(peer, "check_peer_ok: unref_peer: tossing temp ptr to peer from find_peer");
   return res;
}

static void check_pendings

(

struct sip_pvt *

p

)

[static]

Check pending actions on SIP call.

Definition at line 15809 of file chan_sip.c.

References ast_clear_flag, ast_debug, ast_test_flag, sip_pvt::callid, DEFAULT_TRANS_TIMEOUT, FALSE, sip_pvt::flags, INV_CALLING, INV_EARLY_MEDIA, INV_PROCEEDING, sip_pvt::invitestate, sip_pvt::pendinginvite, SIP_BYE, SIP_CANCEL, SIP_NEEDREINVITE, SIP_PENDINGBYE, sip_scheddestroy(), t38properties::state, sip_pvt::t38, T38_LOCAL_REINVITE, transmit_reinvite_with_sdp(), transmit_request(), transmit_request_with_auth(), TRUE, sip_pvt::waitid, and XMIT_RELIABLE.

Referenced by handle_incoming(), handle_response_invite(), and sip_reinvite_retry().

{
   if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
      /* if we can't BYE, then this is really a pending CANCEL */
      if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)
         transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
         /* Actually don't destroy us yet, wait for the 487 on our original 
            INVITE, but do set an autodestruct just in case we never get it. */
      else {
         /* We have a pending outbound invite, don't send someting
            new in-transaction */
         if (p->pendinginvite)
            return;

         /* Perhaps there is an SD change INVITE outstanding */
         transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
      }
      ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);   
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
      /* if we can't REINVITE, hold it for later */
      if (p->pendinginvite || p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA || p->waitid > 0) {
         ast_debug(2, "NOT Sending pending reinvite (yet) on '%s'\n", p->callid);
      } else {
         ast_debug(2, "Sending pending reinvite on '%s'\n", p->callid);
         /* Didn't get to reinvite yet, so do it now */
         transmit_reinvite_with_sdp(p, (p->t38.state == T38_LOCAL_REINVITE ? TRUE : FALSE), FALSE);
         ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE); 
      }
   }
}

static void check_rtp_timeout	(	struct sip_pvt *	dialog,
		time_t	t
	)			[static]

helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked

Todo:

Check video RTP keepalives

Do we need to move the lastrtptx to the RTP structure to have one for audio and one for video? It really does belong to the RTP structure.

Definition at line 20275 of file chan_sip.c.

References ast_channel::_state, ast_channel_trylock, ast_channel_unlock, ast_log(), ast_rtp_get_peer(), ast_rtp_get_rtpholdtimeout(), ast_rtp_get_rtpkeepalive(), ast_rtp_get_rtptimeout(), ast_rtp_sendcng(), ast_rtp_set_rtpholdtimeout(), ast_rtp_set_rtptimeout(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), AST_STATE_UP, ast_test_flag, sip_pvt::flags, sip_pvt::lastrtprx, sip_pvt::lastrtptx, LOG_NOTICE, ast_channel::name, sip_pvt::owner, sip_pvt::redirip, sip_pvt::rtp, SIP_PAGE2_CALL_ONHOLD, sip_pvt_lock, sip_pvt_unlock, t38properties::state, sip_pvt::t38, T38_ENABLED, and sip_pvt::vrtp.

Referenced by dialog_needdestroy().

{
   /* If we have no RTP or no active owner, no need to check timers */
   if (!dialog->rtp || !dialog->owner)
      return;
   /* If the call is not in UP state or redirected outside Asterisk, no need to check timers */

   if (dialog->owner->_state != AST_STATE_UP || dialog->redirip.sin_addr.s_addr)
      return;

   /* If the call is involved in a T38 fax session do not check RTP timeout */
   if (dialog->t38.state == T38_ENABLED)
      return;

   /* If we have no timers set, return now */
   if ((ast_rtp_get_rtpkeepalive(dialog->rtp) == 0) && (ast_rtp_get_rtptimeout(dialog->rtp) == 0) && (ast_rtp_get_rtpholdtimeout(dialog->rtp) == 0))
      return;

   /* Check AUDIO RTP keepalives */
   if (dialog->lastrtptx && ast_rtp_get_rtpkeepalive(dialog->rtp) &&
          (t > dialog->lastrtptx + ast_rtp_get_rtpkeepalive(dialog->rtp))) {
      /* Need to send an empty RTP packet */
      dialog->lastrtptx = time(NULL);
      ast_rtp_sendcng(dialog->rtp, 0);
   }

   /*! \todo Check video RTP keepalives

      Do we need to move the lastrtptx to the RTP structure to have one for audio and one
      for video? It really does belong to the RTP structure.
   */

   /* Check AUDIO RTP timers */
   if (dialog->lastrtprx && (ast_rtp_get_rtptimeout(dialog->rtp) || ast_rtp_get_rtpholdtimeout(dialog->rtp)) &&
          (t > dialog->lastrtprx + ast_rtp_get_rtptimeout(dialog->rtp))) {

      /* Might be a timeout now -- see if we're on hold */
      struct sockaddr_in sin;
      ast_rtp_get_peer(dialog->rtp, &sin);
      if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_get_rtpholdtimeout(dialog->rtp) &&
           (t > dialog->lastrtprx + ast_rtp_get_rtpholdtimeout(dialog->rtp)))) {
         /* Needs a hangup */
         if (ast_rtp_get_rtptimeout(dialog->rtp)) {
            while (dialog->owner && ast_channel_trylock(dialog->owner)) {
               sip_pvt_unlock(dialog);
               usleep(1);
               sip_pvt_lock(dialog);
            }
            ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
               dialog->owner->name, (long) (t - dialog->lastrtprx));
            /* Issue a softhangup */
            ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
            ast_channel_unlock(dialog->owner);
            /* forget the timeouts for this call, since a hangup
               has already been requested and we don't want to
               repeatedly request hangups
            */
            ast_rtp_set_rtptimeout(dialog->rtp, 0);
            ast_rtp_set_rtpholdtimeout(dialog->rtp, 0);
            if (dialog->vrtp) {
               ast_rtp_set_rtptimeout(dialog->vrtp, 0);
               ast_rtp_set_rtpholdtimeout(dialog->vrtp, 0);
            }
         }
      }
   }
}

static int check_sip_domain	(	const char *	domain,
		char *	context,
		size_t	len
	)			[static]

check_sip_domain: Check if domain part of uri is local to our server

Definition at line 21335 of file chan_sip.c.

References ast_copy_string(), AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_strlen_zero(), domain::context, domain::domain, and domain::list.

Referenced by func_check_sipdomain(), get_destination(), handle_request_refer(), and register_verify().

{
   struct domain *d;
   int result = 0;

   AST_LIST_LOCK(&domain_list);
   AST_LIST_TRAVERSE(&domain_list, d, list) {
      if (strcasecmp(d->domain, domain))
         continue;

      if (len && !ast_strlen_zero(d->context))
         ast_copy_string(context, d->context, len);
      
      result = 1;
      break;
   }
   AST_LIST_UNLOCK(&domain_list);

   return result;
}

static int check_user	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	sipmethod,
		char *	uri,
		enum xmittype	reliable,
		struct sockaddr_in *	sin
	)			[static]

Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.

Definition at line 12484 of file chan_sip.c.

References check_user_full().

Referenced by handle_request_invite().

{
   return check_user_full(p, req, sipmethod, uri, reliable, sin, NULL);
}

static enum check_auth_result check_user_full	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	sipmethod,
		char *	uri,
		enum xmittype	reliable,
		struct sockaddr_in *	sin,
		struct sip_peer **	authpeer
	)			[static]

Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.

Returns:

0 on success, non-zero on failure

Definition at line 12364 of file chan_sip.c.

References ast_copy_string(), ast_is_shrinkable_phonenumber(), ast_log(), ast_shrink_phone_number(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_uri_decode(), AUTH_DONT_KNOW, AUTH_FAKE_AUTH, AUTH_SECRET_FAILED, AUTH_SUCCESSFUL, build_contact(), check_peer_ok(), cid_name, cid_num, dummy(), exten, get_calleridname(), get_header(), get_in_brackets(), get_rpid_num(), LOG_NOTICE, parse_uri(), replace_cid(), SIP_TRANSPORT_TLS, strsep(), and terminate_uri().

Referenced by check_user(), and handle_request_subscribe().

{
   char from[256];
   char *dummy;   /* dummy return value for parse_uri */
   char *domain;  /* dummy return value for parse_uri */
   char *of, *of2;
   char rpid_num[50];
   const char *rpid;
   enum check_auth_result res;
   char calleridname[50];
   char *uri2 = ast_strdupa(uri);

   terminate_uri(uri2); /* trim extra stuff */

   ast_copy_string(from, get_header(req, "From"), sizeof(from));
   if (pedanticsipchecking)
      ast_uri_decode(from);
   /* XXX here tries to map the username for invite things */
   memset(calleridname, 0, sizeof(calleridname));
   get_calleridname(from, calleridname, sizeof(calleridname));
   if (calleridname[0])
      ast_string_field_set(p, cid_name, calleridname);

   rpid = get_header(req, "Remote-Party-ID");
   memset(rpid_num, 0, sizeof(rpid_num));
   if (!ast_strlen_zero(rpid)) 
      p->callingpres = get_rpid_num(rpid, rpid_num, sizeof(rpid_num));

   of = get_in_brackets(from);
   if (ast_strlen_zero(p->exten)) {
      char *t = uri2;
      if (!strncasecmp(t, "sip:", 4))
         t+= 4;
      else if (!strncasecmp(t, "sips:", 5))
         t += 5;
      ast_string_field_set(p, exten, t);
      t = strchr(p->exten, '@');
      if (t)
         *t = '\0';
      if (ast_strlen_zero(p->our_contact))
         build_contact(p);
   }
   /* save the URI part of the From header */
   ast_string_field_set(p, from, of);

   of2 = ast_strdupa(of);

   /* ignore all fields but name */
   if (p->socket.type == SIP_TRANSPORT_TLS) {
      if (parse_uri(of, "sips:", &of, &dummy, &domain, &dummy, &dummy)) {
         if (parse_uri(of2, "sip:", &of, &dummy, &domain, &dummy, &dummy))
            ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
      }
   } else {
      if (parse_uri(of, "sip:", &of, &dummy, &domain, &dummy, &dummy))
         ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
   }

   if (ast_strlen_zero(of)) {
      /* XXX note: the original code considered a missing @host
       * as a username-only URI. The SIP RFC (19.1.1) says that
       * this is wrong, and it should be considered as a domain-only URI.
       * For backward compatibility, we keep this block, but it is
       * really a mistake and should go away.
       */
      of = domain;
   }
   {
      char *tmp = ast_strdupa(of);
      /* We need to be able to handle auth-headers looking like
         <sip:8164444422;phone-context=+1@1.2.3.4:5060;user=phone;tag=SDadkoa01-gK0c3bdb43>
      */
      tmp = strsep(&tmp, ";");
      if (ast_is_shrinkable_phonenumber(tmp))
         ast_shrink_phone_number(tmp);
      ast_string_field_set(p, cid_num, tmp);
   }

   if (global_match_auth_username) {
      /*
       * XXX This is experimental code to grab the search key from the
       * Auth header's username instead of the 'From' name, if available.
       * Do not enable this block unless you understand the side effects (if any!)
       * Note, the search for "username" should be done in a more robust way.
       * Note2, at the moment we check both fields, though maybe we should
       * pick one or another depending on the request ? XXX
       */
      const char *hdr = get_header(req, "Authorization");
      if (ast_strlen_zero(hdr))
         hdr = get_header(req, "Proxy-Authorization");

      if ( !ast_strlen_zero(hdr) && (hdr = strstr(hdr, "username=\"")) ) {
         ast_copy_string(from, hdr + strlen("username=\""), sizeof(from));
         of = from;
         of = strsep(&of, "\"");
      }
   }

   res = check_peer_ok(p, of, req, sipmethod, sin,
         authpeer, reliable, rpid_num, calleridname, uri2);
   if (res != AUTH_DONT_KNOW)
      return res;

   /* Finally, apply the guest policy */
   if (global_allowguest) {
      replace_cid(p, rpid_num, calleridname);
      res = AUTH_SUCCESSFUL;
   } else if (global_alwaysauthreject)
      res = AUTH_FAKE_AUTH; /* reject with fake authorization request */
   else
      res = AUTH_SECRET_FAILED; /* we don't want any guests, authentication will fail */

   return res;
}

static void check_via	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

check Via: header for hostname, port and rport request/answer

Definition at line 12049 of file chan_sip.c.

References ahp, ast_copy_string(), ast_gethostbyname(), ast_inet_ntoa(), ast_log(), ast_set_flag, ast_skip_blanks(), ast_verbose, sip_pvt::flags, get_header(), hp, LOG_WARNING, sip_pvt::sa, sip_debug_test_pvt(), sip_nat_mode(), SIP_NAT_ROUTE, sip_real_dst(), and STANDARD_SIP_PORT.

Referenced by handle_request_bye(), handle_request_cancel(), handle_request_invite(), handle_request_register(), and handle_request_subscribe().

{
   char via[512];
   char *c, *pt;
   struct hostent *hp;
   struct ast_hostent ahp;

   ast_copy_string(via, get_header(req, "Via"), sizeof(via));

   /* Work on the leftmost value of the topmost Via header */
   c = strchr(via, ',');
   if (c)
      *c = '\0';

   /* Check for rport */
   c = strstr(via, ";rport");
   if (c && (c[6] != '=')) /* rport query, not answer */
      ast_set_flag(&p->flags[0], SIP_NAT_ROUTE);

   c = strchr(via, ';');
   if (c) 
      *c = '\0';

   c = strchr(via, ' ');
   if (c) {
      *c = '\0';
      c = ast_skip_blanks(c+1);
      if (strcasecmp(via, "SIP/2.0/UDP") && strcasecmp(via, "SIP/2.0/TCP") && strcasecmp(via, "SIP/2.0/TLS")) {
         ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via);
         return;
      }
      pt = strchr(c, ':');
      if (pt)
         *pt++ = '\0';  /* remember port pointer */
      hp = ast_gethostbyname(c, &ahp);
      if (!hp) {
         ast_log(LOG_WARNING, "'%s' is not a valid host\n", c);
         return;
      }
      memset(&p->sa, 0, sizeof(p->sa));
      p->sa.sin_family = AF_INET;
      memcpy(&p->sa.sin_addr, hp->h_addr, sizeof(p->sa.sin_addr));
      p->sa.sin_port = htons(pt ? atoi(pt) : STANDARD_SIP_PORT);

      if (sip_debug_test_pvt(p)) {
         const struct sockaddr_in *dst = sip_real_dst(p);
         ast_verbose("Sending to %s : %d (%s)\n", ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), sip_nat_mode(p));
      }
   }
}

static attribute_unused void check_via_response	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

check received= and rport= in a SIP response. If we get a response with received= and/or rport= in the Via: line, use them as 'p->ourip' (see RFC 3581 for rport, and RFC 3261 for received). Using these two fields SIP can produce the correct address and port in the SIP headers without the need for STUN. The address part is also reused for the media sessions. Note that ast_sip_ouraddrfor() still rewrites p->ourip if you specify externip/seternaddr/stunaddr.

Definition at line 12019 of file chan_sip.c.

References ast_copy_string(), ast_parse_arg(), get_header(), sip_pvt::ourip, PARSE_INADDR, and strsep().

{
   char via[256];
   char *cur, *opts;

   ast_copy_string(via, get_header(req, "Via"), sizeof(via));

   /* Work on the leftmost value of the topmost Via header */
   opts = strchr(via, ',');
   if (opts)
      *opts = '\0';

   /* parse all relevant options */
   opts = strchr(via, ';');
   if (!opts)
      return;  /* no options to parse */
   *opts++ = '\0';
   while ( (cur = strsep(&opts, ";")) ) {
      if (!strncmp(cur, "rport=", 6)) {
         int port = strtol(cur+6, NULL, 10);
         /* XXX add error checking */
         p->ourip.sin_port = ntohs(port);
      } else if (!strncmp(cur, "received=", 9)) {
         if (ast_parse_arg(cur+9, PARSE_INADDR, &p->ourip))
            ;  /* XXX add error checking */
      }
   }
}

static void cleanup_stale_contexts	(	char *	new,
		char *	old
	)			[static]

Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.

Definition at line 13209 of file chan_sip.c.

References ast_context_destroy(), ast_context_find(), ast_copy_string(), AST_MAX_CONTEXT, and strsep().

Referenced by reload_config().

{
   char *oldcontext, *newcontext, *stalecontext, *stringp, newlist[AST_MAX_CONTEXT];

   while ((oldcontext = strsep(&old, "&"))) {
      stalecontext = '\0';
      ast_copy_string(newlist, new, sizeof(newlist));
      stringp = newlist;
      while ((newcontext = strsep(&stringp, "&"))) {
         if (!strcmp(newcontext, oldcontext)) {
            /* This is not the context you're looking for */
            stalecontext = '\0';
            break;
         } else if (strcmp(newcontext, oldcontext)) {
            stalecontext = oldcontext;
         }
         
      }
      if (stalecontext)
         ast_context_destroy(ast_context_find(stalecontext), "SIP");
   }
}

static void clear_peer_mailboxes

(

struct sip_peer *

peer

)

[static]

Destroy all peer-related mailbox subscriptions

Definition at line 3843 of file chan_sip.c.

References AST_LIST_REMOVE_HEAD, destroy_mailbox(), sip_mailbox::entry, mailbox, and sip_peer::mailboxes.

Referenced by set_peer_defaults(), and sip_destroy_peer().

{
   struct sip_mailbox *mailbox;

   while ((mailbox = AST_LIST_REMOVE_HEAD(&peer->mailboxes, entry)))
      destroy_mailbox(mailbox);
}

static int clear_realm_authentication

(

struct sip_auth *

authlist

)

[static]

Clear realm authentication list (at reload).

Definition at line 21426 of file chan_sip.c.

References ast_free, and sip_auth::next.

Referenced by build_peer(), reload_config(), sip_destroy_peer(), and unload_module().

{
   struct sip_auth *a = authlist;
   struct sip_auth *b;

   while (a) {
      b = a;
      a = a->next;
      ast_free(b);
   }

   return 1;
}

static void clear_sip_domains

(

void

)

[static]

Clear our domain list (at reload).

Definition at line 21357 of file chan_sip.c.

References ast_free, AST_LIST_LOCK, AST_LIST_REMOVE_HEAD, AST_LIST_UNLOCK, and domain::list.

Referenced by reload_config(), and unload_module().

{
   struct domain *d;

   AST_LIST_LOCK(&domain_list);
   while ((d = AST_LIST_REMOVE_HEAD(&domain_list, list)))
      ast_free(d);
   AST_LIST_UNLOCK(&domain_list);
}

static const char* cli_yesno

(

int

x

)

[static]

return Yes or No depending on the argument. This is used in many places in CLI command, having a function to generate this helps maintaining a consistent output (and possibly emitting the output in other languages, at some point).

Definition at line 12724 of file chan_sip.c.

Referenced by _sip_show_peer(), show_channels_cb(), sip_show_channel(), sip_show_settings(), sip_show_user(), and sip_show_users().

{
   return x ? "Yes" : "No";
}

static char * complete_sip_peer	(	const char *	word,
		int	state,
		int	flags2
	)			[static]

Do completion on peer name.

Definition at line 14530 of file chan_sip.c.

References ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, ast_test_flag, sip_peer::flags, sip_peer::name, and unref_peer().

Referenced by complete_sip_show_peer(), complete_sipnotify(), sip_do_debug(), and sip_prune_realtime().

{
   char *result = NULL;
   int wordlen = strlen(word);
   int which = 0;
   struct ao2_iterator i = ao2_iterator_init(peers, 0);
   struct sip_peer *peer;

   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
      /* locking of the object is not required because only the name and flags are being compared */
      if (!strncasecmp(word, peer->name, wordlen) &&
            (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
            ++which > state)
         result = ast_strdup(peer->name);
      unref_peer(peer, "toss iterator peer ptr before break");
      if (result) {
         break;
      }
   }
   return result;
}

static char * complete_sip_registered_peer	(	const char *	word,
		int	state,
		int	flags2
	)			[static]

Do completion on registered peer name.

Definition at line 14553 of file chan_sip.c.

References ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, ast_test_flag, sip_peer::expire, sip_peer::flags, sip_peer::name, and unref_peer().

Referenced by complete_sip_unregister().

{
       char *result = NULL;
       int wordlen = strlen(word);
       int which = 0;
      struct ao2_iterator i;
      struct sip_peer *peer;

      i = ao2_iterator_init(peers, 0);
      while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
         if (!strncasecmp(word, peer->name, wordlen) &&
            (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
            ++which > state && peer->expire > 0)
            result = ast_strdup(peer->name);
         if (result) {
            unref_peer(peer, "toss iterator peer ptr before break");
            break;
         }
         unref_peer(peer, "toss iterator peer ptr");
       }
       return result;
}

static char * complete_sip_show_history	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip show history' CLI.

Definition at line 14577 of file chan_sip.c.

References complete_sipch().

Referenced by sip_show_history().

{
   if (pos == 3)
      return complete_sipch(line, word, pos, state);

   return NULL;
}

static char * complete_sip_show_peer	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip show peer' CLI.

Definition at line 14586 of file chan_sip.c.

References complete_sip_peer().

Referenced by sip_qualify_peer(), and sip_show_peer().

{
   if (pos == 3) {
      return complete_sip_peer(word, state, 0);
   }

   return NULL;
}

static char* complete_sip_show_user	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip show user' CLI.

Definition at line 13884 of file chan_sip.c.

References complete_sip_user().

Referenced by sip_show_user().

{
   if (pos == 3)
      return complete_sip_user(word, state);

   return NULL;
}

static char * complete_sip_unregister	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip unregister' CLI.

Definition at line 14596 of file chan_sip.c.

References complete_sip_registered_peer().

Referenced by sip_unregister().

{
       if (pos == 2)
               return complete_sip_registered_peer(word, state, 0);

       return NULL;
}

static char* complete_sip_user	(	const char *	word,
		int	state
	)			[static]

Do completion on user name.

Definition at line 13858 of file chan_sip.c.

References ao2_iterator_init(), ao2_iterator_next, ao2_lock(), ao2_unlock(), ast_strdup, SIP_TYPE_USER, unref_peer(), and user.

Referenced by complete_sip_show_user().

{
   char *result = NULL;
   int wordlen = strlen(word);
   int which = 0;
   struct ao2_iterator user_iter;
   struct sip_peer *user;

   user_iter = ao2_iterator_init(peers, 0);
   while ((user = ao2_iterator_next(&user_iter))) {
      ao2_lock(user);
      if (!(user->type & SIP_TYPE_USER)) {
         ao2_unlock(user);
         unref_peer(user, "complete sip user");
         continue;
      }
      /* locking of the object is not required because only the name and flags are being compared */
      if (!strncasecmp(word, user->name, wordlen) && ++which > state) {
         result = ast_strdup(user->name);
      }
      ao2_unlock(user);
      unref_peer(user, "complete sip user");
   }
   return result;
}

static char* complete_sipch	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip show channel' and 'sip show history' CLI This is in charge of generating all strings that match a prefix in the given position. As many functions of this kind, each invokation has O(state) time complexity so be careful in using it.

Definition at line 14500 of file chan_sip.c.

References ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, sip_pvt::callid, dialog_unref(), dialogs, sip_pvt_lock, and sip_pvt_unlock.

Referenced by complete_sip_show_history(), and sip_show_channel().

{
   int which=0;
   struct sip_pvt *cur;
   char *c = NULL;
   int wordlen = strlen(word);
   struct ao2_iterator i;

   if (pos != 3) {
      return NULL;
   }

   i = ao2_iterator_init(dialogs, 0);
   
   while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      sip_pvt_lock(cur);
      if (!strncasecmp(word, cur->callid, wordlen) && ++which > state) {
         c = ast_strdup(cur->callid);
         sip_pvt_unlock(cur);
         dialog_unref(cur, "drop ref in iterator loop break");
         break;
      }
      sip_pvt_unlock(cur);
      dialog_unref(cur, "drop ref in iterator loop");
   }
   return c;
}

static char * complete_sipnotify	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip notify' CLI.

Definition at line 14605 of file chan_sip.c.

References ast_category_browse(), ast_strdup, complete_sip_peer(), and notify_types.

Referenced by sip_cli_notify().

{
   char *c = NULL;

   if (pos == 2) {
      int which = 0;
      char *cat = NULL;
      int wordlen = strlen(word);

      /* do completion for notify type */

      if (!notify_types)
         return NULL;
      
      while ( (cat = ast_category_browse(notify_types, cat)) ) {
         if (!strncasecmp(word, cat, wordlen) && ++which > state) {
            c = ast_strdup(cat);
            break;
         }
      }
      return c;
   }

   if (pos > 2)
      return complete_sip_peer(word, state, 0);

   return NULL;
}

static int copy_all_header	(	struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)			[static]

Copy all headers from one request to another.

Definition at line 7693 of file chan_sip.c.

References __get_header(), add_header(), and ast_strlen_zero().

Referenced by respprep().

{
   int start = 0;
   int copied = 0;
   for (;;) {
      const char *tmp = __get_header(orig, field, &start);

      if (ast_strlen_zero(tmp))
         break;
      /* Add what we're responding to */
      add_header(req, field, tmp);
      copied++;
   }
   return copied ? 0 : -1;
}

static int copy_header	(	struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)			[static]

Copy one header field from one request to another.

Definition at line 7682 of file chan_sip.c.

References add_header(), ast_log(), ast_strlen_zero(), get_header(), and LOG_NOTICE.

Referenced by reqprep(), and respprep().

{
   const char *tmp = get_header(orig, field);

   if (!ast_strlen_zero(tmp)) /* Add what we're responding to */
      return add_header(req, field, tmp);
   ast_log(LOG_NOTICE, "No field '%s' present to copy\n", field);
   return -1;
}

static void copy_request	(	struct sip_request *	dst,
		const struct sip_request *	src
	)			[static]

copy SIP request (mostly used to save request for responses)

Definition at line 8972 of file chan_sip.c.

References ast_str_create(), ast_str_make_space(), sip_request::data, ast_str::str, and ast_str::used.

Referenced by _sip_tcp_helper_thread(), handle_request_bye(), handle_request_invite(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), initialize_initreq(), parse_copy(), queue_request(), sip_park(), and sip_park_thread().

{
   struct ast_str *duplicate = dst->data;

   /* First copy stuff */
   memcpy(dst, src, sizeof(*dst));
   dst->data = duplicate;

   /* All these + 1's are to account for the need to include the NULL terminator
    * Using typical string functions like ast_copy_string or ast_str_set will not
    * work in this case because the src's data string is riddled with \0's all over
    * the place and so a memcpy is the only way to accurately copy the string
    */

   if (!dst->data && !(dst->data = ast_str_create(src->data->used + 1)))
      return;
   else if (dst->data->len < src->data->used + 1)
      ast_str_make_space(&dst->data, src->data->used + 1);
      
   memcpy(dst->data->str, src->data->str, src->data->used + 1);
   dst->data->used = src->data->used;
}

static void copy_socket_data	(	struct sip_socket *	to_sock,
		const struct sip_socket *	from_sock
	)			[static]

Definition at line 4287 of file chan_sip.c.

References ao2_ref, and sip_socket::tcptls_session.

Referenced by create_addr_from_peer(), handle_request_do(), parse_register_contact(), sip_poke_peer(), and transmit_response_using_temp().

{
   if (to_sock->tcptls_session) {
      ao2_ref(to_sock->tcptls_session, -1);
      to_sock->tcptls_session = NULL;
   }

   if (from_sock->tcptls_session) {
      ao2_ref(from_sock->tcptls_session, +1);
   }

   *to_sock = *from_sock;
}

static struct ast_variable* copy_vars

(

struct ast_variable *

src

)

[static]

duplicate a list of channel variables,

Returns:

the copy.

Definition at line 12178 of file chan_sip.c.

References ast_variable_new(), and ast_variable::next.

Referenced by check_peer_ok().

{
   struct ast_variable *res = NULL, *tmp, *v = NULL;

   for (v = src ; v ; v = v->next) {
      if ((tmp = ast_variable_new(v->name, v->value, v->file))) {
         tmp->next = res;
         res = tmp;
      }
   }
   return res;
}

static int copy_via_headers	(	struct sip_pvt *	p,
		struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)			[static]

Copy SIP VIA Headers from the request to the response.

Note:

If the client indicates that it wishes to know the port we received from, it adds ;rport without an argument to the topmost via header. We need to add the port number (from our point of view) to that parameter.

	We always add ;received=<ip address> to the topmost via header.

Received: RFC 3261, rport RFC 3581

Definition at line 7717 of file chan_sip.c.

References __get_header(), add_header(), ast_copy_string(), ast_inet_ntoa(), ast_log(), ast_strlen_zero(), ast_test_flag, sip_pvt::flags, LOG_NOTICE, sip_pvt::recv, SIP_NAT, SIP_NAT_ALWAYS, and SIP_NAT_RFC3581.

Referenced by respprep().

{
   int copied = 0;
   int start = 0;

   for (;;) {
      char new[512];
      const char *oh = __get_header(orig, field, &start);

      if (ast_strlen_zero(oh))
         break;

      if (!copied) { /* Only check for empty rport in topmost via header */
         char leftmost[512], *others, *rport;

         /* Only work on leftmost value */
         ast_copy_string(leftmost, oh, sizeof(leftmost));
         others = strchr(leftmost, ',');
         if (others)
             *others++ = '\0';

         /* Find ;rport;  (empty request) */
         rport = strstr(leftmost, ";rport");
         if (rport && *(rport+6) == '=') 
            rport = NULL;     /* We already have a parameter to rport */

         /* Check rport if NAT=yes or NAT=rfc3581 (which is the default setting)  */
         if (rport && ((ast_test_flag(&p->flags[0], SIP_NAT) == SIP_NAT_ALWAYS) || (ast_test_flag(&p->flags[0], SIP_NAT) == SIP_NAT_RFC3581))) {
            /* We need to add received port - rport */
            char *end;

            rport = strstr(leftmost, ";rport");

            if (rport) {
               end = strchr(rport + 1, ';');
               if (end)
                  memmove(rport, end, strlen(end) + 1);
               else
                  *rport = '\0';
            }

            /* Add rport to first VIA header if requested */
            snprintf(new, sizeof(new), "%s;received=%s;rport=%d%s%s",
               leftmost, ast_inet_ntoa(p->recv.sin_addr),
               ntohs(p->recv.sin_port),
               others ? "," : "", others ? others : "");
         } else {
            /* We should *always* add a received to the topmost via */
            snprintf(new, sizeof(new), "%s;received=%s%s%s",
               leftmost, ast_inet_ntoa(p->recv.sin_addr),
               others ? "," : "", others ? others : "");
         }
         oh = new;   /* the header to copy */
      }  /* else add the following via headers untouched */
      add_header(req, field, oh);
      copied++;
   }
   if (!copied) {
      ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field);
      return -1;
   }
   return 0;
}

static int create_addr	(	struct sip_pvt *	dialog,
		const char *	opeer,
		struct sockaddr_in *	sin,
		int	newdialog
	)			[static]

create address structure from device name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success

Todo:

Fix this function. When we ask SRC, we should check all transports In the future, we should first check NAPTR to find out transport preference

Definition at line 4448 of file chan_sip.c.

References ahp, ast_copy_string(), ast_get_srv(), ast_gethostbyname(), ast_log(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, create_addr_from_peer(), do_setnat(), FALSE, find_peer(), FINDPEERS, sip_pvt::flags, get_transport(), hp, LOG_WARNING, MAXHOSTNAMELEN, obproxy_get(), sip_socket::port, sip_pvt::recv, ref_proxy(), sip_pvt::sa, service, SIP_NAT, SIP_NAT_ROUTE, SIP_TRANSPORT_TLS, SIP_TRANSPORT_UDP, sip_pvt::socket, STANDARD_SIP_PORT, STANDARD_TLS_PORT, sip_pvt::timer_b, sip_pvt::timer_t1, sip_peer::tohost, TRUE, sip_socket::type, and unref_peer().

{
   struct hostent *hp;
   struct ast_hostent ahp;
   struct sip_peer *peer;
   char *port;
   int portno = 0;
   char host[MAXHOSTNAMELEN], *hostn;
   char peername[256];
   int srv_ret = 0;

   ast_copy_string(peername, opeer, sizeof(peername));
   port = strchr(peername, ':');
   if (port)
      *port++ = '\0';
   dialog->sa.sin_family = AF_INET;
   dialog->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
   dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
   peer = find_peer(peername, NULL, TRUE, FINDPEERS, FALSE);

   if (peer) {
      int res;
      if (newdialog)
         dialog->socket.type = 0;
      res = create_addr_from_peer(dialog, peer);
      if (!ast_strlen_zero(port)) {
         if ((portno = atoi(port))) {
            dialog->sa.sin_port = dialog->recv.sin_port = htons(portno);
         }
      }
      unref_peer(peer, "create_addr: unref peer from find_peer hashtab lookup");
      return res;
   }

   do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE);

   ast_string_field_set(dialog, tohost, peername);

   /* Get the outbound proxy information */
   ref_proxy(dialog, obproxy_get(dialog, NULL));

   if (sin) {
      /* This address should be updated using dnsmgr */
      memcpy(&dialog->sa.sin_addr, &sin->sin_addr, sizeof(dialog->sa.sin_addr));
      if (!sin->sin_port) {
         if (ast_strlen_zero(port) || sscanf(port, "%u", &portno) != 1) {
            portno = (dialog->socket.type & SIP_TRANSPORT_TLS) ?
               STANDARD_TLS_PORT : STANDARD_SIP_PORT;
         }
      } else {
         portno = ntohs(sin->sin_port);
      }
   } else {

      /* Let's see if we can find the host in DNS. First try DNS SRV records,
         then hostname lookup */
      /*! \todo Fix this function. When we ask SRC, we should check all transports 
           In the future, we should first check NAPTR to find out transport preference
       */
      hostn = peername;
      /* Section 4.2 of RFC 3263 specifies that if a port number is specified, then
       * an A record lookup should be used instead of SRV.
       */
      if (!port && global_srvlookup) {
         char service[MAXHOSTNAMELEN];
         int tportno;
   
         snprintf(service, sizeof(service), "_sip._%s.%s", get_transport(dialog->socket.type), peername);
         srv_ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
         if (srv_ret > 0) {
            hostn = host;
            portno = tportno;
         }
      }
      if (!portno)
         portno = port ? atoi(port) : (dialog->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
      hp = ast_gethostbyname(hostn, &ahp);
      if (!hp) {
         ast_log(LOG_WARNING, "No such host: %s\n", peername);
         return -1;
      }
      memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr));
   }

   if (!dialog->socket.type)
      dialog->socket.type = SIP_TRANSPORT_UDP;
   if (!dialog->socket.port)
      dialog->socket.port = bindaddr.sin_port;
   dialog->sa.sin_port = htons(portno);
   dialog->recv = dialog->sa;
   return 0;
}

static int create_addr_from_peer	(	struct sip_pvt *	dialog,
		struct sip_peer *	peer
	)			[static]

Create address structure from peer reference. This function copies data from peer to the dialog, so we don't have to look up the peer again from memory or database during the life time of the dialog.

Returns:

-1 on error, 0 on success.

Definition at line 4308 of file chan_sip.c.

References sip_peer::addr, sip_peer::allowtransfer, sip_pvt::allowtransfer, ao2_t_link, ao2_t_unlink, ast_copy_flags, AST_FORMAT_VIDEO_MASK, ast_inet_ntoa(), ast_rtp_codec_setpref(), ast_rtp_destroy(), AST_RTP_DTMF, ast_rtp_set_rtpholdtimeout(), ast_rtp_set_rtpkeepalive(), ast_rtp_set_rtptimeout(), ast_rtp_setdtmf(), ast_rtp_setdtmfcompensate(), ast_set_flag, ast_strdupa, ast_string_field_build, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_udptl_destroy(), ast_udptl_new_with_bindaddr(), sip_peer::auth, sip_peer::autoframing, sip_pvt::autoframing, sip_peer::call_limit, sip_peer::callgroup, sip_pvt::callgroup, sip_pvt::callid, t38properties::capability, sip_peer::capability, sip_pvt::capability, check_request_transport, sip_peer::context, context, copy_socket_data(), sip_peer::defaddr, dialogs, do_setnat(), sip_peer::flags, sip_pvt::flags, sip_peer::fromdomain, sip_peer::fromuser, sip_peer::fullcontact, sip_request::headers, sip_pvt::initreq, io, t38properties::jointcapability, sip_pvt::jointnoncodeccapability, language, sip_peer::language, sip_peer::lastms, sip_peer::maxcallbitrate, sip_pvt::maxcallbitrate, sip_peer::maxms, sip_peer::md5secret, sip_peer::mohinterpret, mohinterpret, sip_peer::mohsuggest, mohsuggest, sip_peer::name, sip_pvt::noncodeccapability, obproxy_get(), sip_peer::parkinglot, parkinglot, sip_pvt::peerauth, sip_peer::pickupgroup, sip_pvt::pickupgroup, sip_peer::prefs, sip_pvt::prefs, sip_pvt::recv, ref_proxy(), sip_pvt::rtp, sip_peer::rtpholdtimeout, sip_peer::rtpkeepalive, sip_pvt::rtptimeout, sip_peer::rtptimeout, sip_pvt::sa, sched, sip_peer::secret, set_t38_capabilities(), SIP_CALL_LIMIT, SIP_DTMF, SIP_DTMF_AUTO, SIP_DTMF_RFC2833, SIP_FLAGS_TO_COPY, SIP_NAT, SIP_NAT_ROUTE, SIP_PAGE2_FLAGS_TO_COPY, SIP_PAGE2_RFC2833_COMPENSATE, SIP_PAGE2_T38SUPPORT, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_VIDEOSUPPORT, SIP_PAGE2_VIDEOSUPPORT_ALWAYS, sip_peer::socket, sip_pvt::socket, sip_pvt::t38, t38properties::t38support, sip_pvt::timer_b, sip_peer::timer_b, sip_peer::timer_t1, sip_pvt::timer_t1, sip_pvt::tohost, sip_peer::tohost, sip_pvt::trtp, sip_socket::type, sip_pvt::udptl, sip_peer::username, and sip_pvt::vrtp.

Referenced by create_addr(), and sip_send_mwi_to_peer().

{
   /* this checks that the dialog is contacting the peer on a valid
    * transport type based on the peers transport configuration,
    * otherwise, this function bails out */
   if (dialog->socket.type && check_request_transport(peer, dialog))
      return -1;
   copy_socket_data(&dialog->socket, &peer->socket);

   if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
       (!peer->maxms || ((peer->lastms >= 0)  && (peer->lastms <= peer->maxms)))) {
      dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr;
      dialog->recv = dialog->sa;
   } else 
      return -1;

   ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   dialog->capability = peer->capability;
   if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) &&
         (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) ||
            !(dialog->capability & AST_FORMAT_VIDEO_MASK)) &&
         dialog->vrtp) {
      ast_rtp_destroy(dialog->vrtp);
      dialog->vrtp = NULL;
   }
   if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT) && dialog->trtp) {
      ast_rtp_destroy(dialog->trtp);
      dialog->trtp = NULL;
   }
   dialog->prefs = peer->prefs;
   if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
      if (!dialog->udptl) {
         /* t38pt_udptl was enabled in the peer and not in [general] */
         dialog->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
      }
      ast_copy_flags(&dialog->t38.t38support, &peer->flags[1], SIP_PAGE2_T38SUPPORT);
      set_t38_capabilities(dialog);
      dialog->t38.jointcapability = dialog->t38.capability;
   } else if (dialog->udptl) {
      ast_udptl_destroy(dialog->udptl);
      dialog->udptl = NULL;
   }
   do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE);

   if (dialog->rtp) { /* Audio */
      ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
      ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
      ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
      ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
      ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
      /* Set Frame packetization */
      ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
      dialog->autoframing = peer->autoframing;
   }
   if (dialog->vrtp) { /* Video */
      ast_rtp_setdtmf(dialog->vrtp, 0);
      ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
      ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
      ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
      ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
   }
   if (dialog->trtp) { /* Realtime text */
      ast_rtp_setdtmf(dialog->trtp, 0);
      ast_rtp_setdtmfcompensate(dialog->trtp, 0);
      ast_rtp_set_rtptimeout(dialog->trtp, peer->rtptimeout);
      ast_rtp_set_rtpholdtimeout(dialog->trtp, peer->rtpholdtimeout);
      ast_rtp_set_rtpkeepalive(dialog->trtp, peer->rtpkeepalive);
   }

   ast_string_field_set(dialog, peername, peer->name);
   ast_string_field_set(dialog, authname, peer->username);
   ast_string_field_set(dialog, username, peer->username);
   ast_string_field_set(dialog, peersecret, peer->secret);
   ast_string_field_set(dialog, peermd5secret, peer->md5secret);
   ast_string_field_set(dialog, mohsuggest, peer->mohsuggest);
   ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
   ast_string_field_set(dialog, tohost, peer->tohost);
   ast_string_field_set(dialog, fullcontact, peer->fullcontact);
   ast_string_field_set(dialog, context, peer->context);
   ast_string_field_set(dialog, parkinglot, peer->parkinglot);
   ref_proxy(dialog, obproxy_get(dialog, peer));
   dialog->callgroup = peer->callgroup;
   dialog->pickupgroup = peer->pickupgroup;
   dialog->allowtransfer = peer->allowtransfer;
   dialog->jointnoncodeccapability = dialog->noncodeccapability;
   dialog->rtptimeout = peer->rtptimeout;
   dialog->peerauth = peer->auth;
   dialog->maxcallbitrate = peer->maxcallbitrate;
   if (ast_strlen_zero(dialog->tohost))
      ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
   if (!ast_strlen_zero(peer->fromdomain)) {
      ast_string_field_set(dialog, fromdomain, peer->fromdomain);
      if (!dialog->initreq.headers) {
         char *c;
         char *tmpcall = ast_strdupa(dialog->callid);
         /* this sure looks to me like we are going to change the callid on this dialog!! */
         c = strchr(tmpcall, '@');
         if (c) {
            *c = '\0';
            ao2_t_unlink(dialogs, dialog, "About to change the callid -- remove the old name");
            ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
            ao2_t_link(dialogs, dialog, "New dialog callid -- inserted back into table");
         }
      }
   }
   if (!ast_strlen_zero(peer->fromuser)) 
      ast_string_field_set(dialog, fromuser, peer->fromuser);
   if (!ast_strlen_zero(peer->language))
      ast_string_field_set(dialog, language, peer->language);
   /* Set timer T1 to RTT for this peer (if known by qualify=) */
   /* Minimum is settable or default to 100 ms */
   /* If there is a maxms and lastms from a qualify use that over a manual T1
      value. Otherwise, use the peer's T1 value. */
   if (peer->maxms && peer->lastms)
      dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
   else
      dialog->timer_t1 = peer->timer_t1;

   /* Set timer B to control transaction timeouts, the peer setting is the default and overrides
      the known timer */
   if (peer->timer_b)
      dialog->timer_b = peer->timer_b;
   else
      dialog->timer_b = 64 * dialog->timer_t1;

   if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
       (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
      dialog->noncodeccapability |= AST_RTP_DTMF;
   else
      dialog->noncodeccapability &= ~AST_RTP_DTMF;
   if (peer->call_limit)
      ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
   
   return 0;
}

static void destroy_association

(

struct sip_peer *

peer

)

[static]

Remove registration data from realtime database or AST/DB when registration expires.

Definition at line 10377 of file chan_sip.c.

References ast_check_realtime(), ast_db_del(), ast_update_realtime(), sip_peer::deprecated_username, sip_settings::ignore_regexpire, sip_peer::name, sip_settings::peer_rtupdate, sip_peer::rt_fromcontact, SENTINEL, and sip_cfg.

Referenced by build_peer(), expire_register(), and parse_register_contact().

{
   int realtimeregs = ast_check_realtime("sipregs");
   char *tablename = (realtimeregs) ? "sipregs" : "sippeers";

   if (!sip_cfg.ignore_regexpire) {
      if (peer->rt_fromcontact && sip_cfg.peer_rtupdate) {
         ast_update_realtime(tablename, "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", peer->deprecated_username ? "username" : "defaultuser", "", "regserver", "", "useragent", "", "lastms", "", SENTINEL);
      } else {
         ast_db_del("SIP/Registry", peer->name);
      }
   }
}

static void destroy_mailbox

(

struct sip_mailbox *

mailbox

)

[static]

Destroy mailbox subscriptions

Definition at line 3831 of file chan_sip.c.

References ast_event_unsubscribe(), ast_free, and mailbox.

Referenced by clear_peer_mailboxes().

{
   if (mailbox->mailbox)
      ast_free(mailbox->mailbox);
   if (mailbox->context)
      ast_free(mailbox->context);
   if (mailbox->event_sub)
      ast_event_unsubscribe(mailbox->event_sub);
   ast_free(mailbox);
}

static int determine_firstline_parts

(

struct sip_request *

req

)

[static]

Parse first line of incoming SIP request.

Definition at line 9026 of file chan_sip.c.

References ast_debug, ast_skip_blanks(), ast_skip_nonblanks(), ast_trim_blanks(), sip_request::data, sip_request::rlPart1, sip_request::rlPart2, and ast_str::str.

{
   char *e = ast_skip_blanks(req->data->str);   /* there shouldn't be any */
   char *local_rlPart1;

   if (!*e)
      return -1;
   req->rlPart1 = e - req->data->str;  /* method or protocol */
   local_rlPart1 = e;
   e = ast_skip_nonblanks(e);
   if (*e)
      *e++ = '\0';
   /* Get URI or status code */
   e = ast_skip_blanks(e);
   if ( !*e )
      return -1;
   ast_trim_blanks(e);

   if (!strcasecmp(local_rlPart1, "SIP/2.0") ) { /* We have a response */
      if (strlen(e) < 3)   /* status code is 3 digits */
         return -1;
      req->rlPart2 = e - req->data->str;
   } else { /* We have a request */
      if ( *e == '<' ) { /* XXX the spec says it must not be in <> ! */
         ast_debug(3, "Oops. Bogus uri in <> %s\n", e);
         e++;
         if (!*e)
            return -1; 
      }
      req->rlPart2 = e - req->data->str;  /* URI */
      e = ast_skip_nonblanks(e);
      if (*e)
         *e++ = '\0';
      e = ast_skip_blanks(e);
      if (strcasecmp(e, "SIP/2.0") ) {
         ast_debug(3, "Skipping packet - Bad request protocol %s\n", e);
         return -1;
      }
   }
   return 1;
}

static int dialog_cmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Note:

The only member of the dialog used here callid string

Definition at line 1749 of file chan_sip.c.

References sip_pvt::callid, CMP_MATCH, and CMP_STOP.

Referenced by load_module().

{
   struct sip_pvt *pvt = obj, *pvt2 = arg;
   
   return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH | CMP_STOP : 0;
}

static int dialog_dump_func	(	void *	userobj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 13125 of file chan_sip.c.

References ao2_t_ref, ast_cli(), and sip_pvt::callid.

Referenced by sip_show_objects().

{
   struct sip_pvt *pvt = userobj;
   int refc = ao2_t_ref(userobj, 0, "");
   int *fd = arg;
   
   ast_cli(*fd, "name: %s\ntype: dialog\nobjflags: %d\nrefcount: %d\n\n", 
          pvt->callid, 0, refc);
   return 0;
}

static int dialog_hash_cb	(	const void *	obj,
		const int	flags
	)			[static]

Note:

The only member of the dialog used here callid string

Definition at line 1739 of file chan_sip.c.

References ast_str_case_hash(), and sip_pvt::callid.

Referenced by load_module().

{
   const struct sip_pvt *pvt = obj;

   return ast_str_case_hash(pvt->callid);
}

static int dialog_needdestroy	(	void *	dialogobj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 13240 of file chan_sip.c.

References ao2_lock(), ao2_unlock(), ast_debug, ast_rtp_get_bridged(), sip_pvt::callid, check_rtp_timeout(), dialog_unlink_all(), dialogs, FALSE, sip_pvt::method, sip_pvt::needdestroy, sip_pvt::owner, sip_pvt::packets, sip_pvt::rtp, sip_methods, sip_pvt_trylock, sip_pvt_unlock, TRUE, and sip_pvt::vrtp.

Referenced by do_monitor().

{
   struct sip_pvt *dialog = dialogobj;
   time_t *t = arg;
   
   /* log_show_lock(ao2_object_get_lockaddr(dialog)); this is an example of how to use log_show_lock() */

   if (sip_pvt_trylock(dialog)) {
      /* In very short-duration calls via sipp,
         this path gets executed fairly frequently (3% or so) even in low load
         situations; the routines we most commonly fight for a lock with:
         sip_answer (7 out of 9)
         sip_hangup (2 out of 9)
      */
      ao2_unlock(dialogs);
      usleep(1);
      ao2_lock(dialogs);

      /* I had previously returned CMP_STOP here; but changed it to return
         a zero instead, because there is no need to stop at the first sign
         of trouble. The old algorithm would restart in such circumstances,
         but there is no need for this now in astobj2-land. We'll
         just skip over this element this time, and catch it on the
         next traversal, which is about once a second or so. See the 
         ao2_callback call in do_monitor. We don't want the number of
         dialogs to back up too much between runs.
      */
      return 0;
   }
   
   /* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */
   check_rtp_timeout(dialog, *t);

   /* If we have sessions that needs to be destroyed, do it now */
   /* Check if we have outstanding requests not responsed to or an active call
      - if that's the case, wait with destruction */
   if (dialog->needdestroy && !dialog->packets && !dialog->owner) {
      /* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
      if (dialog->rtp && ast_rtp_get_bridged(dialog->rtp)) {
         ast_debug(2, "Bridge still active.  Delaying destruction of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
         sip_pvt_unlock(dialog);
         return 0;
      }
      
      if (dialog->vrtp && ast_rtp_get_bridged(dialog->vrtp)) {
         ast_debug(2, "Bridge still active.  Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
         sip_pvt_unlock(dialog);
         return 0;
      }

      sip_pvt_unlock(dialog);
      /* no, the unlink should handle this: dialog_unref(dialog, "needdestroy: one more refcount decrement to allow dialog to be destroyed"); */
      /* the CMP_MATCH will unlink this dialog from the dialog hash table */
      dialog_unlink_all(dialog, TRUE, FALSE);
      return 0; /* the unlink_all should unlink this from the table, so.... no need to return a match */
   }

   sip_pvt_unlock(dialog);

   return 0;
}

static struct sip_pvt* dialog_ref	(	struct sip_pvt *	p,
		char *	tag
	)			[static]

when we create or delete references, make sure to use these functions so we keep track of the refcounts. To simplify the code, we allow a NULL to be passed to dialog_unref().

Definition at line 1450 of file chan_sip.c.

References ao2_ref, ast_log(), and LOG_ERROR.

Referenced by __sip_reliable_xmit(), dialog_unlink_all(), handle_request_invite(), handle_request_subscribe(), handle_response_invite(), manager_sipnotify(), queue_request(), sip_call(), sip_cli_notify(), sip_new(), sip_poke_peer(), sip_scheddestroy(), sip_send_mwi_to_peer(), start_session_timer(), and transmit_register().

{
   if (p)
      ao2_ref(p, 1);
   else
      ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
   return p;
}

static void * dialog_unlink_all	(	struct sip_pvt *	dialog,
		int	lockowner,
		int	lockdialoglist
	)			[static]

Unlink a dialog from the dialogs container, as well as any other places that it may be currently stored.

Note:

A reference to the dialog must be held before calling this function, and this function does not release that reference.

Definition at line 2535 of file chan_sip.c.

References ao2_t_unlink, ast_channel_lock, ast_channel_unlock, ast_debug, ast_extension_state_del(), ast_free, AST_SCHED_DEL, AST_SCHED_DEL_UNREF, sip_pvt::autokillid, sip_peer::call, sip_registry::call, dialog_ref(), dialog_unref(), dialogs, sip_pvt::initid, sip_peer::mwipvt, ast_channel::name, sip_pkt::next, sip_pkt::owner, sip_pvt::owner, sip_pvt::packets, sip_pvt::registry, registry_unref(), sip_pvt::relatedpeer, sip_pvt::request_queue_sched_id, sip_pkt::retransid, sched, sip_pvt::stateid, sip_pvt::t38id, ast_channel::tech_pvt, and sip_pvt::waitid.

Referenced by __sip_autodestruct(), dialog_needdestroy(), handle_request_subscribe(), manager_sipnotify(), reload_config(), sip_cli_notify(), sip_destroy_peer(), sip_poke_noanswer(), sip_poke_peer(), sip_registry_destroy(), sip_request_call(), sip_send_mwi_to_peer(), transmit_register(), and unload_module().

{
   struct sip_pkt *cp;

   dialog_ref(dialog, "Let's bump the count in the unlink so it doesn't accidentally become dead before we are done");

   ao2_t_unlink(dialogs, dialog, "unlinking dialog via ao2_unlink");

   /* Unlink us from the owner (channel) if we have one */
   if (dialog->owner) {
      if (lockowner)
         ast_channel_lock(dialog->owner);
      ast_debug(1, "Detaching from channel %s\n", dialog->owner->name);
      dialog->owner->tech_pvt = dialog_unref(dialog->owner->tech_pvt, "resetting channel dialog ptr in unlink_all");
      if (lockowner)
         ast_channel_unlock(dialog->owner);
   }
   if (dialog->registry) {
      if (dialog->registry->call == dialog)
         dialog->registry->call = dialog_unref(dialog->registry->call, "nulling out the registry's call dialog field in unlink_all");
      dialog->registry = registry_unref(dialog->registry, "delete dialog->registry");
   }
   if (dialog->stateid > -1) {
      ast_extension_state_del(dialog->stateid, NULL);
      dialog_unref(dialog, "removing extension_state, should unref the associated dialog ptr that was stored there.");
      dialog->stateid = -1; /* shouldn't we 'zero' this out? */
   }
   /* Remove link from peer to subscription of MWI */
   if (dialog->relatedpeer && dialog->relatedpeer->mwipvt == dialog)
      dialog->relatedpeer->mwipvt = dialog_unref(dialog->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
   if (dialog->relatedpeer && dialog->relatedpeer->call == dialog)
      dialog->relatedpeer->call = dialog_unref(dialog->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");

   /* remove all current packets in this dialog */
   while((cp = dialog->packets)) {
      dialog->packets = dialog->packets->next;
      AST_SCHED_DEL(sched, cp->retransid);
      dialog_unref(cp->owner, "remove all current packets in this dialog, and the pointer to the dialog too as part of __sip_destroy");
      ast_free(cp);
   }

   AST_SCHED_DEL_UNREF(sched, dialog->waitid, dialog_unref(dialog, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));

   AST_SCHED_DEL_UNREF(sched, dialog->initid, dialog_unref(dialog, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
   
   if (dialog->autokillid > -1)
      AST_SCHED_DEL_UNREF(sched, dialog->autokillid, dialog_unref(dialog, "when you delete the autokillid sched, you should dec the refcount for the stored dialog ptr"));

   if (dialog->request_queue_sched_id > -1) {
      AST_SCHED_DEL_UNREF(sched, dialog->request_queue_sched_id, dialog_unref(dialog, "when you delete the request_queue_sched_id sched, you should dec the refcount for the stored dialog ptr"));
   }

   if (dialog->t38id > -1) {
      AST_SCHED_DEL_UNREF(sched, dialog->t38id, dialog_unref(dialog, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
   }

   dialog_unref(dialog, "Let's unbump the count in the unlink so the poor pvt can disappear if it is time");
   return NULL;
}

static struct sip_pvt* dialog_unref	(	struct sip_pvt *	p,
		char *	tag
	)			[static]

Definition at line 1459 of file chan_sip.c.

References ao2_ref.

Referenced by __sip_ack(), __sip_autodestruct(), __sip_destroy(), __sip_reliable_xmit(), auto_congest(), complete_sipch(), dialog_unlink_all(), get_also_info(), handle_invite_replaces(), handle_request_do(), handle_request_invite(), handle_request_subscribe(), handle_response_invite(), handle_response_peerpoke(), handle_response_register(), manager_sipnotify(), proc_session_timer(), queue_request(), reload_config(), restart_session_timer(), retrans_pkt(), sip_call(), sip_cancel_destroy(), sip_cli_notify(), sip_destroy_peer(), sip_hangup(), sip_indicate(), sip_poke_noanswer(), sip_poke_peer(), sip_reg_timeout(), sip_registry_destroy(), sip_reinvite_retry(), sip_request_call(), sip_send_mwi_to_peer(), sip_t38_abort(), start_session_timer(), stop_session_timer(), and transmit_register().

{
   if (p)
      ao2_ref(p, -1);
   return NULL;
}

static void * do_monitor

(

void *

data

)

[static]

The SIP monitoring thread.

Note:

This thread monitors all the SIP sessions and peers that needs notification of mwi (and thus do not have a separate thread) indefinitely

Definition at line 20347 of file chan_sip.c.

References ao2_t_callback, ast_debug, ast_io_add(), ast_io_change(), AST_IO_IN, ast_io_remove(), ast_io_wait(), ast_mutex_lock(), ast_mutex_unlock(), ast_sched_runq(), ast_sched_wait(), ast_verb, dialog_needdestroy(), dialogs, FALSE, io, monlock, OBJ_MULTIPLE, OBJ_NODATA, OBJ_UNLINK, sched, sip_do_reload(), sip_reload_lock, sipsock, and sipsock_read().

{
   int res;
   time_t t;
   int reloading;

   /* Add an I/O event to our SIP UDP socket */
   if (sipsock > -1) 
      sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);

   /* From here on out, we die whenever asked */
   for(;;) {
      /* Check for a reload request */
      ast_mutex_lock(&sip_reload_lock);
      reloading = sip_reloading;
      sip_reloading = FALSE;
      ast_mutex_unlock(&sip_reload_lock);
      if (reloading) {
         ast_verb(1, "Reloading SIP\n");
         sip_do_reload(sip_reloadreason);

         /* Change the I/O fd of our UDP socket */
         if (sipsock > -1) {
            if (sipsock_read_id)
               sipsock_read_id = ast_io_change(io, sipsock_read_id, sipsock, NULL, 0, NULL);
            else
               sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);
         } else if (sipsock_read_id) {
            ast_io_remove(io, sipsock_read_id);
            sipsock_read_id = NULL;
         }
      }

      /* Check for dialogs needing to be killed */
      t = time(NULL);
      /* don't scan the dialogs list if it hasn't been a reasonable period
         of time since the last time we did it (when MWI is being sent, we can
         get back to this point every millisecond or less)
      */
      ao2_t_callback(dialogs, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, dialog_needdestroy, &t, 
            "callback to remove dialogs w/needdestroy");

      /* the old methodology would be to restart the search for dialogs to delete with every 
         dialog that was found and destroyed, probably because the list contents would change,
         so we'd need to restart. This isn't the best thing to do with callbacks. */

      /* XXX TODO The scheduler usage in this module does not have sufficient 
       * synchronization being done between running the scheduler and places 
       * scheduling tasks.  As it is written, any scheduled item may not run 
       * any sooner than about  1 second, regardless of whether a sooner time 
       * was asked for. */

      pthread_testcancel();
      /* Wait for sched or io */
      res = ast_sched_wait(sched);
      if ((res < 0) || (res > 1000))
         res = 1000;
      res = ast_io_wait(io, res);
      if (res > 20)
         ast_debug(1, "chan_sip: ast_io_wait ran %d all at once\n", res);
      ast_mutex_lock(&monlock);
      res = ast_sched_runq(sched);
      if (res >= 20)
         ast_debug(1, "chan_sip: ast_sched_runq ran %d all at once\n", res);
      ast_mutex_unlock(&monlock);
   }

   /* Never reached */
   return NULL;
}

static int do_proxy_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum sip_auth_type	code,
		int	sipmethod,
		int	init
	)			[static]

Add authentication on outbound SIP packet.

Definition at line 15262 of file chan_sip.c.

References ast_calloc, ast_debug, sip_invite_param::auth, auth_headers(), sip_invite_param::authheader, sip_pvt::authtries, sip_pvt::options, reply_digest(), SIP_INVITE, sip_methods, cfsip_methods::text, and transmit_invite().

Referenced by handle_response(), handle_response_invite(), handle_response_notify(), and handle_response_refer().

{
   char *header, *respheader;
   char digest[1024];

   if (!p->options && !(p->options = ast_calloc(1, sizeof(*p->options))))
      return -2;

   p->authtries++;
   auth_headers(code, &header, &respheader);
   ast_debug(2, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text);
   memset(digest, 0, sizeof(digest));
   if (reply_digest(p, req, header, sipmethod, digest, sizeof(digest) )) {
      /* No way to authenticate */
      return -1;
   }
   /* Now we have a reply digest */
   p->options->auth = digest;
   p->options->authheader = respheader;
   return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, init); 
}

static int do_register_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum sip_auth_type	code
	)			[static]

Authenticate for outbound registration.

Definition at line 15238 of file chan_sip.c.

References append_history, ast_verbose, auth_headers(), sip_pvt::authtries, sip_pvt::do_history, sip_registry::hostname, sip_pvt::registry, reply_digest(), sip_debug_test_pvt(), SIP_REGISTER, and transmit_register().

Referenced by handle_response_register().

{
   char *header, *respheader;
   char digest[1024];

   p->authtries++;
   auth_headers(code, &header, &respheader);
   memset(digest, 0, sizeof(digest));
   if (reply_digest(p, req, header, SIP_REGISTER, digest, sizeof(digest))) {
      /* There's nothing to use for authentication */
      /* No digest challenge in request */
      if (sip_debug_test_pvt(p) && p->registry)
         ast_verbose("No authentication challenge, sending blank registration to domain/host name %s\n", p->registry->hostname);
         /* No old challenge */
      return -1;
   }
   if (p->do_history)
      append_history(p, "RegistryAuth", "Try: %d", p->authtries);
   if (sip_debug_test_pvt(p) && p->registry)
      ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
   return transmit_register(p->registry, SIP_REGISTER, digest, respheader); 
}

static void do_setnat	(	struct sip_pvt *	p,
		int	natflags
	)			[static]

Set nat mode on the various data sockets.

Definition at line 4213 of file chan_sip.c.

References ast_debug, ast_rtp_setnat(), ast_udptl_setnat(), sip_pvt::rtp, sip_pvt::trtp, sip_pvt::udptl, and sip_pvt::vrtp.

Referenced by check_peer_ok(), create_addr(), create_addr_from_peer(), sip_alloc(), and transmit_response_using_temp().

{
   const char *mode = natflags ? "On" : "Off";

   if (p->rtp) {
      ast_debug(1, "Setting NAT on RTP to %s\n", mode);
      ast_rtp_setnat(p->rtp, natflags);
   }
   if (p->vrtp) {
      ast_debug(1, "Setting NAT on VRTP to %s\n", mode);
      ast_rtp_setnat(p->vrtp, natflags);
   }
   if (p->udptl) {
      ast_debug(1, "Setting NAT on UDPTL to %s\n", mode);
      ast_udptl_setnat(p->udptl, natflags);
   }
   if (p->trtp) {
      ast_debug(1, "Setting NAT on TRTP to %s\n", mode);
      ast_rtp_setnat(p->trtp, natflags);
   }
}

static const char * domain_mode_to_text

(

const enum domain_mode

mode

)

[static]

Print domain mode to cli.

Definition at line 13457 of file chan_sip.c.

References SIP_DOMAIN_AUTO, and SIP_DOMAIN_CONFIG.

Referenced by sip_show_domains().

{
   switch (mode) {
   case SIP_DOMAIN_AUTO:
      return "[Automatic]";
   case SIP_DOMAIN_CONFIG:
      return "[Configured]";
   }

   return "";
}

static const char * dtmfmode2str

(

int

mode

)

const [static]

Convert DTMF mode to printable string.

Definition at line 13181 of file chan_sip.c.

References dtmfstr, and map_x_s().

Referenced by _sip_show_peer(), sip_show_channel(), and sip_show_settings().

{
   return map_x_s(dtmfstr, mode, "<error>");
}

static int expire_register

(

const void *

data

)

[static]

Expire registration of SIP peer.

Definition at line 10392 of file chan_sip.c.

References sip_peer::addr, ao2_t_unlink, ast_debug, AST_DEVICE_UNKNOWN, ast_devstate_changed(), ast_test_flag, destroy_association(), EVENT_FLAG_SYSTEM, sip_peer::expire, FALSE, sip_peer::flags, sip_peer::is_realtime, manager_event, sip_peer::name, peers_by_ip, register_peer_exten(), sip_peer::selfdestruct, SIP_PAGE2_RTAUTOCLEAR, and unref_peer().

Referenced by parse_register_contact(), reg_source_db(), sip_show_sched(), and sip_unregister().

{
   struct sip_peer *peer = (struct sip_peer *)data;
   
   if (!peer)     /* Hmmm. We have no peer. Weird. */
      return 0;

   peer->expire = -1;
   memset(&peer->addr, 0, sizeof(peer->addr));

   destroy_association(peer); /* remove registration data from storage */
   
   manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
   register_peer_exten(peer, FALSE);   /* Remove regexten */
   ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);

   /* Do we need to release this peer from memory? 
      Only for realtime peers and autocreated peers
   */
   if (peer->is_realtime)
      ast_debug(3, "-REALTIME- peer expired registration. Name: %s. Realtime peer objects now %d\n", peer->name, rpeerobjs);

   if (peer->selfdestruct ||
       ast_test_flag(&peer->flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
      ao2_t_unlink(peers, peer, "ao2_unlink of peer from peers table");
      if (peer->addr.sin_addr.s_addr) {
         ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
      }
   }

   unref_peer(peer, "removing peer ref for expire_register");

   return 0;
}

static void extract_uri	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Check Contact: URI of SIP message.

Definition at line 9125 of file chan_sip.c.

References ast_copy_string(), ast_string_field_set, ast_strlen_zero(), get_header(), get_in_brackets(), remove_uri_parameters(), and SIPBUFSIZE.

Referenced by handle_incoming(), and handle_request_invite().

{
   char stripped[SIPBUFSIZE];
   char *c;

   ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped));
   c = get_in_brackets(stripped);
   /* Cut the URI at the at sign after the @, not in the username part */
   c = remove_uri_parameters(c);
   if (!ast_strlen_zero(c))
      ast_string_field_set(p, uri, c);

}

static const char * find_alias	(	const char *	name,
		const char *	_default
	)			[static]

Find compressed SIP alias.

Structure for conversion between compressed SIP and "normal" SIP

Definition at line 5949 of file chan_sip.c.

References aliases, and ARRAY_LEN.

{
   /*! \brief Structure for conversion between compressed SIP and "normal" SIP */
   static const struct cfalias {
      char * const fullname;
      char * const shortname;
   } aliases[] = {
      { "Content-Type",  "c" },
      { "Content-Encoding",    "e" },
      { "From",       "f" },
      { "Call-ID",       "i" },
      { "Contact",       "m" },
      { "Content-Length",   "l" },
      { "Subject",       "s" },
      { "To",         "t" },
      { "Supported",     "k" },
      { "Refer-To",      "r" },
      { "Referred-By",   "b" },
      { "Allow-Events",  "u" },
      { "Event",      "o" },
      { "Via",     "v" },
      { "Accept-Contact",      "a" },
      { "Reject-Contact",      "j" },
      { "Request-Disposition", "d" },
      { "Session-Expires",     "x" },
      { "Identity",            "y" },
      { "Identity-Info",       "n" },
   };
   int x;

   for (x = 0; x < ARRAY_LEN(aliases); x++) {
      if (!strcasecmp(aliases[x].fullname, name))
         return aliases[x].shortname;
   }

   return _default;
}

static int find_by_name	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 4140 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, FINDALLDEVICES, FINDPEERS, FINDUSERS, sip_peer::name, SIP_TYPE_PEER, SIP_TYPE_USER, peer_finding_info::tmp_peer, sip_peer::type, and peer_finding_info::which_objects.

Referenced by find_peer().

{
   struct sip_peer *search = obj;
   struct peer_finding_info *pfi = arg;

   /* Usernames in SIP uri's are case sensitive. Domains are not */
   if (strcmp(search->name, pfi->tmp_peer.name)) {
      return 0;
   }

   switch (pfi->which_objects) {
   case FINDUSERS:
      if (!(search->type & SIP_TYPE_USER)) {
         return 0;
      }
      break;
   case FINDPEERS:
      if (!(search->type & SIP_TYPE_PEER)) {
         return 0;
      }
      break;
   case FINDALLDEVICES:
      break;
   }

   return CMP_MATCH | CMP_STOP;
}

static struct sip_pvt * find_call	(	struct sip_request *	req,
		struct sockaddr_in *	sin,
		const int	intended_method
	)			[static]

find or create a dialog structure for an incoming SIP message. Connect incoming SIP message to current dialog or create new dialog structure Returns a reference to the sip_pvt object, remember to give it back once done. Called by handle_incoming(), sipsock_read

Definition at line 6404 of file chan_sip.c.

References ao2_lock(), ao2_t_callback, ao2_t_find, ao2_unlock(), ast_debug, ast_strlen_zero(), sip_pvt::callid, find_call_cb_arg::callid, CAN_CREATE_DIALOG, dialogs, find_call_cb(), find_call_cb_arg::fromtag, get_header(), gettag(), sip_request::method, find_call_cb_arg::method, OBJ_POINTER, SIP_ACK, sip_alloc(), SIP_BYE, SIP_INFO, sip_methods, SIP_NOTIFY, sip_pvt_lock, sip_pvt_trylock, SIP_REFER, SIP_RESPONSE, find_call_cb_arg::tag, cfsip_methods::text, find_call_cb_arg::totag, and transmit_response_using_temp().

Referenced by handle_request_do().

{
   struct sip_pvt *p = NULL;
   char *tag = "";   /* note, tag is never NULL */
   char totag[128];
   char fromtag[128];
   struct find_call_cb_arg arg;
   const char *callid = get_header(req, "Call-ID");
   const char *from = get_header(req, "From");
   const char *to = get_header(req, "To");
   const char *cseq = get_header(req, "Cseq");
   struct sip_pvt *sip_pvt_ptr;

   /* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
   /* get_header always returns non-NULL so we must use ast_strlen_zero() */
   if (ast_strlen_zero(callid) || ast_strlen_zero(to) ||
         ast_strlen_zero(from) || ast_strlen_zero(cseq))
      return NULL;   /* Invalid packet */

   arg.method = req->method;
   arg.callid = callid;
   arg.fromtag = fromtag;
   arg.totag = totag;
   arg.tag = ""; /* make sure tag is never NULL */

   if (pedanticsipchecking) {
      /* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
         we need more to identify a branch - so we have to check branch, from
         and to tags to identify a call leg.
         For Asterisk to behave correctly, you need to turn on pedanticsipchecking
         in sip.conf
         */
      if (gettag(req, "To", totag, sizeof(totag)))
         req->has_to_tag = 1; /* Used in handle_request/response */
      gettag(req, "From", fromtag, sizeof(fromtag));

      tag = (req->method == SIP_RESPONSE) ? totag : fromtag;

      ast_debug(5, "= Looking for  Call ID: %s (Checking %s) --From tag %s --To-tag %s  \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);

      /* All messages must always have From: tag */
      if (ast_strlen_zero(fromtag)) {
         ast_debug(5, "%s request has no from tag, dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
         return NULL;
      }
      /* reject requests that must always have a To: tag */
      if (ast_strlen_zero(totag) && (req->method == SIP_ACK || req->method == SIP_BYE || req->method == SIP_INFO )) {
         ast_debug(5, "%s must have a to tag. dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
         return NULL;
      }
   }

restartsearch:
   if (!pedanticsipchecking) {
      struct sip_pvt tmp_dialog = {
         .callid = callid,
      };       
      sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find in dialogs");
      if (sip_pvt_ptr) {  /* well, if we don't find it-- what IS in there? */
         /* Found the call */
         sip_pvt_lock(sip_pvt_ptr);
         return sip_pvt_ptr;
      }
   } else { /* in pedantic mode! -- do the fancy linear search */
      ao2_lock(dialogs);
      p = ao2_t_callback(dialogs, 0 /* single, data */, find_call_cb, &arg, "pedantic linear search for dialog");
      if (p) {
         if (sip_pvt_trylock(p)) {
            ao2_unlock(dialogs);
            usleep(1);
            goto restartsearch;
         }
         ao2_unlock(dialogs);
         return p;
      }
      ao2_unlock(dialogs);
   }
 
   /* See if the method is capable of creating a dialog */
   if (sip_methods[intended_method].can_create == CAN_CREATE_DIALOG) {
      if (intended_method == SIP_REFER) {
         /* We do support REFER, but not outside of a dialog yet */
         transmit_response_using_temp(callid, sin, 1, intended_method, req, "603 Declined (no dialog)");
      } else if (intended_method == SIP_NOTIFY) {
         /* We do not support out-of-dialog NOTIFY either,
            like voicemail notification, so cancel that early */
         transmit_response_using_temp(callid, sin, 1, intended_method, req, "489 Bad event");
      } else {
         /* Ok, time to create a new SIP dialog object, a pvt */
         if ((p = sip_alloc(callid, sin, 1, intended_method)))  {
            /* Ok, we've created a dialog, let's go and process it */
            sip_pvt_lock(p);
         } else {
            /* We have a memory or file/socket error (can't allocate RTP sockets or something) so we're not
               getting a dialog from sip_alloc. 
   
               Without a dialog we can't retransmit and handle ACKs and all that, but at least
               send an error message.
   
               Sorry, we apologize for the inconvienience
            */
            transmit_response_using_temp(callid, sin, 1, intended_method, req, "500 Server internal error");
            ast_debug(4, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n");
         }
      }
      return p; /* can be NULL */
   } else if( sip_methods[intended_method].can_create == CAN_CREATE_DIALOG_UNSUPPORTED_METHOD) {
      /* A method we do not support, let's take it on the volley */
      transmit_response_using_temp(callid, sin, 1, intended_method, req, "501 Method Not Implemented");
      ast_debug(2, "Got a request with unsupported SIP method.\n");
   } else if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
      /* This is a request outside of a dialog that we don't know about */
      transmit_response_using_temp(callid, sin, 1, intended_method, req, "481 Call leg/transaction does not exist");
      ast_debug(2, "That's odd...  Got a request in unknown dialog. Callid %s\n", callid ? callid : "<unknown>");
   }
   /* We do not respond to responses for dialogs that we don't know about, we just drop
      the session quickly */
   if (intended_method == SIP_RESPONSE)
      ast_debug(2, "That's odd...  Got a response on a call we dont know about. Callid %s\n", callid ? callid : "<unknown>");

   return NULL;
}

static int find_call_cb	(	void *	__pvt,
		void *	__arg,
		int	flags
	)			[static]

code to determine whether this is the pvt that we are looking for. Return FALSE if not found, true otherwise. p is unlocked.

Definition at line 6363 of file chan_sip.c.

References ast_debug, ast_strlen_zero(), ast_test_flag, find_call_cb_arg::callid, sip_pvt::callid, FALSE, sip_pvt::flags, find_call_cb_arg::method, sip_methods, SIP_PAGE2_DIALOG_ESTABLISHED, SIP_REGISTER, SIP_RESPONSE, sip_pvt::tag, find_call_cb_arg::tag, sip_pvt::theirtag, and find_call_cb_arg::totag.

Referenced by find_call().

{
   struct sip_pvt *p = __pvt;
   struct find_call_cb_arg *arg = __arg;
   /* In pedantic, we do not want packets with bad syntax to be connected to a PVT */
   int found = FALSE;
   
   if (!ast_strlen_zero(p->callid)) { /* XXX double check, do we allow match on empty p->callid ? */
      if (arg->method == SIP_REGISTER)
         found = (!strcmp(p->callid, arg->callid));
      else {
         found = !strcmp(p->callid, arg->callid);
         if (pedanticsipchecking && found) {
            found = ast_strlen_zero(arg->tag) || ast_strlen_zero(p->theirtag) || !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED) || !strcmp(p->theirtag, arg->tag);
         } 
      }
      
      ast_debug(5, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag);
      
      /* If we get a new request within an existing to-tag - check the to tag as well */
      if (pedanticsipchecking && found && arg->method != SIP_RESPONSE) { /* SIP Request */
         if (p->tag[0] == '\0' && arg->totag[0]) {
            /* We have no to tag, but they have. Wrong dialog */
            found = FALSE;
         } else if (arg->totag[0]) { /* Both have tags, compare them */
            if (strcmp(arg->totag, p->tag)) {
               found = FALSE; /* This is not our packet */
            }
         }
         if (!found)
            ast_debug(5, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, arg->totag, sip_methods[arg->method].text);
      }
   }
   return found;
}

static const char* find_closing_quote	(	const char *	start,
		const char *	lim
	)			[static]

Locate closing quote in a string, skipping escaped quotes. optionally with a limit on the search. start must be past the first quote.

Definition at line 3545 of file chan_sip.c.

References s.

Referenced by get_in_brackets().

{
   char last_char = '\0';
   const char *s;
   for (s = start; *s && s != lim; last_char = *s++) {
      if (*s == '"' && last_char != '\\')
         break;
   }
   return s;
}

static struct sip_peer * find_peer	(	const char *	peer,
		struct sockaddr_in *	sin,
		int	realtime,
		int	which_objects,
		int	devstate_only
	)			[static]

Locate device by name or ip address.

Parameters:

which_objects

Define which objects should be matched when doing a lookup by name. Valid options are FINDUSERS, FINDPEERS, or FINDALLDEVICES. Note that this option is not used at all when doing a lookup by IP.

This is used on find matching device on name or ip/port. If the device was declared as type=peer, we don't match on peer name on incoming INVITEs.

Note:

Avoid using this function in new functions if there is a way to avoid it, since it might cause a database lookup.

Definition at line 4181 of file chan_sip.c.

References sip_peer::addr, ao2_t_callback, ao2_t_find, ast_copy_string(), ast_set_flag, find_by_name(), ast_flags::flags, sip_peer::flags, sip_peer::name, OBJ_POINTER, peers_by_ip, realtime_peer(), SIP_INSECURE_PORT, peer_finding_info::tmp_peer, and peer_finding_info::which_objects.

{
   struct sip_peer *p = NULL;

   if (peer) {
      struct peer_finding_info pfi = {
         .which_objects = which_objects,
      };
      ast_copy_string(pfi.tmp_peer.name, peer, sizeof(pfi.tmp_peer.name));
      p = ao2_t_callback(peers, OBJ_POINTER, find_by_name, &pfi, "ao2_callback in peers table");
   } else if (sin) { /* search by addr? */
      struct sip_peer tmp_peer;
      tmp_peer.addr.sin_addr.s_addr = sin->sin_addr.s_addr;
      tmp_peer.addr.sin_port = sin->sin_port;
      tmp_peer.flags[0].flags = 0;
      p = ao2_t_find(peers_by_ip, &tmp_peer, OBJ_POINTER, "ao2_find in peers_by_ip table"); /* WAS:  p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp); */
      if (!p) {
         ast_set_flag(&tmp_peer.flags[0], SIP_INSECURE_PORT);
         p = ao2_t_find(peers_by_ip, &tmp_peer, OBJ_POINTER, "ao2_find in peers_by_ip table 2"); /* WAS:  p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp); */
         if (p) {
            return p;
         }
      }
   }

   if (!p && (realtime || devstate_only))
      p = realtime_peer(peer, sin, devstate_only);

   return p;
}

static struct sip_auth * find_realm_authentication	(	struct sip_auth *	authlist,
		const char *	realm
	)			[static]

Find authentication for a specific realm.

Definition at line 21441 of file chan_sip.c.

References sip_auth::next, and sip_auth::realm.

Referenced by build_reply_digest().

{
   struct sip_auth *a;

   for (a = authlist; a; a = a->next) {
      if (!strcasecmp(a->realm, realm))
         break;
   }

   return a;
}

static int find_sdp

(

struct sip_request *

req

)

[static]

Determine whether a SIP message contains an SDP in its body.

Parameters:

req

the SIP request to process

Returns:

1 if SDP found, 0 if not found

Also updates req->sdp_start and req->sdp_end to indicate where the SDP lives in the message body.

Definition at line 6755 of file chan_sip.c.

References ast_log(), ast_strdupa, ast_strlen_zero(), FALSE, get_header(), sip_request::lines, LOG_WARNING, REQ_OFFSET_TO_STR, sip_request::sdp_end, sip_request::sdp_start, strcasestr(), and TRUE.

Referenced by handle_incoming(), handle_request_invite(), handle_response(), and handle_response_invite().

{
   const char *content_type;
   const char *content_length;
   const char *search;
   char *boundary;
   unsigned int x;
   int boundaryisquoted = FALSE;
   int found_application_sdp = FALSE;
   int found_end_of_headers = FALSE;

   content_length = get_header(req, "Content-Length");

   if (!ast_strlen_zero(content_length)) {
      if (sscanf(content_length, "%ud", &x) != 1) {
         ast_log(LOG_WARNING, "Invalid Content-Length: %s\n", content_length);
         return 0;
      }

      /* Content-Length of zero means there can't possibly be an
         SDP here, even if the Content-Type says there is */
      if (x == 0)
         return 0;
   }

   content_type = get_header(req, "Content-Type");

   /* if the body contains only SDP, this is easy */
   if (!strncasecmp(content_type, "application/sdp", 15)) {
      req->sdp_start = 0;
      req->sdp_end = req->lines;
      return req->lines ? 1 : 0;
   }

   /* if it's not multipart/mixed, there cannot be an SDP */
   if (strncasecmp(content_type, "multipart/mixed", 15))
      return 0;

   /* if there is no boundary marker, it's invalid */
   if ((search = strcasestr(content_type, ";boundary=")))
      search += 10;
   else if ((search = strcasestr(content_type, "; boundary=")))
      search += 11;
   else
      return 0;

   if (ast_strlen_zero(search))
      return 0;

   /* If the boundary is quoted with ", remove quote */
   if (*search == '\"')  {
      search++;
      boundaryisquoted = TRUE;
   }

   /* make a duplicate of the string, with two extra characters
      at the beginning */
   boundary = ast_strdupa(search - 2);
   boundary[0] = boundary[1] = '-';
   /* Remove final quote */
   if (boundaryisquoted)
      boundary[strlen(boundary) - 1] = '\0';

   /* search for the boundary marker, the empty line delimiting headers from
      sdp part and the end boundry if it exists */

   for (x = 0; x < (req->lines); x++) {
      char *line = REQ_OFFSET_TO_STR(req, line[x]);
      if (!strncasecmp(line, boundary, strlen(boundary))){
         if (found_application_sdp && found_end_of_headers) {
            req->sdp_end = x-1;
            return 1;
         }
         found_application_sdp = FALSE;
      }
      if (!strcasecmp(line, "Content-Type: application/sdp"))
         found_application_sdp = TRUE;
      
      if (ast_strlen_zero(line)) {
         if (found_application_sdp && !found_end_of_headers){
            req->sdp_start = x;
            found_end_of_headers = TRUE;
         }
      }
   }
   if (found_application_sdp && found_end_of_headers) {
      req->sdp_end = x;
      return TRUE;
   }
   return FALSE;
}

static int find_sip_method

(

const char *

msg

)

[static]

find_sip_method: Find SIP method from header

Definition at line 2716 of file chan_sip.c.

References ARRAY_LEN, ast_strlen_zero(), cfsip_methods::id, method_match(), and sip_methods.

Referenced by __sip_pretend_ack(), handle_request_do(), and handle_response().

{
   int i, res = 0;
   
   if (ast_strlen_zero(msg))
      return 0;
   for (i = 1; i < ARRAY_LEN(sip_methods) && !res; i++) {
      if (method_match(i, msg))
         res = sip_methods[i].id;
   }
   return res;
}

static struct cfsubscription_types * find_subscription_type

(

enum subscriptiontype

subtype

)

[static]

Find subscription type in array.

Definition at line 14386 of file chan_sip.c.

References ARRAY_LEN, and subscription_types.

Referenced by transmit_state_notify().

{
   int i;

   for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
      if (subscription_types[i].type == subtype) {
         return &subscription_types[i];
      }
   }
   return &subscription_types[0];
}

static void free_old_route

(

struct sip_route *

route

)

[static]

Remove route from route list.

Definition at line 10757 of file chan_sip.c.

References ast_free, sip_route::next, and sip_request::next.

Referenced by __sip_destroy(), and build_route().

{
   struct sip_route *next;

   while (route) {
      next = route->next;
      ast_free(route);
      route = next;
   }
}

static int func_check_sipdomain	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)			[static]

Dial plan function to check if domain is local.

Definition at line 15504 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_strlen_zero(), check_sip_domain(), and LOG_WARNING.

{
   if (ast_strlen_zero(data)) {
      ast_log(LOG_WARNING, "CHECKSIPDOMAIN requires an argument - A domain name\n");
      return -1;
   }
   if (check_sip_domain(data, NULL, 0))
      ast_copy_string(buf, data, len);
   else
      buf[0] = '\0';
   return 0;
}

static int func_header_read	(	struct ast_channel *	chan,
		const char *	function,
		char *	data,
		char *	buf,
		size_t	len
	)			[static]

Read SIP header (dialplan function).

Definition at line 15440 of file chan_sip.c.

References __get_header(), AST_APP_ARG, ast_channel_lock, ast_channel_unlock, ast_copy_string(), AST_DECLARE_APP_ARGS, ast_log(), AST_STANDARD_APP_ARGS, ast_strlen_zero(), chan, sip_request::header, sip_pvt::initreq, IS_SIP_TECH, LOG_WARNING, ast_channel::tech, and ast_channel::tech_pvt.

{
   struct sip_pvt *p;
   const char *content = NULL;
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(header);
      AST_APP_ARG(number);
   );
   int i, number, start = 0;

   if (ast_strlen_zero(data)) {
      ast_log(LOG_WARNING, "This function requires a header name.\n");
      return -1;
   }

   ast_channel_lock(chan);
   if (!IS_SIP_TECH(chan->tech)) {
      ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
      ast_channel_unlock(chan);
      return -1;
   }

   AST_STANDARD_APP_ARGS(args, data);
   if (!args.number) {
      number = 1;
   } else {
      sscanf(args.number, "%d", &number);
      if (number < 1)
         number = 1;
   }

   p = chan->tech_pvt;

   /* If there is no private structure, this channel is no longer alive */
   if (!p) {
      ast_channel_unlock(chan);
      return -1;
   }

   for (i = 0; i < number; i++)
      content = __get_header(&p->initreq, args.header, &start);

   if (ast_strlen_zero(content)) {
      ast_channel_unlock(chan);
      return -1;
   }

   ast_copy_string(buf, content, len);
   ast_channel_unlock(chan);

   return 0;
}

static int function_sipchaninfo_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)			[static]

${SIPCHANINFO()} Dialplan function - reads sip channel data

Definition at line 15648 of file chan_sip.c.

References ast_channel_lock, ast_channel_unlock, ast_copy_string(), ast_inet_ntoa(), ast_log(), chan, sip_pvt::from, IS_SIP_TECH, LOG_WARNING, sip_pvt::peername, sip_pvt::recv, sip_pvt::sa, t38properties::state, sip_pvt::t38, T38_DISABLED, ast_channel::tech, ast_channel::tech_pvt, sip_pvt::uri, and sip_pvt::useragent.

{
   struct sip_pvt *p;
   static int deprecated = 0;

   *buf = 0;
   
   if (!data) {
      ast_log(LOG_WARNING, "This function requires a parameter name.\n");
      return -1;
   }

   ast_channel_lock(chan);
   if (!IS_SIP_TECH(chan->tech)) {
      ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
      ast_channel_unlock(chan);
      return -1;
   }

   if (deprecated++ % 20 == 0) {
      /* Deprecated in 1.6.1 */
      ast_log(LOG_WARNING, "SIPCHANINFO() is deprecated.  Please transition to using CHANNEL().\n");
   }

   p = chan->tech_pvt;

   /* If there is no private structure, this channel is no longer alive */
   if (!p) {
      ast_channel_unlock(chan);
      return -1;
   }

   if (!strcasecmp(data, "peerip")) {
      ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", len);
   } else  if (!strcasecmp(data, "recvip")) {
      ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", len);
   } else  if (!strcasecmp(data, "from")) {
      ast_copy_string(buf, p->from, len);
   } else  if (!strcasecmp(data, "uri")) {
      ast_copy_string(buf, p->uri, len);
   } else  if (!strcasecmp(data, "useragent")) {
      ast_copy_string(buf, p->useragent, len);
   } else  if (!strcasecmp(data, "peername")) {
      ast_copy_string(buf, p->peername, len);
   } else if (!strcasecmp(data, "t38passthrough")) {
      if (p->t38.state == T38_DISABLED)
         ast_copy_string(buf, "0", sizeof("0"));
      else    /* T38 is offered or enabled in this call */
         ast_copy_string(buf, "1", sizeof("1"));
   } else {
      ast_channel_unlock(chan);
      return -1;
   }
   ast_channel_unlock(chan);

   return 0;
}

static int function_sippeer	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)			[static]

${SIPPEER()} Dialplan function - reads peer data

Todo:

Will be deprecated after 1.4

Definition at line 15529 of file chan_sip.c.

References sip_peer::accountcode, sip_peer::addr, ast_codec_pref_index(), ast_copy_string(), ast_getformatname(), ast_getformatname_multiple(), ast_inet_ntoa(), ast_log(), ast_print_group(), ast_str_alloca, sip_peer::busy_level, sip_peer::call_limit, sip_peer::callgroup, sip_peer::capability, chanvar, sip_peer::chanvars, sip_peer::cid_name, sip_peer::cid_num, sip_peer::context, sip_peer::expire, FALSE, find_peer(), FINDPEERS, sip_peer::host_dynamic, sip_peer::inUse, sip_peer::language, LOG_WARNING, ast_variable::name, ast_variable::next, peer_mailboxes_to_str(), peer_status(), sip_peer::pickupgroup, sip_peer::prefs, sip_peer::regexten, ast_str::str, strsep(), TRUE, unref_peer(), sip_peer::useragent, and ast_variable::value.

{
   struct sip_peer *peer;
   char *colname;

   if ((colname = strchr(data, ':'))) {   /*! \todo Will be deprecated after 1.4 */
      static int deprecation_warning = 0;
      *colname++ = '\0';
      if (deprecation_warning++ % 10 == 0)
         ast_log(LOG_WARNING, "SIPPEER(): usage of ':' to separate arguments is deprecated.  Please use ',' instead.\n");
   } else if ((colname = strchr(data, ',')))
      *colname++ = '\0';
   else
      colname = "ip";

   if (!(peer = find_peer(data, NULL, TRUE, FINDPEERS, FALSE)))
      return -1;

   if (!strcasecmp(colname, "ip")) {
      ast_copy_string(buf, peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "", len);
   } else  if (!strcasecmp(colname, "port")) {
      snprintf(buf, len, "%d", ntohs(peer->addr.sin_port));
   } else  if (!strcasecmp(colname, "status")) {
      peer_status(peer, buf, len);
   } else  if (!strcasecmp(colname, "language")) {
      ast_copy_string(buf, peer->language, len);
   } else  if (!strcasecmp(colname, "regexten")) {
      ast_copy_string(buf, peer->regexten, len);
   } else  if (!strcasecmp(colname, "limit")) {
      snprintf(buf, len, "%d", peer->call_limit);
   } else  if (!strcasecmp(colname, "busylevel")) {
      snprintf(buf, len, "%d", peer->busy_level);
   } else  if (!strcasecmp(colname, "curcalls")) {
      snprintf(buf, len, "%d", peer->inUse);
   } else  if (!strcasecmp(colname, "accountcode")) {
      ast_copy_string(buf, peer->accountcode, len);
   } else  if (!strcasecmp(colname, "callgroup")) {
      ast_print_group(buf, len, peer->callgroup);
   } else  if (!strcasecmp(colname, "pickupgroup")) {
      ast_print_group(buf, len, peer->pickupgroup);
   } else  if (!strcasecmp(colname, "useragent")) {
      ast_copy_string(buf, peer->useragent, len);
   } else  if (!strcasecmp(colname, "mailbox")) {
      struct ast_str *mailbox_str = ast_str_alloca(512);
      peer_mailboxes_to_str(&mailbox_str, peer);
      ast_copy_string(buf, mailbox_str->str, len);
   } else  if (!strcasecmp(colname, "context")) {
      ast_copy_string(buf, peer->context, len);
   } else  if (!strcasecmp(colname, "expire")) {
      snprintf(buf, len, "%d", peer->expire);
   } else  if (!strcasecmp(colname, "dynamic")) {
      ast_copy_string(buf, peer->host_dynamic ? "yes" : "no", len);
   } else  if (!strcasecmp(colname, "callerid_name")) {
      ast_copy_string(buf, peer->cid_name, len);
   } else  if (!strcasecmp(colname, "callerid_num")) {
      ast_copy_string(buf, peer->cid_num, len);
   } else  if (!strcasecmp(colname, "codecs")) {
      ast_getformatname_multiple(buf, len -1, peer->capability);
   } else  if (!strncasecmp(colname, "chanvar[", 8)) {
      char *chanvar=colname + 8;
      struct ast_variable *v;
   
      chanvar = strsep(&chanvar, "]");
      for (v = peer->chanvars ; v ; v = v->next)
         if (!strcasecmp(v->name, chanvar))
            ast_copy_string(buf, v->value, len);
   } else  if (!strncasecmp(colname, "codec[", 6)) {
      char *codecnum;
      int codec = 0;
      
      codecnum = colname + 6; /* move past the '[' */
      codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */
      if((codec = ast_codec_pref_index(&peer->prefs, atoi(codecnum)))) {
         ast_copy_string(buf, ast_getformatname(codec), len);
      } else {
         buf[0] = '\0';
      }
   } else {
      buf[0] = '\0';
   }

   unref_peer(peer, "unref_peer from function_sippeer, just before return");

   return 0;
}

static char * generate_random_string	(	char *	buf,
		size_t	size
	)			[static]

Generate 32 byte random string for callid's etc.

Definition at line 6150 of file chan_sip.c.

References ast_random().

Referenced by build_callid_pvt(), and build_callid_registry().

{
   long val[4];
   int x;

   for (x=0; x<4; x++)
      val[x] = ast_random();
   snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]);

   return buf;
}

static int get_also_info	(	struct sip_pvt *	p,
		struct sip_request *	oreq
	)			[static]

Call transfer support (old way, deprecated by the IETF)--.

Definition at line 11946 of file chan_sip.c.

References ast_canmatch_extension(), ast_copy_string(), ast_debug, ast_exists_extension(), ast_log(), ast_string_field_set, ast_strlen_zero(), ast_uri_decode(), ast_verbose, context, sip_pvt::context, dialog_unref(), get_header(), get_in_brackets(), sip_pvt::initreq, LOG_WARNING, ast_channel::macrocontext, sip_pvt::owner, pbx_builtin_getvar_helper(), sip_pvt::refer, sip_refer::refer_call, sip_refer::refer_contact, sip_refer::refer_to, sip_refer::refer_to_domain, sip_refer::referred_by, S_OR, sip_debug_test_pvt(), and sip_refer_allocate().

Referenced by handle_request_bye().

{
   char tmp[256] = "", *c, *a;
   struct sip_request *req = oreq ? oreq : &p->initreq;
   struct sip_refer *referdata = NULL;
   const char *transfer_context = NULL;
   
   if (!p->refer && !sip_refer_allocate(p))
      return -1;

   referdata = p->refer;

   ast_copy_string(tmp, get_header(req, "Also"), sizeof(tmp));
   c = get_in_brackets(tmp);

   if (pedanticsipchecking)
      ast_uri_decode(c);

   if (!strncasecmp(c, "sip:", 4)) {
      c += 4;
   } else if (!strncasecmp(c, "sips:", 5)) {
      c += 5;
   } else {
      ast_log(LOG_WARNING, "Huh?  Not a SIP header in Also: transfer (%s)?\n", c);
      return -1;
   }

   if ((a = strchr(c, ';')))  /* Remove arguments */
      *a = '\0';
   
   if ((a = strchr(c, '@'))) {   /* Separate Domain */
      *a++ = '\0';
      ast_copy_string(referdata->refer_to_domain, a, sizeof(referdata->refer_to_domain));
   }
   
   if (sip_debug_test_pvt(p))
      ast_verbose("Looking for %s in %s\n", c, p->context);

   if (p->owner)  /* Mimic behaviour in res_features.c */
      transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT");

   /* By default, use the context in the channel sending the REFER */
   if (ast_strlen_zero(transfer_context)) {
      transfer_context = S_OR(p->owner->macrocontext,
               S_OR(p->context, default_context));
   }
   if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) {
      /* This is a blind transfer */
      ast_debug(1, "SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context);
      ast_copy_string(referdata->refer_to, c, sizeof(referdata->refer_to));
      ast_copy_string(referdata->referred_by, "", sizeof(referdata->referred_by));
      ast_copy_string(referdata->refer_contact, "", sizeof(referdata->refer_contact));
      referdata->refer_call = dialog_unref(referdata->refer_call, "unreffing referdata->refer_call");
      /* Set new context */
      ast_string_field_set(p, context, transfer_context);
      return 0;
   } else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) {
      return 1;
   }

   return -1;
}

static char* get_body	(	struct sip_request *	req,
		char *	name
	)			[static]

Get a specific line from the message body.

Definition at line 5933 of file chan_sip.c.

References get_body_by_line(), sip_request::len, sip_request::lines, and REQ_OFFSET_TO_STR.

Referenced by handle_request_info().

{
   int x;
   int len = strlen(name);
   char *r;

   for (x = 0; x < req->lines; x++) {
      r = get_body_by_line(REQ_OFFSET_TO_STR(req, line[x]), name, len);
      if (r[0] != '\0')
         return r;
   }

   return "";
}

static char* get_body_by_line	(	const char *	line,
		const char *	name,
		int	nameLen
	)			[static]

Reads one line of SIP message body.

Definition at line 5899 of file chan_sip.c.

References ast_skip_blanks().

Referenced by get_body(), and get_sdp_iterate().

{
   if (!strncasecmp(line, name, nameLen) && line[nameLen] == '=')
      return ast_skip_blanks(line + nameLen + 1);

   return "";
}

static int get_cached_mwi	(	struct sip_peer *	peer,
		int *	new,
		int *	old
	)			[static]

Get cached MWI info.

Return values:

	0	At least one message is waiting
	1	no messages waiting

Definition at line 20190 of file chan_sip.c.

References ast_event_destroy(), ast_event_get_cached(), ast_event_get_ie_uint(), AST_EVENT_IE_CONTEXT, AST_EVENT_IE_END, AST_EVENT_IE_MAILBOX, AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_OLDMSGS, AST_EVENT_IE_PLTYPE_STR, AST_EVENT_MWI, AST_LIST_TRAVERSE, sip_mailbox::entry, mailbox, sip_peer::mailboxes, and S_OR.

Referenced by sip_send_mwi_to_peer().

{
   struct sip_mailbox *mailbox;

   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      struct ast_event *event;
      event = ast_event_get_cached(AST_EVENT_MWI,
         AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox->mailbox,
         AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, S_OR(mailbox->context, "default"),
         AST_EVENT_IE_END);
      if (!event)
         continue;
      *new += ast_event_get_ie_uint(event, AST_EVENT_IE_NEWMSGS);
      *old += ast_event_get_ie_uint(event, AST_EVENT_IE_OLDMSGS);
      ast_event_destroy(event);
   }

   return (*new || *old) ? 0 : 1;
}

static char * get_calleridname	(	const char *	input,
		char *	output,
		size_t	outputsize
	)			[static]

Get caller id name from SIP headers.

Definition at line 12101 of file chan_sip.c.

References ast_copy_string(), and ast_skip_blanks().

Referenced by check_user_full().

{
   const char *end = strchr(input, '<');  /* first_bracket */
   const char *tmp = strchr(input, '"');  /* first quote */
   int bytes = 0;
   int maxbytes = outputsize - 1;

   if (!end || end == input)  /* we require a part in brackets */
      return NULL;

   end--; /* move just before "<" */

   if (tmp && tmp <= end) {
      /* The quote (tmp) precedes the bracket (end+1).
       * Find the matching quote and return the content.
       */
      end = strchr(tmp+1, '"');
      if (!end)
         return NULL;
      bytes = (int) (end - tmp);
      /* protect the output buffer */
      if (bytes > maxbytes)
         bytes = maxbytes;
      ast_copy_string(output, tmp + 1, bytes);
   } else {
      /* No quoted string, or it is inside brackets. */
      /* clear the empty characters in the begining*/
      input = ast_skip_blanks(input);
      /* clear the empty characters in the end */
      while(*end && *end < 33 && end > input)
         end--;
      if (end >= input) {
         bytes = (int) (end - input) + 2;
         /* protect the output buffer */
         if (bytes > maxbytes)
            bytes = maxbytes;
         ast_copy_string(output, input, bytes);
      } else
         return NULL;
   }
   return output;
}

static int get_destination	(	struct sip_pvt *	p,
		struct sip_request *	oreq
	)			[static]

Find out who the call is for. We use the request uri as a destination. This code assumes authentication has been done, so that the device (peer/user) context is already set.

Returns:

0 on success (found a matching extension), 1 for pickup extension or overlap dialling support (if we support it), -1 on error.

Definition at line 11556 of file chan_sip.c.

References ast_canmatch_extension(), ast_copy_string(), ast_debug, ast_exists_extension(), ast_get_hint(), AST_LIST_EMPTY, ast_log(), AST_MAX_EXTENSION, ast_pickup_ext(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_decode(), ast_verbose, check_sip_domain(), sip_pvt::cid_num, sip_pvt::context, context, sip_pvt::domain, exten, sip_pvt::exten, get_header(), get_in_brackets(), global_flags, sip_pvt::initreq, LOG_WARNING, sip_request::method, REQ_OFFSET_TO_STR, sip_request::rlPart2, S_OR, sip_debug_test_pvt(), SIP_INVITE, sip_methods, SIP_PAGE2_ALLOWOVERLAP, SIP_REFER, SIP_SUBSCRIBE, strsep(), and sip_pvt::subscribecontext.

Referenced by handle_request_invite(), handle_request_options(), and handle_request_subscribe().

{
   char tmp[256] = "", *uri, *a;
   char tmpf[256] = "", *from = NULL;
   struct sip_request *req;
   char *colon;
   char *decoded_uri;
   
   req = oreq;
   if (!req)
      req = &p->initreq;

   /* Find the request URI */
   if (req->rlPart2)
      ast_copy_string(tmp, REQ_OFFSET_TO_STR(req, rlPart2), sizeof(tmp));
   
   if (pedanticsipchecking)
      ast_uri_decode(tmp);

   uri = get_in_brackets(tmp);
   
   if (!strncasecmp(uri, "sip:", 4)) {
      uri += 4;
   } else if (!strncasecmp(uri, "sips:", 5)) {
      uri += 5;
   } else {
      ast_log(LOG_WARNING, "Huh?  Not a SIP header (%s)?\n", uri);
      return -1;
   }

   /* Now find the From: caller ID and name */
   /* XXX Why is this done in get_destination? Isn't it already done?
      Needs to be checked 
        */
   ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf));
   if (!ast_strlen_zero(tmpf)) {
      if (pedanticsipchecking)
         ast_uri_decode(tmpf);
      from = get_in_brackets(tmpf);
   } 
   
   if (!ast_strlen_zero(from)) {
      if (!strncasecmp(from, "sip:", 4)) {
         from += 4;
      } else if (!strncasecmp(from, "sips:", 5)) {
         from += 5;
      } else {
         ast_log(LOG_WARNING, "Huh?  Not a SIP header (%s)?\n", from);
         return -1;
      }
      if ((a = strchr(from, '@')))
         *a++ = '\0';
      else
         a = from;   /* just a domain */
      from = strsep(&from, ";"); /* Remove userinfo options */
      a = strsep(&a, ";");    /* Remove URI options */
      ast_string_field_set(p, fromdomain, a);
   }

   /* Skip any options and find the domain */

   /* Get the target domain */
   if ((a = strchr(uri, '@'))) {
      *a++ = '\0';
   } else { /* No username part */
      a = uri;
      uri = "s";  /* Set extension to "s" */
   }
   colon = strchr(a, ':'); /* Remove :port */
   if (colon)
      *colon = '\0';

   uri = strsep(&uri, ";");   /* Remove userinfo options */
   a = strsep(&a, ";");    /* Remove URI options */

   ast_string_field_set(p, domain, a);

   if (!AST_LIST_EMPTY(&domain_list)) {
      char domain_context[AST_MAX_EXTENSION];

      domain_context[0] = '\0';
      if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) {
         if (!allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
            ast_debug(1, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
            return -2;
         }
      }
      /* If we have a context defined, overwrite the original context */
      if (!ast_strlen_zero(domain_context))
         ast_string_field_set(p, context, domain_context);
   }

   /* If the request coming in is a subscription and subscribecontext has been specified use it */
   if (req->method == SIP_SUBSCRIBE && !ast_strlen_zero(p->subscribecontext))
      ast_string_field_set(p, context, p->subscribecontext);

   if (sip_debug_test_pvt(p))
      ast_verbose("Looking for %s in %s (domain %s)\n", uri, p->context, p->domain);

   /* If this is a subscription we actually just need to see if a hint exists for the extension */
   if (req->method == SIP_SUBSCRIBE) {
      char hint[AST_MAX_EXTENSION];
      return (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, p->exten) ? 0 : -1);
   } else {
      decoded_uri = ast_strdupa(uri);
      ast_uri_decode(decoded_uri);
      /* Check the dialplan for the username part of the request URI,
         the domain will be stored in the SIPDOMAIN variable
         Since extensions.conf can have unescaped characters, try matching a decoded
         uri in addition to the non-decoded uri
         Return 0 if we have a matching extension */
      if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from)) || ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from)) ||
          !strcmp(decoded_uri, ast_pickup_ext())) {
         if (!oreq)
            ast_string_field_set(p, exten, decoded_uri);
         return 0;
      } 
   }

   /* Return 1 for pickup extension or overlap dialling support (if we support it) */
   if((ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) && 
       ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))) ||
       !strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri))) {
      return 1;
   }
   
   return -1;
}

static const char * get_header	(	const struct sip_request *	req,
		const char *	name
	)			[static]

Get header from SIP request.

Returns:

Always return something, so don't check for NULL because it won't happen :-)

Definition at line 6026 of file chan_sip.c.

References __get_header().

{
   int start = 0;
   return __get_header(req, name, &start);
}

static char * get_in_brackets

(

char *

tmp

)

[static]

Pick out text in brackets from character string.

Returns:

pointer to terminated stripped string

Parameters:

tmp

input string that will be modified Examples: "foo" <bar> valid input, returns bar foo returns the whole string < "foo ... > returns the string between brackets < "foo... bogus (missing closing bracket), returns the whole string XXX maybe should still skip the opening bracket

Definition at line 3568 of file chan_sip.c.

References ast_log(), find_closing_quote(), LOG_WARNING, and parse().

Referenced by check_user_full(), extract_uri(), get_also_info(), get_destination(), get_rdnis(), get_refer_info(), parse_moved_contact(), parse_ok_contact(), parse_register_contact(), register_verify(), reqprep(), transmit_refer(), and transmit_state_notify().

{
   const char *parse = tmp;
   char *first_bracket;

   /*
    * Skip any quoted text until we find the part in brackets.
   * On any error give up and return the full string.
   */
   while ( (first_bracket = strchr(parse, '<')) ) {
      char *first_quote = strchr(parse, '"');

      if (!first_quote || first_quote > first_bracket)
         break; /* no need to look at quoted part */
      /* the bracket is within quotes, so ignore it */
      parse = find_closing_quote(first_quote + 1, NULL);
      if (!*parse) { /* not found, return full string ? */
         /* XXX or be robust and return in-bracket part ? */
         ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
         break;
      }
      parse++;
   }
   if (first_bracket) {
      char *second_bracket = strchr(first_bracket + 1, '>');
      if (second_bracket) {
         *second_bracket = '\0';
         tmp = first_bracket + 1;
      } else {
         ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
      }
   }
   
   return tmp;
}

static struct ast_variable * get_insecure_variable_from_config

(

struct ast_config *

config

)

[static]

Definition at line 3912 of file chan_sip.c.

References ast_category_browse(), ast_category_root(), ast_test_flag, ast_variable_retrieve(), ast_flags::flags, set_insecure_flags(), SIP_INSECURE_PORT, and var.

{
   struct ast_variable *var = NULL;
   struct ast_flags flags = {0};
   char *cat = NULL;
   const char *insecure;
   while ((cat = ast_category_browse(cfg, cat))) {
      insecure = ast_variable_retrieve(cfg, cat, "insecure");
      set_insecure_flags(&flags, insecure, -1);
      if (ast_test_flag(&flags, SIP_INSECURE_PORT)) {
         var = ast_category_root(cfg, cat);
         break;
      }
   }
   return var;
}

static int get_msg_text	(	char *	buf,
		int	len,
		struct sip_request *	req,
		int	addnewline
	)			[static]

Get text out of a SIP MESSAGE packet.

Definition at line 12490 of file chan_sip.c.

References sip_request::lines, and REQ_OFFSET_TO_STR.

Referenced by handle_request_info(), handle_request_notify(), and receive_message().

{
   int x;
   int y;

   buf[0] = '\0';
   /*XXX isn't strlen(buf) going to always be 0? */
   y = len - strlen(buf) - 5;
   if (y < 0)
      y = 0;
   for (x = 0; x < req->lines; x++) {
      char *line = REQ_OFFSET_TO_STR(req, line[x]);
      strncat(buf, line, y); /* safe */
      y -= strlen(line) + 1;
      if (y < 0)
         y = 0;
      if (y != 0 && addnewline)
         strcat(buf, "\n"); /* safe */
   }
   return 0;
}

static const char * get_name_from_variable	(	struct ast_variable *	var,
		const char *	newpeername
	)			[static]

Definition at line 3929 of file chan_sip.c.

References ast_variable::name, ast_variable::next, ast_variable::value, and var.

{
   struct ast_variable *tmp;
   for (tmp = var; tmp; tmp = tmp->next) {
      if (!newpeername && !strcasecmp(tmp->name, "name"))
         newpeername = tmp->value;
   }
   return newpeername;
}

static void get_our_media_address	(	struct sip_pvt *	p,
		int	needvideo,
		struct sockaddr_in *	sin,
		struct sockaddr_in *	vsin,
		struct sockaddr_in *	tsin,
		struct sockaddr_in *	dest,
		struct sockaddr_in *	vdest
	)			[static]

Set all IP media addresses for this call.

Note:

called from add_sdp()

Definition at line 8558 of file chan_sip.c.

References ast_rtp_get_us(), sip_pvt::ourip, sip_pvt::redirip, sip_pvt::rtp, sip_pvt::trtp, sip_pvt::vredirip, and sip_pvt::vrtp.

Referenced by add_sdp().

{
   /* First, get our address */
   ast_rtp_get_us(p->rtp, sin);
   if (p->vrtp)
      ast_rtp_get_us(p->vrtp, vsin);
   if (p->trtp)
      ast_rtp_get_us(p->trtp, tsin);

   /* Now, try to figure out where we want them to send data */
   /* Is this a re-invite to move the media out, then use the original offer from caller  */
   if (p->redirip.sin_addr.s_addr) {   /* If we have a redirection IP, use it */
      dest->sin_port = p->redirip.sin_port;
      dest->sin_addr = p->redirip.sin_addr;
   } else {
      dest->sin_addr = p->ourip.sin_addr;
      dest->sin_port = sin->sin_port;
   }
   if (needvideo) {
      /* Determine video destination */
      if (p->vredirip.sin_addr.s_addr) {
         vdest->sin_addr = p->vredirip.sin_addr;
         vdest->sin_port = p->vredirip.sin_port;
      } else {
         vdest->sin_addr = p->ourip.sin_addr;
         vdest->sin_port = vsin->sin_port;
      }
   }

}

static int get_rdnis	(	struct sip_pvt *	p,
		struct sip_request *	oreq
	)			[static]

Get referring dnis.

Definition at line 11489 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_string_field_set, ast_strip_quoted(), ast_strlen_zero(), ast_verbose, exten, get_header(), get_in_brackets(), sip_pvt::initreq, LOG_WARNING, sip_pvt::owner, pbx_builtin_setvar_helper(), sip_debug_test_pvt(), sip_set_redirstr(), strcasestr(), and strsep().

Referenced by handle_request_invite().

{
   char tmp[256], *exten, *rexten, *rdomain;
   char *params, *reason = NULL;
   struct sip_request *req;
   
   req = oreq ? oreq : &p->initreq;

   ast_copy_string(tmp, get_header(req, "Diversion"), sizeof(tmp));
   if (ast_strlen_zero(tmp))
      return 0;

   params = strchr(tmp, ';');

   exten = get_in_brackets(tmp);
   if (!strncasecmp(exten, "sip:", 4)) {
      exten += 4;
   } else if (!strncasecmp(exten, "sips:", 5)) {
      exten += 5;
   } else {
      ast_log(LOG_WARNING, "Huh?  Not an RDNIS SIP header (%s)?\n", exten);
      return -1;
   }

   /* Get diversion-reason param if present */
   if (params) {
      *params = '\0';   /* Cut off parameters  */
      params++;
      while (*params == ';' || *params == ' ')
         params++;
      /* Check if we have a reason parameter */
      if ((reason = strcasestr(params, "reason="))) {
         reason+=7;
         /* Remove enclosing double-quotes */
         if (*reason == '"') 
            ast_strip_quoted(reason, "\"", "\"");
         if (!ast_strlen_zero(reason)) {
            sip_set_redirstr(p, reason);
            if (p->owner) {
               pbx_builtin_setvar_helper(p->owner, "__PRIREDIRECTREASON", p->redircause);
               pbx_builtin_setvar_helper(p->owner, "__SIPREDIRECTREASON", reason);
            }
         }
      }
   }

   rdomain = exten;
   rexten = strsep(&rdomain, "@");  /* trim anything after @ */
   if (p->owner) 
      pbx_builtin_setvar_helper(p->owner, "__SIPRDNISDOMAIN", rdomain);

   if (sip_debug_test_pvt(p))
      ast_verbose("RDNIS for this call is is %s (reason %s)\n", exten, reason ? reason : "");

   ast_string_field_set(p, rdnis, rexten);

   return 0;
}

static int get_refer_info	(	struct sip_pvt *	transferer,
		struct sip_request *	outgoing_req
	)			[static]

Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.

Definition at line 11770 of file chan_sip.c.

References ast_bridged_channel(), ast_copy_string(), ast_debug, ast_exists_extension(), ast_log(), ast_strdupa, ast_strlen_zero(), ast_uri_decode(), ast_verbose, sip_refer::attendedtransfer, sip_pvt::context, get_header(), get_in_brackets(), sip_pvt::initreq, LOG_WARNING, ast_channel::macrocontext, sip_pvt::owner, pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), sip_pvt::refer, sip_refer::refer_to, sip_refer::refer_to_context, sip_refer::refer_to_domain, sip_refer::refer_to_urioption, sip_refer::referred_by, sip_refer::referred_by_name, sip_refer::replaces_callid, sip_refer::replaces_callid_fromtag, sip_refer::replaces_callid_totag, S_OR, sip_debug_test_pvt(), and strcasestr().

Referenced by handle_request_refer().

{

   const char *p_referred_by = NULL;
   char *h_refer_to = NULL; 
   char *h_referred_by = NULL;
   char *refer_to;
   const char *p_refer_to;
   char *referred_by_uri = NULL;
   char *ptr;
   struct sip_request *req = NULL;
   const char *transfer_context = NULL;
   struct sip_refer *referdata;


   req = outgoing_req;
   referdata = transferer->refer;

   if (!req)
      req = &transferer->initreq;

   p_refer_to = get_header(req, "Refer-To");
   if (ast_strlen_zero(p_refer_to)) {
      ast_log(LOG_WARNING, "Refer-To Header missing. Skipping transfer.\n");
      return -2;  /* Syntax error */
   }
   h_refer_to = ast_strdupa(p_refer_to);
   refer_to = get_in_brackets(h_refer_to);
   if (pedanticsipchecking)
      ast_uri_decode(refer_to);

   if (!strncasecmp(refer_to, "sip:", 4)) {
      refer_to += 4;       /* Skip sip: */
   } else if (!strncasecmp(refer_to, "sips:", 5)) {
      refer_to += 5;
   } else {
      ast_log(LOG_WARNING, "Can't transfer to non-sip: URI.  (Refer-to: %s)?\n", refer_to);
      return -3;
   }

   /* Get referred by header if it exists */
   p_referred_by = get_header(req, "Referred-By");

   /* Give useful transfer information to the dialplan */
   if (transferer->owner) {
      struct ast_channel *peer = ast_bridged_channel(transferer->owner);
      if (peer) {
         pbx_builtin_setvar_helper(peer, "SIPREFERRINGCONTEXT", transferer->context);
         pbx_builtin_setvar_helper(peer, "SIPREFERREDBYHDR", p_referred_by);
      }
   }

   if (!ast_strlen_zero(p_referred_by)) {
      char *lessthan;
      h_referred_by = ast_strdupa(p_referred_by);
      if (pedanticsipchecking)
         ast_uri_decode(h_referred_by);

      /* Store referrer's caller ID name */
      ast_copy_string(referdata->referred_by_name, h_referred_by, sizeof(referdata->referred_by_name));
      if ((lessthan = strchr(referdata->referred_by_name, '<'))) {
         *(lessthan - 1) = '\0'; /* Space */
      }

      referred_by_uri = get_in_brackets(h_referred_by);
      if (!strncasecmp(referred_by_uri, "sip:", 4)) {
         referred_by_uri += 4;      /* Skip sip: */
      } else if (!strncasecmp(referred_by_uri, "sips:", 5)) {
         referred_by_uri += 5;      /* Skip sips: */
      } else {
         ast_log(LOG_WARNING, "Huh?  Not a sip: header (Referred-by: %s). Skipping.\n", referred_by_uri);
         referred_by_uri = NULL;
      }
   }

   /* Check for arguments in the refer_to header */
   if ((ptr = strcasestr(refer_to, "replaces="))) {
      char *to = NULL, *from = NULL;
      
      /* This is an attended transfer */
      referdata->attendedtransfer = 1;
      ast_copy_string(referdata->replaces_callid, ptr+9, sizeof(referdata->replaces_callid));
      ast_uri_decode(referdata->replaces_callid);
      if ((ptr = strchr(referdata->replaces_callid, ';')))  /* Find options */ {
         *ptr++ = '\0';
      }
      
      if (ptr) {
         /* Find the different tags before we destroy the string */
         to = strcasestr(ptr, "to-tag=");
         from = strcasestr(ptr, "from-tag=");
      }
      
      /* Grab the to header */
      if (to) {
         ptr = to + 7;
         if ((to = strchr(ptr, '&')))
            *to = '\0';
         if ((to = strchr(ptr, ';')))
            *to = '\0';
         ast_copy_string(referdata->replaces_callid_totag, ptr, sizeof(referdata->replaces_callid_totag));
      }
      
      if (from) {
         ptr = from + 9;
         if ((to = strchr(ptr, '&')))
            *to = '\0';
         if ((to = strchr(ptr, ';')))
            *to = '\0';
         ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag));
      }
      
      if (!pedanticsipchecking)
         ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid );
      else
         ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" );
   }
   
   if ((ptr = strchr(refer_to, '@'))) {   /* Separate domain */
      char *urioption = NULL, *domain;
      *ptr++ = '\0';

      if ((urioption = strchr(ptr, ';'))) /* Separate urioptions */
         *urioption++ = '\0';
      
      domain = ptr;
      if ((ptr = strchr(domain, ':'))) /* Remove :port */
         *ptr = '\0';
      
      /* Save the domain for the dial plan */
      ast_copy_string(referdata->refer_to_domain, domain, sizeof(referdata->refer_to_domain));
      if (urioption)
         ast_copy_string(referdata->refer_to_urioption, urioption, sizeof(referdata->refer_to_urioption));
   }

   if ((ptr = strchr(refer_to, ';')))  /* Remove options */
      *ptr = '\0';
   ast_copy_string(referdata->refer_to, refer_to, sizeof(referdata->refer_to));
   
   if (referred_by_uri) {
      if ((ptr = strchr(referred_by_uri, ';')))    /* Remove options */
         *ptr = '\0';
      ast_copy_string(referdata->referred_by, referred_by_uri, sizeof(referdata->referred_by));
   } else {
      referdata->referred_by[0] = '\0';
   }

   /* Determine transfer context */
   if (transferer->owner)  /* Mimic behaviour in res_features.c */
      transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT");

   /* By default, use the context in the channel sending the REFER */
   if (ast_strlen_zero(transfer_context)) {
      transfer_context = S_OR(transferer->owner->macrocontext,
               S_OR(transferer->context, default_context));
   }

   ast_copy_string(referdata->refer_to_context, transfer_context, sizeof(referdata->refer_to_context));
   
   /* Either an existing extension or the parking extension */
   if (referdata->attendedtransfer || ast_exists_extension(NULL, transfer_context, refer_to, 1, NULL) ) {
      if (sip_debug_test_pvt(transferer)) {
         ast_verbose("SIP transfer to extension %s@%s by %s\n", refer_to, transfer_context, referred_by_uri);
      }
      /* We are ready to transfer to the extension */
      return 0;
   } 
   if (sip_debug_test_pvt(transferer))
      ast_verbose("Failed SIP Transfer to non-existing extension %s in context %s\n n", refer_to, transfer_context);

   /* Failure, we can't find this extension */
   return -1;
}

static int get_rpid_num	(	const char *	input,
		char *	output,
		int	maxlen
	)			[static]

Get caller id number from Remote-Party-ID header field Returns true if number should be restricted (privacy setting found) output is set to NULL if no number found.

Definition at line 12148 of file chan_sip.c.

References ast_copy_string(), and AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED.

Referenced by check_user_full().

{
   char *start;
   char *end;

   start = strchr(input, ':');
   if (!start) {
      output[0] = '\0';
      return 0;
   }
   start++;

   /* we found "number" */
   ast_copy_string(output, start, maxlen);
   output[maxlen-1] = '\0';

   end = strchr(output, '@');
   if (end)
      *end = '\0';
   else
      output[0] = '\0';
   if (strstr(input, "privacy=full") || strstr(input, "privacy=uri"))
      return AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;

   return 0;
}

static const char * get_sdp	(	struct sip_request *	req,
		const char *	name
	)			[static]

Get a line from an SDP message body.

Definition at line 5925 of file chan_sip.c.

References dummy(), and get_sdp_iterate().

{
   int dummy = 0;

   return get_sdp_iterate(&dummy, req, name);
}

static const char * get_sdp_iterate	(	int *	start,
		struct sip_request *	req,
		const char *	name
	)			[static]

Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.

Definition at line 5911 of file chan_sip.c.

References get_body_by_line(), sip_request::len, REQ_OFFSET_TO_STR, and sip_request::sdp_end.

{
   int len = strlen(name);

   while (*start < req->sdp_end) {
      const char *r = get_body_by_line(REQ_OFFSET_TO_STR(req, line[(*start)++]), name, len);
      if (r[0] != '\0')
         return r;
   }

   return "";
}

static struct sip_pvt * get_sip_pvt_byid_locked	(	const char *	callid,
		const char *	totag,
		const char *	fromtag
	)			[static]

Lock dialog lock and find matching pvt lock.

Returns:

a reference, remember to release it when done

Definition at line 11688 of file chan_sip.c.

References ao2_t_find, ast_channel_trylock, ast_debug, ast_strlen_zero(), sip_pvt::callid, dialogs, OBJ_POINTER, sip_pvt_lock, sip_pvt_unlock, and TRUE.

Referenced by handle_request_invite(), and local_attended_transfer().

{
   struct sip_pvt *sip_pvt_ptr;
   struct sip_pvt tmp_dialog = {
      .callid = callid,
   };

   if (totag)
      ast_debug(4, "Looking for callid %s (fromtag %s totag %s)\n", callid, fromtag ? fromtag : "<no fromtag>", totag ? totag : "<no totag>");

   /* Search dialogs and find the match */
   
   sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find of dialog in dialogs table");
   if (sip_pvt_ptr) {
      /* Go ahead and lock it (and its owner) before returning */
      sip_pvt_lock(sip_pvt_ptr);
      if (pedanticsipchecking) {
         unsigned char frommismatch = 0, tomismatch = 0;

         if (ast_strlen_zero(fromtag)) {
            sip_pvt_unlock(sip_pvt_ptr);
            ast_debug(4, "Matched %s call for callid=%s - no from tag specified, pedantic check fails\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
            return NULL;
         }

         if (ast_strlen_zero(totag)) {
            sip_pvt_unlock(sip_pvt_ptr);
            ast_debug(4, "Matched %s call for callid=%s - no to tag specified, pedantic check fails\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
            return NULL;
         }
         /* RFC 3891
          * > 3.  User Agent Server Behavior: Receiving a Replaces Header
          * > The Replaces header contains information used to match an existing
          * > SIP dialog (call-id, to-tag, and from-tag).  Upon receiving an INVITE
          * > with a Replaces header, the User Agent (UA) attempts to match this
          * > information with a confirmed or early dialog.  The User Agent Server
          * > (UAS) matches the to-tag and from-tag parameters as if they were tags
          * > present in an incoming request.  In other words, the to-tag parameter
          * > is compared to the local tag, and the from-tag parameter is compared
          * > to the remote tag.
          *
          * Thus, the totag is always compared to the local tag, regardless if
          * this our call is an incoming or outgoing call.
          */
         frommismatch = !!strcmp(fromtag, sip_pvt_ptr->theirtag);
         tomismatch = !!strcmp(totag, sip_pvt_ptr->tag);
         if (frommismatch || tomismatch) {
            sip_pvt_unlock(sip_pvt_ptr);
            if (frommismatch) {
               ast_debug(4, "Matched %s call for callid=%s - But the pedantic check rejected the match; their tag is %s Our tag is %s\n",
                       sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid, 
                       fromtag, sip_pvt_ptr->theirtag);
            }
            if (tomismatch) {
               ast_debug(4, "Matched %s call for callid=%s - pedantic to tag check fails; their tag is %s our tag is %s\n",
                    sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid, 
                    totag, sip_pvt_ptr->tag);
            }
            return NULL;
         }
      }
      
      if (totag)
         ast_debug(4, "Matched %s call - their tag is %s Our tag is %s\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING",
                 sip_pvt_ptr->theirtag, sip_pvt_ptr->tag);

      /* deadlock avoidance... */
      while (sip_pvt_ptr->owner && ast_channel_trylock(sip_pvt_ptr->owner)) {
         sip_pvt_unlock(sip_pvt_ptr);
         usleep(1);
         sip_pvt_lock(sip_pvt_ptr);
      }
   }
   
   return sip_pvt_ptr;
}

static const char* get_transport

(

enum sip_transport

t

)

[inline, static]

Definition at line 2838 of file chan_sip.c.

References SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, and SIP_TRANSPORT_UDP.

Referenced by _sip_show_peer(), build_contact(), build_peer(), create_addr(), get_transport_pvt(), handle_request_do(), parse_moved_contact(), sip_show_tcp(), and transmit_register().

{
   switch (t) {
   case SIP_TRANSPORT_UDP:
      return "UDP";
   case SIP_TRANSPORT_TCP:
      return "TCP";
   case SIP_TRANSPORT_TLS:
      return "TLS";
   }

   return "UNKNOWN";
}

static const char* get_transport_list

(

struct sip_peer *

peer

)

[inline, static]

Definition at line 2818 of file chan_sip.c.

References SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, SIP_TRANSPORT_UDP, and sip_peer::transports.

                                                                    {
   switch (peer->transports) {
      case SIP_TRANSPORT_UDP:
         return "UDP";
      case SIP_TRANSPORT_TCP:
         return "TCP";
      case SIP_TRANSPORT_TLS:
         return "TLS";
      case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TCP:
         return "TCP,UDP";
      case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TLS:
         return "TLS,UDP";
      case SIP_TRANSPORT_TCP | SIP_TRANSPORT_TLS:
         return "TLS,TCP";
      default:
         return peer->transports ? 
            "TLS,TCP,UDP" : "UNKNOWN"; 
   }
}

static const char* get_transport_pvt

(

struct sip_pvt *

p

)

[inline, static]

Definition at line 2852 of file chan_sip.c.

References get_transport(), sip_pvt::outboundproxy, sip_pvt::socket, sip_proxy::transport, and sip_socket::type.

Referenced by __sip_xmit(), and build_via().

{
   if (p->outboundproxy && p->outboundproxy->transport)
      p->socket.type = p->outboundproxy->transport;

   return get_transport(p->socket.type);
}

static const char * gettag	(	const struct sip_request *	req,
		const char *	header,
		char *	tagbuf,
		int	tagbufsize
	)			[static]

Get tag from packet.

Returns:

Returns the pointer to the provided tag buffer, or NULL if the tag was not found.

Definition at line 17160 of file chan_sip.c.

References ast_copy_string(), get_header(), strcasestr(), and strsep().

Referenced by find_call(), handle_incoming(), handle_request_subscribe(), and handle_response().

{
   const char *thetag;

   if (!tagbuf)
      return NULL;
   tagbuf[0] = '\0';    /* reset the buffer */
   thetag = get_header(req, header);
   thetag = strcasestr(thetag, ";tag=");
   if (thetag) {
      thetag += 5;
      ast_copy_string(tagbuf, thetag, tagbufsize);
      return strsep(&tagbuf, ";");
   }
   return NULL;
}

static int handle_common_options	(	struct ast_flags *	flags,
		struct ast_flags *	mask,
		struct ast_variable *	v
	)			[static]

Handle flag-type options common to configuration of devices - peers.

Parameters:

	flags	array of two struct ast_flags
	mask	array of two struct ast_flags
	v	linked list of config variables to process

Returns:

non-zero if any config options were handled, zero otherwise

Definition at line 21181 of file chan_sip.c.

References ast_clear_flag, ast_copy_string(), ast_false(), ast_log(), ast_set2_flag, ast_set_flag, ast_true(), buf, ast_channel::flags, ast_variable::lineno, LOG_WARNING, ast_variable::name, ast_channel::next, set_insecure_flags(), SIP_CAN_REINVITE, SIP_CAN_REINVITE_NAT, SIP_DTMF, SIP_DTMF_AUTO, SIP_DTMF_INBAND, SIP_DTMF_INFO, SIP_DTMF_RFC2833, SIP_DTMF_SHORTINFO, SIP_G726_NONSTANDARD, SIP_INSECURE, SIP_NAT, SIP_NAT_ALWAYS, SIP_NAT_NEVER, SIP_NAT_RFC3581, SIP_NAT_ROUTE, SIP_PAGE2_ALLOWOVERLAP, SIP_PAGE2_ALLOWSUBSCRIBE, SIP_PAGE2_BUGGY_MWI, SIP_PAGE2_RFC2833_COMPENSATE, SIP_PAGE2_T38SUPPORT_RTP, SIP_PAGE2_T38SUPPORT_TCP, SIP_PAGE2_T38SUPPORT_UDPTL, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_UDPTL_DESTINATION, SIP_PAGE2_VIDEOSUPPORT, SIP_PAGE2_VIDEOSUPPORT_ALWAYS, SIP_PROG_INBAND, SIP_PROG_INBAND_NO, SIP_PROG_INBAND_YES, SIP_PROMISCREDIR, SIP_REINVITE, SIP_REINVITE_UPDATE, SIP_SENDRPID, SIP_TRUSTRPID, SIP_USECLIENTCODE, strsep(), and ast_variable::value.

Referenced by build_peer().

{
   int res = 1;

   if (!strcasecmp(v->name, "trustrpid")) {
      ast_set_flag(&mask[0], SIP_TRUSTRPID);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_TRUSTRPID);
   } else if (!strcasecmp(v->name, "sendrpid")) {
      ast_set_flag(&mask[0], SIP_SENDRPID);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_SENDRPID);
   } else if (!strcasecmp(v->name, "g726nonstandard")) {
      ast_set_flag(&mask[0], SIP_G726_NONSTANDARD);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_G726_NONSTANDARD);
   } else if (!strcasecmp(v->name, "useclientcode")) {
      ast_set_flag(&mask[0], SIP_USECLIENTCODE);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_USECLIENTCODE);
   } else if (!strcasecmp(v->name, "dtmfmode")) {
      ast_set_flag(&mask[0], SIP_DTMF);
      ast_clear_flag(&flags[0], SIP_DTMF);
      if (!strcasecmp(v->value, "inband"))
         ast_set_flag(&flags[0], SIP_DTMF_INBAND);
      else if (!strcasecmp(v->value, "rfc2833"))
         ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
      else if (!strcasecmp(v->value, "info"))
         ast_set_flag(&flags[0], SIP_DTMF_INFO);
      else if (!strcasecmp(v->value, "shortinfo"))
         ast_set_flag(&flags[0], SIP_DTMF_SHORTINFO);
      else if (!strcasecmp(v->value, "auto"))
         ast_set_flag(&flags[0], SIP_DTMF_AUTO);
      else {
         ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno);
         ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
      }
   } else if (!strcasecmp(v->name, "nat")) {
      ast_set_flag(&mask[0], SIP_NAT);
      ast_clear_flag(&flags[0], SIP_NAT);
      if (!strcasecmp(v->value, "never"))
         ast_set_flag(&flags[0], SIP_NAT_NEVER);
      else if (!strcasecmp(v->value, "route"))
         ast_set_flag(&flags[0], SIP_NAT_ROUTE);
      else if (ast_true(v->value))
         ast_set_flag(&flags[0], SIP_NAT_ALWAYS);
      else
         ast_set_flag(&flags[0], SIP_NAT_RFC3581);
   } else if (!strcasecmp(v->name, "canreinvite")) {
      ast_set_flag(&mask[0], SIP_REINVITE);
      ast_clear_flag(&flags[0], SIP_REINVITE);
      if (ast_true(v->value)) {
         ast_set_flag(&flags[0], SIP_CAN_REINVITE | SIP_CAN_REINVITE_NAT);
      } else if (!ast_false(v->value)) {
         char buf[64];
         char *word, *next = buf;

         ast_copy_string(buf, v->value, sizeof(buf));
         while ((word = strsep(&next, ","))) {
            if (!strcasecmp(word, "update")) {
               ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_CAN_REINVITE);
            } else if (!strcasecmp(word, "nonat")) {
               ast_set_flag(&flags[0], SIP_CAN_REINVITE);
               ast_clear_flag(&flags[0], SIP_CAN_REINVITE_NAT);
            } else {
               ast_log(LOG_WARNING, "Unknown canreinvite mode '%s' on line %d\n", v->value, v->lineno);
            }
         }
      }
   } else if (!strcasecmp(v->name, "insecure")) {
      ast_set_flag(&mask[0], SIP_INSECURE);
      ast_clear_flag(&flags[0], SIP_INSECURE);
      set_insecure_flags(&flags[0], v->value, v->lineno);   
   } else if (!strcasecmp(v->name, "progressinband")) {
      ast_set_flag(&mask[0], SIP_PROG_INBAND);
      ast_clear_flag(&flags[0], SIP_PROG_INBAND);
      if (ast_true(v->value))
         ast_set_flag(&flags[0], SIP_PROG_INBAND_YES);
      else if (strcasecmp(v->value, "never"))
         ast_set_flag(&flags[0], SIP_PROG_INBAND_NO);
   } else if (!strcasecmp(v->name, "promiscredir")) {
      ast_set_flag(&mask[0], SIP_PROMISCREDIR);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_PROMISCREDIR);
   } else if (!strcasecmp(v->name, "videosupport")) {
      if (!strcasecmp(v->value, "always")) {
         ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
         ast_set_flag(&flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
      } else {
         ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT);
         ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_VIDEOSUPPORT);
      }
   } else if (!strcasecmp(v->name, "textsupport")) {
      ast_set_flag(&mask[1], SIP_PAGE2_TEXTSUPPORT);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_TEXTSUPPORT);
      res = 1;
   } else if (!strcasecmp(v->name, "allowoverlap")) {
      ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWOVERLAP);
   } else if (!strcasecmp(v->name, "allowsubscribe")) {
      ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
   } else if (!strcasecmp(v->name, "t38pt_udptl")) {
      ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_UDPTL);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_UDPTL);
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
   } else if (!strcasecmp(v->name, "t38pt_rtp")) {
      ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_RTP);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_RTP);
   } else if (!strcasecmp(v->name, "t38pt_tcp")) {
      ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_TCP);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_TCP);
#endif
   } else if (!strcasecmp(v->name, "rfc2833compensate")) {
      ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE);
   } else if (!strcasecmp(v->name, "buggymwi")) {
      ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
   } else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
      ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
   } else
      res = 0;

   return res;
}

static int handle_incoming	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct sockaddr_in *	sin,
		int *	recount,
		int *	nounlock
	)			[static]

Handle incoming SIP requests (methods).

Note:

This is where all incoming requests go first

Definition at line 19554 of file chan_sip.c.

References __sip_ack(), append_history, ast_debug, ast_inet_ntoa(), ast_log(), ast_skip_blanks(), ast_string_field_set, ast_strlen_zero(), ast_verbose, check_pendings(), sip_request::debug, debug, DEFAULT_TRANS_TIMEOUT, extract_uri(), find_sdp(), get_header(), gettag(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_message(), handle_request_notify(), handle_request_options(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), handle_response(), sip_request::has_to_tag, sip_request::headers, cfsip_methods::id, sip_request::ignore, sip_pvt::initreq, INV_TERMINATED, len(), LOG_ERROR, LOG_NOTICE, LOG_WARNING, sip_request::method, sip_pvt::needdestroy, process_sdp(), REQ_OFFSET_TO_STR, SDP_T38_NONE, SIP_ACK, SIP_BYE, SIP_CANCEL, sip_debug_test_pvt(), SIP_INFO, SIP_INVITE, SIP_MESSAGE, sip_methods, SIP_NOTIFY, SIP_OPTIONS, SIP_REFER, SIP_REGISTER, SIP_RESPONSE, sip_scheddestroy(), SIP_SUBSCRIBE, cfsip_methods::text, sip_pvt::theirtag, transmit_response(), transmit_response_reliable(), transmit_response_with_allow(), and sip_pvt::useragent.

Referenced by handle_request_do(), and process_request_queue().

{
   /* Called with p->lock held, as well as p->owner->lock if appropriate, keeping things
      relatively static */
   const char *cmd;
   const char *cseq;
   const char *useragent;
   int seqno;
   int len;
   int respid;
   int res = 0;
   int debug = sip_debug_test_pvt(p);
   char *e;
   int error = 0;

   /* Get Method and Cseq */
   cseq = get_header(req, "Cseq");
   cmd = REQ_OFFSET_TO_STR(req, header[0]);

   /* Must have Cseq */
   if (ast_strlen_zero(cmd) || ast_strlen_zero(cseq)) {
      ast_log(LOG_ERROR, "Missing Cseq. Dropping this SIP message, it's incomplete.\n");
      error = 1;
   }
   if (!error && sscanf(cseq, "%d%n", &seqno, &len) != 1) {
      ast_log(LOG_ERROR, "No seqno in '%s'. Dropping incomplete message.\n", cmd);
      error = 1;
   }
   if (error) {
      if (!p->initreq.headers)   /* New call */
         p->needdestroy = 1; /* Make sure we destroy this dialog */
      return -1;
   }
   /* Get the command XXX */

   cmd = REQ_OFFSET_TO_STR(req, rlPart1);
   e = REQ_OFFSET_TO_STR(req, rlPart2);

   /* Save useragent of the client */
   useragent = get_header(req, "User-Agent");
   if (!ast_strlen_zero(useragent))
      ast_string_field_set(p, useragent, useragent);

   /* Find out SIP method for incoming request */
   if (req->method == SIP_RESPONSE) {  /* Response to our request */
      /* When we get here, we know this is a SIP dialog where we've sent
       * a request and have a response, or at least get a response
       * within an existing dialog. Do some sanity checks, then
       * possibly process the request. In all cases, there function
       * terminates at the end of this block
       */
      int ret = 0;

      if (p->ocseq < seqno && seqno != p->lastnoninvite) {
         ast_debug(1, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq);
         ret = -1;
      } else if (p->ocseq != seqno && seqno != p->lastnoninvite) {
         /* ignore means "don't do anything with it" but still have to 
          * respond appropriately.
          * But in this case this is a response already, so we really
          * have nothing to do with this message, and even setting the
          * ignore flag is pointless.
          */
         req->ignore = 1;
         append_history(p, "Ignore", "Ignoring this retransmit\n");
      } else if (e) {
         e = ast_skip_blanks(e);
         if (sscanf(e, "%d %n", &respid, &len) != 1) {
            ast_log(LOG_WARNING, "Invalid response: '%s'\n", e);
            /* XXX maybe should do ret = -1; */
         } else if (respid <= 0) {
            ast_log(LOG_WARNING, "Invalid SIP response code: '%d'\n", respid);
            /* XXX maybe should do ret = -1; */
         } else { /* finally, something worth processing */
            /* More SIP ridiculousness, we have to ignore bogus contacts in 100 etc responses */
            if ((respid == 200) || ((respid >= 300) && (respid <= 399)))
               extract_uri(p, req);
            handle_response(p, respid, e + len, req, seqno);
         }
      }
      return 0;
   }

   /* New SIP request coming in 
      (could be new request in existing SIP dialog as well...) 
    */         
   
   p->method = req->method;   /* Find out which SIP method they are using */
   ast_debug(4, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd); 

   if (p->icseq && (p->icseq > seqno) ) {
      if (p->pendinginvite && seqno == p->pendinginvite && (req->method == SIP_ACK || req->method == SIP_CANCEL)) {
         ast_debug(2, "Got CANCEL or ACK on INVITE with transactions in between.\n");
      }  else {
         ast_debug(1, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq);
         if (req->method != SIP_ACK)
            transmit_response(p, "503 Server error", req);  /* We must respond according to RFC 3261 sec 12.2 */
         return -1;
      }
   } else if (p->icseq &&
         p->icseq == seqno &&
         req->method != SIP_ACK &&
         (p->method != SIP_CANCEL || p->alreadygone)) {
      /* ignore means "don't do anything with it" but still have to 
         respond appropriately.  We do this if we receive a repeat of
         the last sequence number  */
      req->ignore = 1;
      ast_debug(3, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno);
   }
      
   if (seqno >= p->icseq)
      /* Next should follow monotonically (but not necessarily 
         incrementally -- thanks again to the genius authors of SIP --
         increasing */
      p->icseq = seqno;

   /* Find their tag if we haven't got it */
   if (ast_strlen_zero(p->theirtag)) {
      char tag[128];

      gettag(req, "From", tag, sizeof(tag));
      ast_string_field_set(p, theirtag, tag);
   }
   snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);

   if (pedanticsipchecking) {
      /* If this is a request packet without a from tag, it's not
         correct according to RFC 3261  */
      /* Check if this a new request in a new dialog with a totag already attached to it,
         RFC 3261 - section 12.2 - and we don't want to mess with recovery  */
      if (!p->initreq.headers && req->has_to_tag) {
         /* If this is a first request and it got a to-tag, it is not for us */
         if (!req->ignore && req->method == SIP_INVITE) {
            transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
            /* Will cease to exist after ACK */
         } else if (req->method != SIP_ACK) {
            transmit_response(p, "481 Call/Transaction Does Not Exist", req);
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         } else {
            ast_debug(1, "Got ACK for unknown dialog... strange.\n");
         }
         return res;
      }
   }

   if (!e && (p->method == SIP_INVITE || p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_NOTIFY)) {
      transmit_response(p, "400 Bad request", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return -1;
   }

   /* Handle various incoming SIP methods in requests */
   switch (p->method) {
   case SIP_OPTIONS:
      res = handle_request_options(p, req);
      break;
   case SIP_INVITE:
      res = handle_request_invite(p, req, debug, seqno, sin, recount, e, nounlock);
      break;
   case SIP_REFER:
      res = handle_request_refer(p, req, debug, seqno, nounlock);
      break;
   case SIP_CANCEL:
      res = handle_request_cancel(p, req);
      break;
   case SIP_BYE:
      res = handle_request_bye(p, req);
      break;
   case SIP_MESSAGE:
      res = handle_request_message(p, req);
      break;
   case SIP_SUBSCRIBE:
      res = handle_request_subscribe(p, req, sin, seqno, e);
      break;
   case SIP_REGISTER:
      res = handle_request_register(p, req, sin, e);
      break;
   case SIP_INFO:
      if (req->debug)
         ast_verbose("Receiving INFO!\n");
      if (!req->ignore) 
         handle_request_info(p, req);
      else  /* if ignoring, transmit response */
         transmit_response(p, "200 OK", req);
      break;
   case SIP_NOTIFY:
      res = handle_request_notify(p, req, sin, seqno, e);
      break;
   case SIP_ACK:
      /* Make sure we don't ignore this */
      if (seqno == p->pendinginvite) {
         p->invitestate = INV_TERMINATED;
         p->pendinginvite = 0;
         __sip_ack(p, seqno, 1 /* response */, 0);
         if (find_sdp(req)) {
            if (process_sdp(p, req, SDP_T38_NONE))
               return -1;
         }
         check_pendings(p);
      } else if (p->glareinvite == seqno) {
         /* handle ack for the 491 pending sent for glareinvite */
         p->glareinvite = 0;
         __sip_ack(p, seqno, 1, 0);
      }
      /* Got an ACK that we did not match. Ignore silently */
      if (!p->lastinvite && ast_strlen_zero(p->randdata))
         p->needdestroy = 1;
      break;
   default:
      transmit_response_with_allow(p, "501 Method Not Implemented", req, 0);
      ast_log(LOG_NOTICE, "Unknown SIP command '%s' from '%s'\n", 
         cmd, ast_inet_ntoa(p->sa.sin_addr));
      /* If this is some new method, and we don't have a call, destroy it now */
      if (!p->initreq.headers)
         p->needdestroy = 1;
      break;
   }
   return res;
}

static int handle_invite_replaces	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	debug,
		int	seqno,
		struct sockaddr_in *	sin
	)			[static]

Handle the transfer part of INVITE with a replaces: header, meaning a target pickup or an attended transfer. Used only once. XXX 'ignore' is unused.

Definition at line 17356 of file chan_sip.c.

References ast_channel::_state, append_history, ast_bridged_channel(), AST_CAUSE_SWITCH_CONGESTION, ast_channel_masquerade(), ast_channel_unlock, ast_debug, ast_frfree, ast_hangup(), ast_log(), ast_quiet_chan(), ast_read(), ast_set_flag, ast_setstate(), ast_state2str(), AST_STATE_DOWN, AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, DEFAULT_TRANS_TIMEOUT, dialog_unref(), f, FALSE, sip_pvt::flags, sip_request::ignore, LOG_ERROR, LOG_NOTICE, LOG_WARNING, ast_channel::name, sip_pvt::owner, sip_pvt::refer, sip_refer::refer_call, SIP_DEFER_BYE_ON_TRANSFER, sip_pvt_unlock, sip_scheddestroy(), transmit_response(), transmit_response_reliable(), transmit_response_with_sdp(), and XMIT_RELIABLE.

Referenced by handle_request_invite().

{
   struct ast_frame *f;
   int earlyreplace = 0;
   int oneleggedreplace = 0;     /* Call with no bridge, propably IVR or voice message */
   struct ast_channel *c = p->owner;   /* Our incoming call */
   struct ast_channel *replacecall = p->refer->refer_call->owner; /* The channel we're about to take over */
   struct ast_channel *targetcall;     /* The bridge to the take-over target */

   struct ast_channel *test;

   /* Check if we're in ring state */
   if (replacecall->_state == AST_STATE_RING)
      earlyreplace = 1;

   /* Check if we have a bridge */
   if (!(targetcall = ast_bridged_channel(replacecall))) {
      /* We have no bridge */
      if (!earlyreplace) {
         ast_debug(2, " Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", replacecall->name);
         oneleggedreplace = 1;
      }
   } 
   if (targetcall && targetcall->_state == AST_STATE_RINGING)
      ast_debug(4, "SIP transfer: Target channel is in ringing state\n");

   if (targetcall) 
      ast_debug(4, "SIP transfer: Invite Replace incoming channel should bridge to channel %s while hanging up channel %s\n", targetcall->name, replacecall->name); 
   else
      ast_debug(4, "SIP transfer: Invite Replace incoming channel should replace and hang up channel %s (one call leg)\n", replacecall->name); 

   if (req->ignore) {
      ast_log(LOG_NOTICE, "Ignoring this INVITE with replaces in a stupid way.\n");
      /* We should answer something here. If we are here, the
         call we are replacing exists, so an accepted 
         can't harm */
      transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE);
      /* Do something more clever here */
      ast_channel_unlock(c);
      sip_pvt_unlock(p->refer->refer_call);
      return 1;
   } 
   if (!c) {
      /* What to do if no channel ??? */
      ast_log(LOG_ERROR, "Unable to create new channel.  Invite/replace failed.\n");
      transmit_response_reliable(p, "503 Service Unavailable", req);
      append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      sip_pvt_unlock(p->refer->refer_call);
      return 1;
   }
   append_history(p, "Xfer", "INVITE/Replace received");
   /* We have three channels to play with
      channel c: New incoming call
      targetcall: Call from PBX to target
      p->refer->refer_call: SIP pvt dialog from transferer to pbx.
      replacecall: The owner of the previous
      We need to masq C into refer_call to connect to 
      targetcall;
      If we are talking to internal audio stream, target call is null.
   */

   /* Fake call progress */
   transmit_response(p, "100 Trying", req);
   ast_setstate(c, AST_STATE_RING);

   /* Masquerade the new call into the referred call to connect to target call 
      Targetcall is not touched by the masq */

   /* Answer the incoming call and set channel to UP state */
   transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE);
      
   ast_setstate(c, AST_STATE_UP);
   
   /* Stop music on hold and other generators */
   ast_quiet_chan(replacecall);
   ast_quiet_chan(targetcall);
   ast_debug(4, "Invite/Replaces: preparing to masquerade %s into %s\n", c->name, replacecall->name);
   /* Unlock clone, but not original (replacecall) */
   if (!oneleggedreplace)
      ast_channel_unlock(c);

   /* Unlock PVT */
   sip_pvt_unlock(p->refer->refer_call);

   /* Make sure that the masq does not free our PVT for the old call */
   if (! earlyreplace && ! oneleggedreplace )
      ast_set_flag(&p->refer->refer_call->flags[0], SIP_DEFER_BYE_ON_TRANSFER);  /* Delay hangup */
      
   /* Prepare the masquerade - if this does not happen, we will be gone */
   if(ast_channel_masquerade(replacecall, c))
      ast_log(LOG_ERROR, "Failed to masquerade C into Replacecall\n");
   else
      ast_debug(4, "Invite/Replaces: Going to masquerade %s into %s\n", c->name, replacecall->name);

   /* The masquerade will happen as soon as someone reads a frame from the channel */

   /* C should now be in place of replacecall */
   /* ast_read needs to lock channel */
   ast_channel_unlock(c);
   
   if (earlyreplace || oneleggedreplace ) {
      /* Force the masq to happen */
      if ((f = ast_read(replacecall))) {  /* Force the masq to happen */
         ast_frfree(f);
         f = NULL;
         ast_debug(4, "Invite/Replace:  Could successfully read frame from RING channel!\n");
      } else {
         ast_log(LOG_WARNING, "Invite/Replace:  Could not read frame from RING channel \n");
      }
      c->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
      if (!oneleggedreplace)
         ast_channel_unlock(replacecall);
   } else { /* Bridged call, UP channel */
      if ((f = ast_read(replacecall))) {  /* Force the masq to happen */
         /* Masq ok */
         ast_frfree(f);
         f = NULL;
         ast_debug(3, "Invite/Replace:  Could successfully read frame from channel! Masq done.\n");
      } else {
         ast_log(LOG_WARNING, "Invite/Replace:  Could not read frame from channel. Transfer failed\n");
      }
      ast_channel_unlock(replacecall);
   }
   sip_pvt_unlock(p->refer->refer_call);

   ast_setstate(c, AST_STATE_DOWN);
   ast_debug(4, "After transfer:----------------------------\n");
   ast_debug(4, " -- C:        %s State %s\n", c->name, ast_state2str(c->_state));
   if (replacecall)
      ast_debug(4, " -- replacecall:        %s State %s\n", replacecall->name, ast_state2str(replacecall->_state));
   if (p->owner) {
      ast_debug(4, " -- P->owner: %s State %s\n", p->owner->name, ast_state2str(p->owner->_state));
      test = ast_bridged_channel(p->owner);
      if (test)
         ast_debug(4, " -- Call bridged to P->owner: %s State %s\n", test->name, ast_state2str(test->_state));
      else
         ast_debug(4, " -- No call bridged to C->owner \n");
   } else 
      ast_debug(4, " -- No channel yet \n");
   ast_debug(4, "End After transfer:----------------------------\n");

   ast_channel_unlock(p->owner); /* Unlock new owner */
   if (!oneleggedreplace)
      sip_pvt_unlock(p);   /* Unlock SIP structure */

   /* The call should be down with no ast_channel, so hang it up */
   c->tech_pvt = dialog_unref(c->tech_pvt, "unref dialog c->tech_pvt");
   ast_hangup(c);
   return 0;
}

static int handle_request_bye	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Handle incoming BYE request.

Definition at line 19043 of file chan_sip.c.

References __sip_pretend_ack(), append_history, ast_async_goto(), ast_bridged_channel(), AST_CAUSE_PROTOCOL_ERROR, ast_clear_flag, AST_CONTROL_UNHOLD, ast_debug, ast_inet_ntoa(), ast_log(), ast_queue_control(), ast_queue_hangup(), ast_queue_hangup_with_cause(), ast_rtp_get_quality(), ast_rtp_set_vars(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, sip_pvt::callid, check_via(), context, sip_pvt::context, copy_request(), DEFAULT_TRANS_TIMEOUT, sip_pvt::do_history, sip_pvt::flags, get_also_info(), get_header(), sip_request::ignore, sip_pvt::initreq, INV_TERMINATED, sip_pvt::invitestate, IS_SIP_TECH, LOG_NOTICE, LOG_WARNING, sip_request::method, sip_pvt::owner, pbx_builtin_setvar_helper(), sip_pvt::pendinginvite, sip_pvt::recv, sip_pvt::refer, sip_refer::refer_to, sip_pvt::rtp, RTPQOS_JITTER, RTPQOS_LOSS, RTPQOS_RTT, RTPQOS_SUMMARY, sip_alreadygone(), sip_methods, SIP_OUTGOING, SIP_PAGE2_DIALOG_ESTABLISHED, sip_scheddestroy(), stop_media_flows(), stop_session_timer(), ast_channel::tech, ast_channel::tech_pvt, transmit_response(), transmit_response_reliable(), sip_pvt::trtp, and sip_pvt::vrtp.

Referenced by handle_incoming().

{
   struct ast_channel *c=NULL;
   int res;
   struct ast_channel *bridged_to;
   
   /* If we have an INCOMING invite that we haven't answered, terminate that transaction */
   if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !req->ignore) {
      transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
   }

   __sip_pretend_ack(p);

   p->invitestate = INV_TERMINATED;

   copy_request(&p->initreq, req);
   if (sipdebug)
      ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   check_via(p, req);
   sip_alreadygone(p);

   /* Get RTCP quality before end of call */
   if (p->do_history || p->owner) {
      struct ast_channel *bridge = p->owner ? ast_bridged_channel(p->owner) : NULL;
      char *videoqos, *textqos;

      if (p->rtp) {  
         if (p->do_history) {
            char *audioqos,
                 *audioqos_jitter,
                 *audioqos_loss,
                 *audioqos_rtt;

            audioqos        = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_SUMMARY);
            audioqos_jitter = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_JITTER);
            audioqos_loss   = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_LOSS);
            audioqos_rtt    = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_RTT);

            append_history(p, "RTCPaudio", "Quality:%s", audioqos);
            append_history(p, "RTCPaudioJitter", "Quality:%s", audioqos_jitter);
            append_history(p, "RTCPaudioLoss", "Quality:%s", audioqos_loss);
            append_history(p, "RTCPaudioRTT", "Quality:%s", audioqos_rtt);
         }
         
         if (p->owner) {
            ast_rtp_set_vars(p->owner, p->rtp);
         }
      }

      if (bridge) {
         struct sip_pvt *q = bridge->tech_pvt;

         if (IS_SIP_TECH(bridge->tech) && q && q->rtp)
            ast_rtp_set_vars(bridge, q->rtp);
      }

      if (p->vrtp) {
         videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY);
         if (p->do_history)
            append_history(p, "RTCPvideo", "Quality:%s", videoqos);
         if (p->owner)
            pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
      }

      if (p->trtp) {
         textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY);
         if (p->do_history)
            append_history(p, "RTCPtext", "Quality:%s", textqos);
         if (p->owner)
            pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", textqos);
      }
   }

   stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
   stop_session_timer(p); /* Stop Session-Timer */

   if (!ast_strlen_zero(get_header(req, "Also"))) {
      ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method.  Ask vendor to support REFER instead\n",
         ast_inet_ntoa(p->recv.sin_addr));
      if (ast_strlen_zero(p->context))
         ast_string_field_set(p, context, default_context);
      res = get_also_info(p, req);
      if (!res) {
         c = p->owner;
         if (c) {
            bridged_to = ast_bridged_channel(c);
            if (bridged_to) {
               /* Don't actually hangup here... */
               ast_queue_control(c, AST_CONTROL_UNHOLD);
               ast_async_goto(bridged_to, p->context, p->refer->refer_to, 1);
            } else
               ast_queue_hangup(p->owner);
         }
      } else {
         ast_log(LOG_WARNING, "Invalid transfer information from '%s'\n", ast_inet_ntoa(p->recv.sin_addr));
         if (p->owner)
            ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
      }
   } else if (p->owner) {
      ast_queue_hangup(p->owner);
      ast_debug(3, "Received bye, issuing owner hangup\n");
   } else {
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      ast_debug(3, "Received bye, no owner, selfdestruct soon.\n");
   }
   ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
   transmit_response(p, "200 OK", req);

   return 1;
}

static int handle_request_cancel	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Handle incoming CANCEL request.

Definition at line 18898 of file chan_sip.c.

References __sip_pretend_ack(), ast_debug, ast_queue_hangup(), AST_STATE_UP, ast_test_flag, check_via(), DEC_CALL_LIMIT, DEFAULT_TRANS_TIMEOUT, INV_CANCELLED, INV_TERMINATED, sip_pvt::invitestate, sip_dual::req, sip_alreadygone(), SIP_INC_COUNT, SIP_PAGE2_CALL_ONHOLD, sip_scheddestroy(), stop_media_flows(), transmit_response(), transmit_response_reliable(), and update_call_counter().

Referenced by handle_incoming().

{
      
   check_via(p, req);
   sip_alreadygone(p);

   /* At this point, we could have cancelled the invite at the same time
      as the other side sends a CANCEL. Our final reply with error code
      might not have been received by the other side before the CANCEL
      was sent, so let's just give up retransmissions and waiting for
      ACK on our error code. The call is hanging up any way. */
   if (p->invitestate == INV_TERMINATED)
      __sip_pretend_ack(p);
   else
      p->invitestate = INV_CANCELLED;
   
   if (p->owner && p->owner->_state == AST_STATE_UP) {
      /* This call is up, cancel is ignored, we need a bye */
      transmit_response(p, "200 OK", req);
      ast_debug(1, "Got CANCEL on an answered call. Ignoring... \n");
      return 0;
   }

   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) 
      update_call_counter(p, DEC_CALL_LIMIT);

   stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
   if (p->owner)
      ast_queue_hangup(p->owner);
   else
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   if (p->initreq.len > 0) {
      transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
      transmit_response(p, "200 OK", req);
      return 1;
   } else {
      transmit_response(p, "481 Call Leg Does Not Exist", req);
      return 0;
   }
}

static int handle_request_do	(	struct sip_request *	req,
		struct sockaddr_in *	sin
	)			[static]

Definition at line 19914 of file chan_sip.c.

References ao2_t_ref, append_history, ast_channel_trylock, ast_channel_unlock, ast_debug, ast_inet_ntoa(), AST_LIST_EMPTY, ast_log(), ast_mutex_lock(), ast_mutex_unlock(), AST_SCHED_DEL_UNREF, ast_str_reset(), ast_update_use_count(), ast_verbose, sip_pvt::callid, copy_socket_data(), sip_request::data, sip_request::debug, dialog_unref(), sip_pvt::do_history, find_call(), find_sip_method(), get_header(), get_transport(), handle_incoming(), sip_request::headers, sip_request::len, sip_request::lines, LOG_ERROR, lws2sws(), sip_request::method, ast_channel::name, netlock, sip_pvt::owner, parse_request(), process_request_queue(), queue_request(), sip_pvt::recv, REQ_OFFSET_TO_STR, sip_pvt::request_queue, sip_pvt::request_queue_sched_id, S_OR, sched, SIP_ACK, sip_debug_test_addr(), sip_pvt_unlock, sip_pvt::socket, sip_request::socket, ast_str::str, transmit_response(), and sip_socket::type.

Referenced by _sip_tcp_helper_thread(), and sipsock_read().

{
   struct sip_pvt *p;
   int recount = 0;
   int nounlock = 0;
   int lockretry;

   if (sip_debug_test_addr(sin)) /* Set the debug flag early on packet level */
      req->debug = 1;
   if (pedanticsipchecking)
      req->len = lws2sws(req->data->str, req->len);   /* Fix multiline headers */
   if (req->debug) {
      ast_verbose("\n<--- SIP read from %s://%s:%d --->\n%s\n<------------->\n", 
         get_transport(req->socket.type), ast_inet_ntoa(sin->sin_addr), 
         ntohs(sin->sin_port), req->data->str);
   }

   if (parse_request(req) == -1) { /* Bad packet, can't parse */
      ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
      return 1;
   }
   req->method = find_sip_method(REQ_OFFSET_TO_STR(req, rlPart1));

   if (req->debug)
      ast_verbose("--- (%d headers %d lines)%s ---\n", req->headers, req->lines, (req->headers + req->lines == 0) ? " Nat keepalive" : "");

   if (req->headers < 2) { /* Must have at least two headers */
      ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
      return 1;
   }

   /* Process request, with netlock held, and with usual deadlock avoidance */
   for (lockretry = 10; lockretry > 0; lockretry--) {
      ast_mutex_lock(&netlock);

      /* Find the active SIP dialog or create a new one */
      p = find_call(req, sin, req->method);  /* returns p locked */
      if (p == NULL) {
         ast_debug(1, "Invalid SIP message - rejected , no callid, len %d\n", req->len);
         ast_mutex_unlock(&netlock);
         return 1;
      }

      copy_socket_data(&p->socket, &req->socket);

      /* Go ahead and lock the owner if it has one -- we may need it */
      /* becaues this is deadlock-prone, we need to try and unlock if failed */
      if (!p->owner || !ast_channel_trylock(p->owner))
         break;   /* locking succeeded */

      if (lockretry != 1) {
         sip_pvt_unlock(p);
         ao2_t_ref(p, -1, "release p (from find_call) inside lockretry loop"); /* we'll look for it again, but p is dead now */
         ast_mutex_unlock(&netlock);
         /* Sleep for a very short amount of time */
         usleep(1);
      }
   }
   p->recv = *sin;

   if (p->do_history) /* This is a request or response, note what it was for */
      append_history(p, "Rx", "%s / %s / %s", req->data->str, get_header(req, "CSeq"), REQ_OFFSET_TO_STR(req, rlPart2));

   if (!lockretry) {
      if (!queue_request(p, req)) {
         /* the request has been queued for later handling */
         sip_pvt_unlock(p);
         ao2_t_ref(p, -1, "release p (from find_call) after queueing request");
         ast_mutex_unlock(&netlock);
         return 1;
      }

      if (p->owner)
         ast_log(LOG_ERROR, "Channel lock for %s could not be obtained, and request was unable to be queued.\n", S_OR(p->owner->name, "- no channel name ??? - "));
      ast_log(LOG_ERROR, "SIP transaction failed: %s \n", p->callid);
      if (req->method != SIP_ACK)
         transmit_response(p, "503 Server error", req);  /* We must respond according to RFC 3261 sec 12.2 */
      /* XXX We could add retry-after to make sure they come back */
      append_history(p, "LockFail", "Owner lock failed, transaction failed.");
      sip_pvt_unlock(p);
      ao2_t_ref(p, -1, "release p (from find_call) at end of lockretry"); /* p is gone after the return */
      ast_mutex_unlock(&netlock);
      return 1;
   }

   /* if there are queued requests on this sip_pvt, process them first, so that everything is
      handled in order
   */
   if (!AST_LIST_EMPTY(&p->request_queue)) {
      AST_SCHED_DEL_UNREF(sched, p->request_queue_sched_id, dialog_unref(p, "when you delete the request_queue_sched_id sched, you should dec the refcount for the stored dialog ptr"));
      process_request_queue(p, &recount, &nounlock);
   }

   if (handle_incoming(p, req, sin, &recount, &nounlock) == -1) {
      /* Request failed */
      ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
   }
      
   if (recount)
      ast_update_use_count();

   if (p->owner && !nounlock)
      ast_channel_unlock(p->owner);
   sip_pvt_unlock(p);
   ast_mutex_unlock(&netlock);
   ao2_t_ref(p, -1, "throw away dialog ptr from find_call at end of routine"); /* p is gone after the return */
   return 1;
}

static void handle_request_info	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Receive SIP INFO Message.

Todo:

Note: Doesn't read the duration of the DTMF. Should be fixed.

Definition at line 14834 of file chan_sip.c.

References ast_bridged_channel(), ast_cdr_setuserfield(), AST_CONTROL_FLASH, AST_CONTROL_VIDUPDATE, ast_copy_string(), ast_debug, ast_find_call_feature(), AST_FRAME_CONTROL, AST_FRAME_DTMF, ast_log(), ast_queue_control(), ast_queue_frame(), ast_rdlock_call_features(), ast_strlen_zero(), ast_test_flag, ast_unlock_call_features(), ast_verbose, buf, sip_pvt::callid, ast_channel::cdr, DEFAULT_TRANS_TIMEOUT, sip_history::event, ast_call_feature::exten, f, sip_pvt::flags, get_body(), get_header(), get_msg_text(), LOG_WARNING, sip_pvt::owner, sip_scheddestroy(), SIP_USECLIENTCODE, transmit_response(), and TRUE.

Referenced by handle_incoming().

{
   char buf[1024];
   unsigned int event;
   const char *c = get_header(req, "Content-Type");

   /* Need to check the media/type */
   if (!strcasecmp(c, "application/dtmf-relay") ||
       !strcasecmp(c, "application/vnd.nortelnetworks.digits")) {
      unsigned int duration = 0;

      if (!p->owner) {  /* not a PBX call */
         transmit_response(p, "481 Call leg/transaction does not exist", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return;
      }

      /* Try getting the "signal=" part */
      if (ast_strlen_zero(c = get_body(req, "Signal")) && ast_strlen_zero(c = get_body(req, "d"))) {
         ast_log(LOG_WARNING, "Unable to retrieve DTMF signal from INFO message from %s\n", p->callid);
         transmit_response(p, "200 OK", req); /* Should return error */
         return;
      } else {
         ast_copy_string(buf, c, sizeof(buf));
      }

      if (!ast_strlen_zero((c = get_body(req, "Duration"))))
         duration = atoi(c);
      if (!duration)
         duration = 100; /* 100 ms */


      if (ast_strlen_zero(buf)) {
         transmit_response(p, "200 OK", req);
         return;
      }

      if (buf[0] == '*')
         event = 10;
      else if (buf[0] == '#')
         event = 11;
      else if ((buf[0] >= 'A') && (buf[0] <= 'D'))
         event = 12 + buf[0] - 'A';
      else if (buf[0] == '!')
         event = 16;
      else
         event = atoi(buf);
      if (event == 16) {
         /* send a FLASH event */
         struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, };
         ast_queue_frame(p->owner, &f);
         if (sipdebug)
            ast_verbose("* DTMF-relay event received: FLASH\n");
      } else {
         /* send a DTMF event */
         struct ast_frame f = { AST_FRAME_DTMF, };
         if (event < 10) {
            f.subclass = '0' + event;
         } else if (event < 11) {
            f.subclass = '*';
         } else if (event < 12) {
            f.subclass = '#';
         } else if (event < 16) {
            f.subclass = 'A' + (event - 12);
         }
         f.len = duration;
         ast_queue_frame(p->owner, &f);
         if (sipdebug)
            ast_verbose("* DTMF-relay event received: %c\n", f.subclass);
      }
      transmit_response(p, "200 OK", req);
      return;
   } else if (!strcasecmp(c, "application/dtmf")) {
      /*! \todo Note: Doesn't read the duration of the DTMF. Should be fixed. */
      unsigned int duration = 0;

      if (!p->owner) {  /* not a PBX call */
         transmit_response(p, "481 Call leg/transaction does not exist", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return;
      }

      get_msg_text(buf, sizeof(buf), req, TRUE);
      duration = 100; /* 100 ms */

      if (ast_strlen_zero(buf)) {
         transmit_response(p, "200 OK", req);
         return;
      }
      event = atoi(buf);
      if (event == 16) {
         /* send a FLASH event */
         struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, };
         ast_queue_frame(p->owner, &f);
         if (sipdebug)
            ast_verbose("* DTMF-relay event received: FLASH\n");
      } else {
         /* send a DTMF event */
         struct ast_frame f = { AST_FRAME_DTMF, };
         if (event < 10) {
            f.subclass = '0' + event;
         } else if (event < 11) {
            f.subclass = '*';
         } else if (event < 12) {
            f.subclass = '#';
         } else if (event < 16) {
            f.subclass = 'A' + (event - 12);
         }
         f.len = duration;
         ast_queue_frame(p->owner, &f);
         if (sipdebug)
            ast_verbose("* DTMF-relay event received: %c\n", f.subclass);
      }
      transmit_response(p, "200 OK", req);
      return;

   } else if (!strcasecmp(c, "application/media_control+xml")) {
      /* Eh, we'll just assume it's a fast picture update for now */
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_VIDUPDATE);
      transmit_response(p, "200 OK", req);
      return;
   } else if (!ast_strlen_zero(c = get_header(req, "X-ClientCode"))) {
      /* Client code (from SNOM phone) */
      if (ast_test_flag(&p->flags[0], SIP_USECLIENTCODE)) {
         if (p->owner && p->owner->cdr)
            ast_cdr_setuserfield(p->owner, c);
         if (p->owner && ast_bridged_channel(p->owner) && ast_bridged_channel(p->owner)->cdr)
            ast_cdr_setuserfield(ast_bridged_channel(p->owner), c);
         transmit_response(p, "200 OK", req);
      } else {
         transmit_response(p, "403 Forbidden", req);
      }
      return;
   } else if (!ast_strlen_zero(c = get_header(req, "Record"))) {
      /* INFO messages generated by some phones to start/stop recording
         on phone calls. 
         OEJ: I think this should be something that is enabled/disabled
         per device. I don't want incoming callers to record calls in my
         pbx.
      */
      /* first, get the feature string, if it exists */
      struct ast_call_feature *feat;
      int j;
      struct ast_frame f = { AST_FRAME_DTMF, };

      ast_rdlock_call_features();
      feat = ast_find_call_feature("automon");
      if (!feat || ast_strlen_zero(feat->exten)) {
         ast_log(LOG_WARNING, "Recording requested, but no One Touch Monitor registered. (See features.conf)\n");
         /* 403 means that we don't support this feature, so don't request it again */
         transmit_response(p, "403 Forbidden", req);
         ast_unlock_call_features();
         return;
      } 
      /* Send the feature code to the PBX as DTMF, just like the handset had sent it */
      f.len = 100;
      for (j=0; j < strlen(feat->exten); j++) {
         f.subclass = feat->exten[j];
         ast_queue_frame(p->owner, &f);
         if (sipdebug)
            ast_verbose("* DTMF-relay event faked: %c\n", f.subclass);
      }
      ast_unlock_call_features();

      ast_debug(1, "Got a Request to Record the channel, state %s\n", c);
      transmit_response(p, "200 OK", req);
      return;
   } else if (ast_strlen_zero(c = get_header(req, "Content-Length")) || !strcasecmp(c, "0")) {
      /* This is probably just a packet making sure the signalling is still up, just send back a 200 OK */
      transmit_response(p, "200 OK", req);
      return;
   }

   /* Other type of INFO message, not really understood by Asterisk */
   /* if (get_msg_text(buf, sizeof(buf), req)) { */

   ast_log(LOG_WARNING, "Unable to parse INFO message from %s. Content %s\n", p->callid, buf);
   transmit_response(p, "415 Unsupported media type", req);
   return;
}

static int handle_request_invite	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	debug,
		int	seqno,
		struct sockaddr_in *	sin,
		int *	recount,
		char *	e,
		int *	nounlock
	)			[static]

Handle incoming INVITE request.

Note:

If the INVITE has a Replaces header, it is part of an attended transfer. If so, we do not go through the dial plan but tries to find the active call and masquerade into it

This is a spiral. What we need to do is to just change the outgoing INVITE so that it now routes to the new Request URI. Since we created the INVITE ourselves that should be all we need to do.

Todo:

XXX This needs to be reviewed. YOu don't change the request URI really, you route the packet correctly instead...

Definition at line 17811 of file chan_sip.c.

References __sip_ack(), ast_channel::_state, append_history, AST_CAUSE_CALL_REJECTED, AST_CAUSE_NORMAL_CLEARING, ast_channel_lock, ast_channel_unlock, ast_clear_flag, ast_debug, ast_hangup(), ast_log(), ast_null_frame, AST_PBX_CALL_LIMIT, AST_PBX_FAILED, ast_pbx_start(), AST_PBX_SUCCESS, ast_pickup_call(), ast_pickup_ext(), ast_queue_frame(), ast_rtp_setdtmf(), ast_rtp_setdtmfcompensate(), ast_sched_add(), AST_SCHED_DEL, ast_set_flag, ast_setstate(), ast_skip_blanks(), AST_STATE_DOWN, AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_decode(), ast_verbose, AUTH_CHALLENGE_SENT, AUTH_FAKE_AUTH, sip_pvt::autokillid, build_contact(), build_route(), sip_pvt::callid, sip_pvt::capability, check_user(), check_via(), context, sip_pvt::context, copy_request(), create_addr(), DEC_CALL_LIMIT, DEFAULT_TRANS_TIMEOUT, dialog_ref(), dialog_unref(), sip_pvt::do_history, exten, sip_pvt::exten, extract_uri(), FALSE, find_sdp(), sip_pvt::flags, get_destination(), get_header(), get_rdnis(), get_sip_pvt_byid_locked(), sip_pvt::glareinvite, handle_invite_replaces(), ast_channel::hangupcause, sip_request::headers, sip_pvt::icseq, sip_request::ignore, INC_CALL_LIMIT, sip_pvt::initreq, INV_COMPLETED, INV_PROCEEDING, INV_TERMINATED, sip_pvt::invitestate, sip_pvt::jointcapability, sip_pvt::lastinvite, LOG_DEBUG, LOG_ERROR, LOG_NOTICE, LOG_WARNING, make_our_tag(), sip_request::method, sip_pkt::method, msg, ast_channel::name, sip_pkt::next, option_debug, sip_pvt::owner, sip_pvt::packets, parse_minse(), parse_ok_contact(), parse_session_expires(), parse_sip_options(), sip_pvt::peername, sip_pvt::pendinginvite, process_sdp(), sip_pvt::refer, sip_refer::refer_call, REQ_OFFSET_TO_STR, sip_pvt::reqsipoptions, restart_session_timer(), sip_pkt::retransid, sip_pvt::rtp, S_OR, sched, SDP_T38_INITIATE, sip_pkt::seqno, sip_pvt::session_modify, SESSION_TIMER_MODE_ACCEPT, SESSION_TIMER_MODE_ORIGINATE, SESSION_TIMER_MODE_REFUSE, SESSION_TIMER_REFRESHER_AUTO, SESSION_TIMER_REFRESHER_UAS, sip_alreadygone(), sip_cancel_destroy(), SIP_DTMF, SIP_DTMF_RFC2833, SIP_INVITE, sip_methods, sip_new(), SIP_OPT_REPLACES, SIP_OPT_TIMER, SIP_OUTGOING, SIP_PAGE2_ALLOWOVERLAP, SIP_PAGE2_DIALOG_ESTABLISHED, SIP_PAGE2_RFC2833_COMPENSATE, sip_pvt_lock, sip_pvt_unlock, sip_refer_allocate(), sip_scheddestroy(), sip_st_alloc(), sip_t38_abort(), sip_uri_cmp(), sip_pvt::sipoptions, sip_st_dlg::st_active, sip_st_dlg::st_active_peer_ua, sip_st_dlg::st_expirys, st_get_mode(), st_get_refresher(), st_get_se(), sip_st_dlg::st_interval, sip_st_dlg::st_ref, start_session_timer(), t38properties::state, sip_pvt::stimer, strcasestr(), strsep(), sip_pvt::t38, T38_DISABLED, T38_ENABLED, T38_PEER_REINVITE, sip_pvt::t38id, sip_pvt::tag, transmit_fake_auth_response(), transmit_invite(), transmit_response(), transmit_response_reliable(), transmit_response_with_minse(), transmit_response_with_sdp(), transmit_response_with_t38_sdp(), transmit_response_with_unsupported(), TRUE, update_call_counter(), sip_pvt::username, XMIT_CRITICAL, XMIT_RELIABLE, and XMIT_UNRELIABLE.

Referenced by handle_incoming().

{
   int res = 1;
   int gotdest;
   const char *p_replaces;
   char *replace_id = NULL;
   const char *required;
   unsigned int required_profile = 0;
   struct ast_channel *c = NULL;    /* New channel */
   int reinvite = 0;
   int rtn;

   const char *p_uac_se_hdr;       /* UAC's Session-Expires header string                      */
   const char *p_uac_min_se;       /* UAC's requested Min-SE interval (char string)            */
   int uac_max_se = -1;            /* UAC's Session-Expires in integer format                  */
   int uac_min_se = -1;            /* UAC's Min-SE in integer format                           */
   int st_active = FALSE;          /* Session-Timer on/off boolean                             */
   int st_interval = 0;            /* Session-Timer negotiated refresh interval                */
   enum st_refresher st_ref;       /* Session-Timer session refresher                          */
   int dlg_min_se = -1;
   st_ref = SESSION_TIMER_REFRESHER_AUTO;

   /* Find out what they support */
   if (!p->sipoptions) {
      const char *supported = get_header(req, "Supported");
      if (!ast_strlen_zero(supported))
         parse_sip_options(p, supported);
   }

   /* Find out what they require */
   required = get_header(req, "Require");
   if (!ast_strlen_zero(required)) {
      required_profile = parse_sip_options(NULL, required);
      if (required_profile && required_profile != SIP_OPT_REPLACES && required_profile != SIP_OPT_TIMER) {
         /* At this point we only support REPLACES and Session-Timer */
         transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, required);
         ast_log(LOG_WARNING, "Received SIP INVITE with unsupported required extension: %s\n", required);
         p->invitestate = INV_COMPLETED;
         if (!p->lastinvite)
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return -1;
      }
   }

   /* The option tags may be present in Supported: or Require: headers.
   Include the Require: option tags for further processing as well */
   p->sipoptions |= required_profile;
   p->reqsipoptions = required_profile;

   /* Check if this is a loop */
   if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->owner->_state != AST_STATE_UP)) {
      /* This is a call to ourself.  Send ourselves an error code and stop
         processing immediately, as SIP really has no good mechanism for
         being able to call yourself */
      /* If pedantic is on, we need to check the tags. If they're different, this is
         in fact a forked call through a SIP proxy somewhere. */
      int different;
      char *initial_rlPart2 = REQ_OFFSET_TO_STR(&p->initreq, rlPart2);
      char *this_rlPart2 = REQ_OFFSET_TO_STR(req, rlPart2);
      if (pedanticsipchecking)
         different = sip_uri_cmp(initial_rlPart2, this_rlPart2);
      else
         different = strcmp(initial_rlPart2, this_rlPart2);
      if (!different) {
         transmit_response(p, "482 Loop Detected", req);
         p->invitestate = INV_COMPLETED;
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return 0;
      } else {
         /*! This is a spiral. What we need to do is to just change the outgoing INVITE
          * so that it now routes to the new Request URI. Since we created the INVITE ourselves
          * that should be all we need to do.
          * 
          * \todo XXX This needs to be reviewed.  YOu don't change the request URI really, you route the packet
          * correctly instead...
          */
         char *uri = ast_strdupa(this_rlPart2);
         char *at = strchr(uri, '@');
         char *peerorhost;
         struct sip_pkt *pkt = NULL;
         if (option_debug > 2) {
            ast_log(LOG_DEBUG, "Potential spiral detected. Original RURI was %s, new RURI is %s\n", initial_rlPart2, this_rlPart2);
         }
         if (at) {
            *at = '\0';
         }
         /* Parse out "sip:" */
         if ((peerorhost = strchr(uri, ':'))) {
            *peerorhost++ = '\0';
         }
         create_addr(p, peerorhost, NULL, 0);
         ast_string_field_set(p, theirtag, NULL);
         for (pkt = p->packets; pkt; pkt = pkt->next) {
            if (pkt->seqno == p->icseq && pkt->method == SIP_INVITE) {
               AST_SCHED_DEL(sched, pkt->retransid);
            }
         }
         return transmit_invite(p, SIP_INVITE, 1, 3);
      }
   }

   if (!req->ignore && p->pendinginvite) {
      if (!ast_test_flag(&p->flags[0], SIP_OUTGOING) && ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
         /* We have received a reINVITE on an incoming call to which we have sent a 200 OK but not yet received
          * an ACK. According to RFC 5407, Section 3.1.4, the proper way to handle this race condition is to accept
          * the reINVITE since we have established a dialog.
          */
          
         /* Note that this will both clear the pendinginvite flag and cancel the 
          * retransmission of the 200 OK. Basically, we're accepting this reINVITE as both an ACK
          * and a reINVITE in one request.
          */
         __sip_ack(p, p->lastinvite, 1, 0);
      } else {
         /* We already have a pending invite. Sorry. You are on hold. */
         p->glareinvite = seqno;
         transmit_response_reliable(p, "491 Request Pending", req);
         ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
         /* Don't destroy dialog here */
         return 0;
      }
   }

   p_replaces = get_header(req, "Replaces");
   if (!ast_strlen_zero(p_replaces)) {
      /* We have a replaces header */
      char *ptr;
      char *fromtag = NULL;
      char *totag = NULL;
      char *start, *to;
      int error = 0;

      if (p->owner) {
         ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
         transmit_response_reliable(p, "400 Bad request", req);   /* The best way to not not accept the transfer */
         /* Do not destroy existing call */
         return -1;
      }

      if (sipdebug)
         ast_debug(3, "INVITE part of call transfer. Replaces [%s]\n", p_replaces);
      /* Create a buffer we can manipulate */
      replace_id = ast_strdupa(p_replaces);
      ast_uri_decode(replace_id);

      if (!p->refer && !sip_refer_allocate(p)) {
         transmit_response_reliable(p, "500 Server Internal Error", req);
         append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         p->invitestate = INV_COMPLETED;
         return -1;
      }

      /*  Todo: (When we find phones that support this)
         if the replaces header contains ";early-only"
         we can only replace the call in early
         stage, not after it's up.

         If it's not in early mode, 486 Busy.
      */
      
      /* Skip leading whitespace */
      replace_id = ast_skip_blanks(replace_id);

      start = replace_id;
      while ( (ptr = strsep(&start, ";")) ) {
         ptr = ast_skip_blanks(ptr); /* XXX maybe unnecessary ? */
         if ( (to = strcasestr(ptr, "to-tag=") ) )
            totag = to + 7;   /* skip the keyword */
         else if ( (to = strcasestr(ptr, "from-tag=") ) ) {
            fromtag = to + 9; /* skip the keyword */
            fromtag = strsep(&fromtag, "&"); /* trim what ? */
         }
      }

      if (sipdebug) 
         ast_debug(4, "Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n", replace_id, fromtag ? fromtag : "<no from tag>", totag ? totag : "<no to tag>");


      /* Try to find call that we are replacing 
         If we have a Replaces  header, we need to cancel that call if we succeed with this call 
      */
      if ((p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
         transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
         error = 1;
      }

      /* At this point, bot the pvt and the owner of the call to be replaced is locked */

      /* The matched call is the call from the transferer to Asterisk .
         We want to bridge the bridged part of the call to the 
         incoming invite, thus taking over the refered call */

      if (p->refer->refer_call == p) {
         ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
         p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
         transmit_response_reliable(p, "400 Bad request", req);   /* The best way to not not accept the transfer */
         error = 1;
      }

      if (!error && !p->refer->refer_call->owner) {
         /* Oops, someting wrong anyway, no owner, no call */
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
         /* Check for better return code */
         transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
         error = 1;
      }

      if (!error && p->refer->refer_call->owner->_state != AST_STATE_RINGING && p->refer->refer_call->owner->_state != AST_STATE_RING && p->refer->refer_call->owner->_state != AST_STATE_UP ) {
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
         transmit_response_reliable(p, "603 Declined (Replaces)", req);
         error = 1;
      }

      if (error) {   /* Give up this dialog */
         append_history(p, "Xfer", "INVITE/Replace Failed.");
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         sip_pvt_unlock(p);
         if (p->refer->refer_call) {
            sip_pvt_unlock(p->refer->refer_call);
            if (p->refer->refer_call->owner) {
               ast_channel_unlock(p->refer->refer_call->owner);
            }
         }
         p->invitestate = INV_COMPLETED;
         return -1;
      }
   }

   /* Check if this is an INVITE that sets up a new dialog or
      a re-invite in an existing dialog */

   if (!req->ignore) {
      int newcall = (p->initreq.headers ? TRUE : FALSE);

      if (sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      /* This also counts as a pending invite */
      p->pendinginvite = seqno;
      check_via(p, req);

      copy_request(&p->initreq, req);     /* Save this INVITE as the transaction basis */
      if (sipdebug)
         ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
      if (!p->owner) {  /* Not a re-invite */
         if (debug)
            ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
         if (newcall)
            append_history(p, "Invite", "New call: %s", p->callid);
         parse_ok_contact(p, req);
      } else { /* Re-invite on existing call */
         ast_clear_flag(&p->flags[0], SIP_OUTGOING);  /* This is now an inbound dialog */
         /* Handle SDP here if we already have an owner */
         if (find_sdp(req)) {
            if (process_sdp(p, req, SDP_T38_INITIATE)) {
               transmit_response_reliable(p, "488 Not acceptable here", req);
               if (!p->lastinvite)
                  sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
               return -1;
            }
         } else {
            p->jointcapability = p->capability;
            ast_debug(1, "Hm....  No sdp for the moment\n");
         }
         if (p->do_history) /* This is a response, note what it was for */
            append_history(p, "ReInv", "Re-invite received");
      }
   } else if (debug)
      ast_verbose("Ignoring this INVITE request\n");

   
   if (!p->lastinvite && !req->ignore && !p->owner) {
      /* This is a new invite */
      /* Handle authentication if this is our first invite */
      res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
      if (res == AUTH_CHALLENGE_SENT) {
         p->invitestate = INV_COMPLETED;     /* Needs to restart in another INVITE transaction */
         return 0;
      }
      if (res < 0) { /* Something failed in authentication */
         if (res == AUTH_FAKE_AUTH) {
            ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
            transmit_fake_auth_response(p, SIP_INVITE, req, XMIT_RELIABLE);
         } else {
            ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
            transmit_response_reliable(p, "403 Forbidden", req);
         }
         p->invitestate = INV_COMPLETED;  
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         ast_string_field_set(p, theirtag, NULL);
         return 0;
      }

      /* We have a succesful authentication, process the SDP portion if there is one */
      if (find_sdp(req)) {
         if (process_sdp(p, req, SDP_T38_INITIATE)) {
            /* Unacceptable codecs */
            transmit_response_reliable(p, "488 Not acceptable here", req);
            p->invitestate = INV_COMPLETED;  
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            ast_debug(1, "No compatible codecs for this SIP call.\n");
            return -1;
         }
      } else { /* No SDP in invite, call control session */
         p->jointcapability = p->capability;
         ast_debug(2, "No SDP in Invite, third party call control\n");
      }

      /* Queue NULL frame to prod ast_rtp_bridge if appropriate */
      /* This seems redundant ... see !p-owner above */
      if (p->owner)
         ast_queue_frame(p->owner, &ast_null_frame);


      /* Initialize the context if it hasn't been already */
      if (ast_strlen_zero(p->context))
         ast_string_field_set(p, context, default_context);


      /* Check number of concurrent calls -vs- incoming limit HERE */
      ast_debug(1, "Checking SIP call limits for device %s\n", p->username);
      if ((res = update_call_counter(p, INC_CALL_LIMIT))) {
         if (res < 0) {
            ast_log(LOG_NOTICE, "Failed to place call for device %s, too many calls\n", p->username);
            transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            p->invitestate = INV_COMPLETED;  
         }
         return 0;
      }
      gotdest = get_destination(p, NULL); /* Get destination right away */
      get_rdnis(p, NULL);        /* Get redirect information */
      extract_uri(p, req);       /* Get the Contact URI */
      build_contact(p);       /* Build our contact header */

      if (p->rtp) {
         ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
         ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
      }

      if (!replace_id && gotdest) { /* No matching extension found */
         if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP))
            transmit_response_reliable(p, "484 Address Incomplete", req);
         else {
            char *decoded_exten = ast_strdupa(p->exten);
            
            transmit_response_reliable(p, "404 Not Found", req);
            ast_uri_decode(decoded_exten);
            ast_log(LOG_NOTICE, "Call from '%s' to extension"
               " '%s' rejected because extension not found.\n",
               S_OR(p->username, p->peername), decoded_exten);
         }
         p->invitestate = INV_COMPLETED;  
         update_call_counter(p, DEC_CALL_LIMIT);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return 0;
      } else {

         /* If no extension was specified, use the s one */
         /* Basically for calling to IP/Host name only */
         if (ast_strlen_zero(p->exten))
            ast_string_field_set(p, exten, "s");
         /* Initialize our tag */   

         make_our_tag(p->tag, sizeof(p->tag));
         /* First invitation - create the channel */
         c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL));
         *recount = 1;

         /* Save Record-Route for any later requests we make on this dialogue */
         build_route(p, req, 0);

         if (c) {
            /* Pre-lock the call */
            ast_channel_lock(c);
         }
      }
   } else {
      if (sipdebug) {
         if (!req->ignore)
            ast_debug(2, "Got a SIP re-invite for call %s\n", p->callid);
         else
            ast_debug(2, "Got a SIP re-transmit of INVITE for call %s\n", p->callid);
      }
      if (!req->ignore)
         reinvite = 1;
      c = p->owner;
   }

   /* Session-Timers */
   if (p->sipoptions == SIP_OPT_TIMER) {
      /* The UAC has requested session-timers for this session. Negotiate
      the session refresh interval and who will be the refresher */
      ast_debug(2, "Incoming INVITE with 'timer' option enabled\n");

      /* Allocate Session-Timers struct w/in the dialog */
      if (!p->stimer)
         sip_st_alloc(p);

      /* Parse the Session-Expires header */
      p_uac_se_hdr = get_header(req, "Session-Expires");
      if (!ast_strlen_zero(p_uac_se_hdr)) {
         rtn = parse_session_expires(p_uac_se_hdr, &uac_max_se, &st_ref);
         if (rtn != 0) {
            transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req);
            p->invitestate = INV_COMPLETED;
            if (!p->lastinvite) {
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
            return -1;
         }
      }

      /* Parse the Min-SE header */
      p_uac_min_se = get_header(req, "Min-SE");
      if (!ast_strlen_zero(p_uac_min_se)) {
         rtn = parse_minse(p_uac_min_se, &uac_min_se); 
         if (rtn != 0) {
               transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
                     p->invitestate = INV_COMPLETED;
                     if (!p->lastinvite) {
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
            return -1;
         }
      }

      dlg_min_se = st_get_se(p, FALSE);
      switch (st_get_mode(p)) {
      case SESSION_TIMER_MODE_ACCEPT:
      case SESSION_TIMER_MODE_ORIGINATE:
         if (uac_max_se > 0 && uac_max_se < dlg_min_se) {
            transmit_response_with_minse(p, "422 Session Interval Too Small", req, dlg_min_se);
            p->invitestate = INV_COMPLETED;
            if (!p->lastinvite) {
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
            return -1;
         }

         p->stimer->st_active_peer_ua = TRUE;
         st_active = TRUE;
         if (st_ref == SESSION_TIMER_REFRESHER_AUTO) {
            st_ref = st_get_refresher(p);
         }

         if (uac_max_se > 0) {
            int dlg_max_se = st_get_se(p, TRUE);
            if (dlg_max_se >= uac_min_se) {
               st_interval = (uac_max_se < dlg_max_se) ? uac_max_se : dlg_max_se;
            } else {
               st_interval = uac_max_se;
            }
         } else {
            st_interval = uac_min_se;
         }
         break;

      case SESSION_TIMER_MODE_REFUSE:
         if (p->reqsipoptions == SIP_OPT_TIMER) {
            transmit_response_with_unsupported(p, "420 Option Disabled", req, required);
            ast_log(LOG_WARNING, "Received SIP INVITE with supported but disabled option: %s\n", required);
            p->invitestate = INV_COMPLETED;
            if (!p->lastinvite) {
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
            return -1;
         }
         break;

      default:
         ast_log(LOG_ERROR, "Internal Error %d at %s:%d\n", st_get_mode(p), __FILE__, __LINE__);
         break;
      }
   } else {
      /* The UAC did not request session-timers.  Asterisk (UAS), will now decide
      (based on session-timer-mode in sip.conf) whether to run session-timers for
      this session or not. */
      switch (st_get_mode(p)) {
      case SESSION_TIMER_MODE_ORIGINATE:
         st_active = TRUE;
         st_interval = st_get_se(p, TRUE);
         st_ref = SESSION_TIMER_REFRESHER_UAS;
         p->stimer->st_active_peer_ua = FALSE;
         break;

      default:
         break;
      }
   }

   if (reinvite == 0) {
      /* Session-Timers: Start session refresh timer based on negotiation/config */
      if (st_active == TRUE) {
         p->stimer->st_active   = TRUE;
         p->stimer->st_interval = st_interval;
         p->stimer->st_ref      = st_ref;
         start_session_timer(p);
      }
   } else {
      if (p->stimer->st_active == TRUE) {
         /* Session-Timers:  A re-invite request sent within a dialog will serve as 
         a refresh request, no matter whether the re-invite was sent for refreshing 
         the session or modifying it.*/
         ast_debug (2, "Restarting session-timers on a refresh - %s\n", p->callid);

         /* The UAC may be adjusting the session-timers mid-session */
         if (st_interval > 0) {
            p->stimer->st_interval = st_interval;
            p->stimer->st_ref      = st_ref;
         }

         restart_session_timer(p);
         if (p->stimer->st_expirys > 0) {
            p->stimer->st_expirys--;
         }
      }
   }

   if (!req->ignore && p)
      p->lastinvite = seqno;

   if (replace_id) {    /* Attended transfer or call pickup - we're the target */
      /* Go and take over the target call */
      if (sipdebug)
         ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid);
      return handle_invite_replaces(p, req, debug, seqno, sin);
   }


   if (c) { /* We have a call  -either a new call or an old one (RE-INVITE) */
      switch(c->_state) {
      case AST_STATE_DOWN:
         ast_debug(2, "%s: New call is still down.... Trying... \n", c->name);
         transmit_response(p, "100 Trying", req);
         p->invitestate = INV_PROCEEDING;
         ast_setstate(c, AST_STATE_RING);
         if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
            enum ast_pbx_result result;

            result = ast_pbx_start(c);

            switch(result) {
            case AST_PBX_FAILED:
               ast_log(LOG_WARNING, "Failed to start PBX :(\n");
               p->invitestate = INV_COMPLETED;
               transmit_response_reliable(p, "503 Unavailable", req);
               break;
            case AST_PBX_CALL_LIMIT:
               ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
               p->invitestate = INV_COMPLETED;
               transmit_response_reliable(p, "480 Temporarily Unavailable", req);
               break;
            case AST_PBX_SUCCESS:
               /* nothing to do */
               break;
            }

            if (result) {

               /* Unlock locks so ast_hangup can do its magic */
               ast_channel_unlock(c);
               sip_pvt_unlock(p);
               ast_hangup(c);
               sip_pvt_lock(p);
               c = NULL;
            }
         } else { /* Pickup call in call group */
            ast_channel_unlock(c);
            *nounlock = 1;
            if (ast_pickup_call(c)) {
               ast_log(LOG_NOTICE, "Nothing to pick up for %s\n", p->callid);
               transmit_response_reliable(p, "503 Unavailable", req);
               sip_alreadygone(p);
               /* Unlock locks so ast_hangup can do its magic */
               sip_pvt_unlock(p);
               c->hangupcause = AST_CAUSE_CALL_REJECTED;
            } else {
               sip_pvt_unlock(p);
               ast_setstate(c, AST_STATE_DOWN);
               c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
            }
            p->invitestate = INV_COMPLETED;
            ast_hangup(c);
            sip_pvt_lock(p);
            c = NULL;
         }
         break;
      case AST_STATE_RING:
         transmit_response(p, "100 Trying", req);
         p->invitestate = INV_PROCEEDING;
         break;
      case AST_STATE_RINGING:
         transmit_response(p, "180 Ringing", req);
         p->invitestate = INV_PROCEEDING;
         break;
      case AST_STATE_UP:
         ast_debug(2, "%s: This call is UP.... \n", c->name);

         transmit_response(p, "100 Trying", req);

         if (p->t38.state == T38_PEER_REINVITE) {
            p->t38id = ast_sched_add(sched, 5000, sip_t38_abort, dialog_ref(p, "passing dialog ptr into sched structure based on t38id for sip_t38_abort."));
         } else if (p->t38.state == T38_ENABLED) {
            ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
            transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)));
         } else if (p->t38.state == T38_DISABLED) {
            /* If this is not a re-invite or something to ignore - it's critical */
            ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
            transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE:TRUE); 
         }

         p->invitestate = INV_TERMINATED;
         break;
      default:
         ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
         transmit_response(p, "100 Trying", req);
         break;
      }
   } else {
      if (p && (p->autokillid == -1)) {
         const char *msg;

         if (!p->jointcapability)
            msg = "488 Not Acceptable Here (codec error)";
         else {
            ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
            msg = "503 Unavailable";
         }
         transmit_response_reliable(p, msg, req);
         p->invitestate = INV_COMPLETED;
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      }
   }
   return res;
}

static int handle_request_message	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Handle incoming MESSAGE request.

Definition at line 19155 of file chan_sip.c.

References ast_verbose, sip_request::debug, sip_request::ignore, receive_message(), and transmit_response().

Referenced by handle_incoming().

{
   if (!req->ignore) {
      if (req->debug)
         ast_verbose("Receiving message!\n");
      receive_message(p, req);
   } else
      transmit_response(p, "202 Accepted", req);
   return 1;
}

static int handle_request_notify	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct sockaddr_in *	sin,
		int	seqno,
		char *	e
	)			[static]

Handle incoming notifications.

Definition at line 17178 of file chan_sip.c.

References ast_debug, ast_log(), ast_skip_blanks(), buf, sip_pvt::callid, DEFAULT_TRANS_TIMEOUT, FALSE, get_header(), get_msg_text(), sip_pvt::lastinvite, LOG_NOTICE, LOG_WARNING, sip_scheddestroy(), transmit_response(), and TRUE.

Referenced by handle_incoming().

{
   /* This is mostly a skeleton for future improvements */
   /* Mostly created to return proper answers on notifications on outbound REFER's */
   int res = 0;
   const char *event = get_header(req, "Event");
   char *eventid = NULL;
   char *sep;

   if( (sep = strchr(event, ';')) ) {  /* XXX bug here - overwriting string ? */
      *sep++ = '\0';
      eventid = sep;
   }
   
   if (sipdebug)
      ast_debug(2, "Got NOTIFY Event: %s\n", event);

   if (strcmp(event, "refer")) {
      /* We don't understand this event. */
      /* Here's room to implement incoming voicemail notifications :-) */
      transmit_response(p, "489 Bad event", req);
      res = -1;
   } else {
      /* Save nesting depth for now, since there might be other events we will
         support in the future */

      /* Handle REFER notifications */

      char buf[1024];
      char *cmd, *code;
      int respcode;
      int success = TRUE;

      /* EventID for each transfer... EventID is basically the REFER cseq 

       We are getting notifications on a call that we transfered
       We should hangup when we are getting a 200 OK in a sipfrag
       Check if we have an owner of this event */
      
      /* Check the content type */
      if (strncasecmp(get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) {
         /* We need a sipfrag */
         transmit_response(p, "400 Bad request", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return -1;
      }

      /* Get the text of the attachment */
      if (get_msg_text(buf, sizeof(buf), req, TRUE)) {
         ast_log(LOG_WARNING, "Unable to retrieve attachment from NOTIFY %s\n", p->callid);
         transmit_response(p, "400 Bad request", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return -1;
      }

      /*
      From the RFC...
      A minimal, but complete, implementation can respond with a single
      NOTIFY containing either the body:
         SIP/2.0 100 Trying
      
      if the subscription is pending, the body:
         SIP/2.0 200 OK
      if the reference was successful, the body:
         SIP/2.0 503 Service Unavailable
      if the reference failed, or the body:
         SIP/2.0 603 Declined

      if the REFER request was accepted before approval to follow the
      reference could be obtained and that approval was subsequently denied
      (see Section 2.4.7).
      
      If there are several REFERs in the same dialog, we need to
      match the ID of the event header...
      */
      ast_debug(3, "* SIP Transfer NOTIFY Attachment: \n---%s\n---\n", buf);
      cmd = ast_skip_blanks(buf);
      code = cmd;
      /* We are at SIP/2.0 */
      while(*code && (*code > 32)) {   /* Search white space */
         code++;
      }
      *code++ = '\0';
      code = ast_skip_blanks(code);
      sep = code;
      sep++;
      while(*sep && (*sep > 32)) {  /* Search white space */
         sep++;
      }
      *sep++ = '\0';       /* Response string */
      respcode = atoi(code);
      switch (respcode) {
      case 100:   /* Trying: */
      case 101:   /* dialog establishment */
         /* Don't do anything yet */
         break;
      case 183:   /* Ringing: */
         /* Don't do anything yet */
         break;
      case 200:   /* OK: The new call is up, hangup this call */
         /* Hangup the call that we are replacing */
         break;
      case 301: /* Moved permenantly */
      case 302: /* Moved temporarily */
         /* Do we get the header in the packet in this case? */
         success = FALSE;
         break;
      case 503:   /* Service Unavailable: The new call failed */
            /* Cancel transfer, continue the call */
         success = FALSE;
         break;
      case 603:   /* Declined: Not accepted */
            /* Cancel transfer, continue the current call */
         success = FALSE;
         break;
      }
      if (!success) {
         ast_log(LOG_NOTICE, "Transfer failed. Sorry. Nothing further to do with this call\n");
      }
      
      /* Confirm that we received this packet */
      transmit_response(p, "200 OK", req);
   };

   if (!p->lastinvite)
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

   return res;
}

static int handle_request_options	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Handle incoming OPTIONS request An OPTIONS request should be answered like an INVITE from the same UA, including SDP.

XXX get_destination assumes we're already authenticated. This means that a request from a known device (peer) will end up in the wrong context if this is out-of-dialog. However, we want to handle OPTIONS as light as possible, so we might want to have a configuration option whether we care or not. Some devices use this for testing capabilities, which means that we need to match device to answer with proper capabilities (including SDP).

Todo:

Fix handle_request_options device handling with optional authentication (this needs to be fixed in 1.4 as well)

Definition at line 17311 of file chan_sip.c.

References ast_shutting_down(), ast_string_field_set, ast_strlen_zero(), build_contact(), context, sip_pvt::context, DEFAULT_TRANS_TIMEOUT, get_destination(), sip_pvt::lastinvite, sip_scheddestroy(), and transmit_response_with_allow().

Referenced by handle_incoming().

{
   int res;

   /*! XXX get_destination assumes we're already authenticated. This means that a request from
      a known device (peer) will end up in the wrong context if this is out-of-dialog.
      However, we want to handle OPTIONS as light as possible, so we might want to have
      a configuration option whether we care or not. Some devices use this for testing
      capabilities, which means that we need to match device to answer with proper 
      capabilities (including SDP).
      \todo Fix handle_request_options device handling with optional authentication
         (this needs to be fixed in 1.4 as well)
   */

   if (p->lastinvite) {
      /* if this is a request in an active dialog, just confirm that the dialog exists. */
      transmit_response_with_allow(p, "200 OK", req, 0);
      return 0;
   }

   res = get_destination(p, req);
   build_contact(p);

   if (ast_strlen_zero(p->context))
      ast_string_field_set(p, context, default_context);

   if (ast_shutting_down())
      transmit_response_with_allow(p, "503 Unavailable", req, 0);
   else if (res < 0)
      transmit_response_with_allow(p, "404 Not Found", req, 0);
   else 
      transmit_response_with_allow(p, "200 OK", req, 0);

   /* Destroy if this OPTIONS was the opening request, but not if
      it's in the middle of a normal call flow. */
   sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

   return res;
}

static int handle_request_refer	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	debug,
		int	seqno,
		int *	nounlock
	)			[static]

Definition at line 18615 of file chan_sip.c.

References sip_pvt::allowtransfer, append_history, ast_async_goto(), ast_bridged_channel(), AST_CAUSE_NORMAL_CLEARING, ast_channel_unlock, ast_clear_flag, AST_CONTROL_UNHOLD, ast_debug, AST_LIST_EMPTY, ast_parking_ext(), ast_queue_control(), ast_set_flag, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verbose, sip_refer::attendedtransfer, sip_pvt::callid, sip_dual::chan1, sip_dual::chan2, check_sip_domain(), context, sip_pvt::context, copy_request(), sip_request::data, sip_request::debug, EVENT_FLAG_CALL, FALSE, sip_pvt::flags, get_refer_info(), ast_channel::hangupcause, sip_request::ignore, local_attended_transfer(), sip_refer::localtransfer, manager_event, ast_channel::name, sip_pvt::needdestroy, sip_pvt::owner, pbx_builtin_setvar_helper(), sip_pvt::refer, REFER_200OK, REFER_FAILED, REFER_SENT, sip_refer::refer_to, sip_refer::refer_to_context, sip_refer::refer_to_domain, sip_refer::referred_by, sip_refer::replaces_callid, sip_refer::replaces_callid_fromtag, sip_refer::replaces_callid_totag, sip_dual::req, sip_alreadygone(), SIP_DEFER_BYE_ON_TRANSFER, SIP_GOTREFER, SIP_OUTGOING, sip_park(), sip_refer_allocate(), SIPBUFSIZE, sip_refer::status, TRANSFER_CLOSED, transmit_notify_with_sipfrag(), transmit_response(), TRUE, and ast_channel::uniqueid.

Referenced by handle_incoming().

{
   struct sip_dual current;   /* Chan1: Call between asterisk and transferer */
               /* Chan2: Call between asterisk and transferee */

   int res = 0;
   current.req.data = NULL;

   if (req->debug)
      ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n", p->callid, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller");

   if (!p->owner) {
      /* This is a REFER outside of an existing SIP dialog */
      /* We can't handle that, so decline it */
      ast_debug(3, "Call %s: Declined REFER, outside of dialog...\n", p->callid);
      transmit_response(p, "603 Declined (No dialog)", req);
      if (!req->ignore) {
         append_history(p, "Xfer", "Refer failed. Outside of dialog.");
         sip_alreadygone(p);
         p->needdestroy = 1;
      }
      return 0;
   }  


   /* Check if transfer is allowed from this device */
   if (p->allowtransfer == TRANSFER_CLOSED ) {
      /* Transfer not allowed, decline */
      transmit_response(p, "603 Declined (policy)", req);
      append_history(p, "Xfer", "Refer failed. Allowtransfer == closed.");
      /* Do not destroy SIP session */
      return 0;
   }

   if (!req->ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
      /* Already have a pending REFER */  
      transmit_response(p, "491 Request pending", req);
      append_history(p, "Xfer", "Refer failed. Request pending.");
      return 0;
   }

   /* Allocate memory for call transfer data */
   if (!p->refer && !sip_refer_allocate(p)) {
      transmit_response(p, "500 Internal Server Error", req);
      append_history(p, "Xfer", "Refer failed. Memory allocation error.");
      return -3;
   }

   res = get_refer_info(p, req); /* Extract headers */

   p->refer->status = REFER_SENT;

   if (res != 0) {
      switch (res) {
      case -2: /* Syntax error */
         transmit_response(p, "400 Bad Request (Refer-to missing)", req);
         append_history(p, "Xfer", "Refer failed. Refer-to missing.");
         if (req->debug)
            ast_debug(1, "SIP transfer to black hole can't be handled (no refer-to: )\n");
         break;
      case -3:
         transmit_response(p, "603 Declined (Non sip: uri)", req);
         append_history(p, "Xfer", "Refer failed. Non SIP uri");
         if (req->debug)
            ast_debug(1, "SIP transfer to non-SIP uri denied\n");
         break;
      default:
         /* Refer-to extension not found, fake a failed transfer */
         transmit_response(p, "202 Accepted", req);
         append_history(p, "Xfer", "Refer failed. Bad extension.");
         transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE);
         ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
         if (req->debug)
            ast_debug(1, "SIP transfer to bad extension: %s\n", p->refer->refer_to);
         break;
      } 
      return 0;
   }
   if (ast_strlen_zero(p->context))
      ast_string_field_set(p, context, default_context);

   /* If we do not support SIP domains, all transfers are local */
   if (allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
      p->refer->localtransfer = 1;
      if (sipdebug)
         ast_debug(3, "This SIP transfer is local : %s\n", p->refer->refer_to_domain);
   } else if (AST_LIST_EMPTY(&domain_list) || check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
      /* This PBX doesn't bother with SIP domains or domain is local, so this transfer is local */
      p->refer->localtransfer = 1;
   } else if (sipdebug)
         ast_debug(3, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain);
   
   /* Is this a repeat of a current request? Ignore it */
   /* Don't know what else to do right now. */
   if (req->ignore) 
      return res;

   /* If this is a blind transfer, we have the following
   channels to work with:
   - chan1, chan2: The current call between transferer and transferee (2 channels)
   - target_channel: A new call from the transferee to the target (1 channel)
   We need to stay tuned to what happens in order to be able
   to bring back the call to the transferer */

   /* If this is a attended transfer, we should have all call legs within reach:
   - chan1, chan2: The call between the transferer and transferee (2 channels)
   - target_channel, targetcall_pvt: The call between the transferer and the target (2 channels)
   We want to bridge chan2 with targetcall_pvt!
   
   The replaces call id in the refer message points
   to the call leg between Asterisk and the transferer.
   So we need to connect the target and the transferee channel
   and hangup the two other channels silently 
   
   If the target is non-local, the call ID could be on a remote
   machine and we need to send an INVITE with replaces to the
   target. We basically handle this as a blind transfer
   and let the sip_call function catch that we need replaces
   header in the INVITE.
   */


   /* Get the transferer's channel */
   current.chan1 = p->owner;

   /* Find the other part of the bridge (2) - transferee */
   current.chan2 = ast_bridged_channel(current.chan1);
   
   if (sipdebug)
      ast_debug(3, "SIP %s transfer: Transferer channel %s, transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind", current.chan1->name, current.chan2 ? current.chan2->name : "<none>");

   if (!current.chan2 && !p->refer->attendedtransfer) {
      /* No bridged channel, propably IVR or echo or similar... */
      /* Guess we should masquerade or something here */
      /* Until we figure it out, refuse transfer of such calls */
      if (sipdebug)
         ast_debug(3, "Refused SIP transfer on non-bridged channel.\n");
      p->refer->status = REFER_FAILED;
      append_history(p, "Xfer", "Refer failed. Non-bridged channel.");
      transmit_response(p, "603 Declined", req);
      return -1;
   }

   if (current.chan2) {
      if (sipdebug)
         ast_debug(4, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name);

      ast_queue_control(current.chan1, AST_CONTROL_UNHOLD);
   }

   ast_set_flag(&p->flags[0], SIP_GOTREFER); 

   /* Attended transfer: Find all call legs and bridge transferee with target*/
   if (p->refer->attendedtransfer) {
      if ((res = local_attended_transfer(p, &current, req, seqno)))
         return res; /* We're done with the transfer */
      /* Fall through for remote transfers that we did not find locally */
      if (sipdebug)
         ast_debug(4, "SIP attended transfer: Still not our call - generating INVITE with replaces\n");
      /* Fallthrough if we can't find the call leg internally */
   }


   /* Parking a call */
   if (p->refer->localtransfer && !strcmp(p->refer->refer_to, ast_parking_ext())) {
      /* Must release c's lock now, because it will not longer be accessible after the transfer! */
      *nounlock = 1;
      ast_channel_unlock(current.chan1);
      copy_request(&current.req, req);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
      p->refer->status = REFER_200OK;
      append_history(p, "Xfer", "REFER to call parking.");
      manager_event(EVENT_FLAG_CALL, "Transfer", "TransferMethod: SIP\r\nTransferType: Blind\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\nTransferExten: %s\r\nTransfer2Parking: Yes\r\n",
         current.chan1->name,
         current.chan1->uniqueid,
         p->callid,
         current.chan2->name,
         current.chan2->uniqueid,
         p->refer->refer_to);
      if (sipdebug)
         ast_debug(4, "SIP transfer to parking: trying to park %s. Parked by %s\n", current.chan2->name, current.chan1->name);
      sip_park(current.chan2, current.chan1, req, seqno);
      return res;
   } 

   /* Blind transfers and remote attended xfers */
   transmit_response(p, "202 Accepted", req);

   if (current.chan1 && current.chan2) {
      ast_debug(3, "chan1->name: %s\n", current.chan1->name);
      pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER", current.chan2->name);
   }
   if (current.chan2) {
      pbx_builtin_setvar_helper(current.chan2, "BLINDTRANSFER", current.chan1->name);
      pbx_builtin_setvar_helper(current.chan2, "SIPDOMAIN", p->refer->refer_to_domain);
      pbx_builtin_setvar_helper(current.chan2, "SIPTRANSFER", "yes");
      /* One for the new channel */
      pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER", "yes");
      /* Attended transfer to remote host, prepare headers for the INVITE */
      if (p->refer->referred_by) 
         pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", p->refer->referred_by);
   }
   /* Generate a Replaces string to be used in the INVITE during attended transfer */
   if (!ast_strlen_zero(p->refer->replaces_callid)) {
      char tempheader[SIPBUFSIZE];
      snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s", p->refer->replaces_callid, 
            p->refer->replaces_callid_totag ? ";to-tag=" : "", 
            p->refer->replaces_callid_totag, 
            p->refer->replaces_callid_fromtag ? ";from-tag=" : "",
            p->refer->replaces_callid_fromtag);
      if (current.chan2)
         pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REPLACES", tempheader);
   }
   /* Must release lock now, because it will not longer
      be accessible after the transfer! */
   *nounlock = 1;
   ast_channel_unlock(current.chan1);

   /* Connect the call */

   /* FAKE ringing if not attended transfer */
   if (!p->refer->attendedtransfer)
      transmit_notify_with_sipfrag(p, seqno, "183 Ringing", FALSE); 
      
   /* For blind transfer, this will lead to a new call */
   /* For attended transfer to remote host, this will lead to
      a new SIP call with a replaces header, if the dial plan allows it 
   */
   if (!current.chan2) {
      /* We have no bridge, so we're talking with Asterisk somehow */
      /* We need to masquerade this call */
      /* What to do to fix this situation:
         * Set up the new call in a new channel 
         * Let the new channel masq into this channel
         Please add that code here :-)
      */
      p->refer->status = REFER_FAILED;
      transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable (can't handle one-legged xfers)", TRUE);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
      append_history(p, "Xfer", "Refer failed (only bridged calls).");
      return -1;
   }
   ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);   /* Delay hangup */


   /* For blind transfers, move the call to the new extensions. For attended transfers on multiple
      servers - generate an INVITE with Replaces. Either way, let the dial plan decided  */
   res = ast_async_goto(current.chan2, p->refer->refer_to_context, p->refer->refer_to, 1);

   if (!res) {
      manager_event(EVENT_FLAG_CALL, "Transfer", "TransferMethod: SIP\r\nTransferType: Blind\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\nTransferExten: %s\r\nTransferContext: %s\r\n",
         current.chan1->name,
         current.chan1->uniqueid,
         p->callid,
         current.chan2->name,
         current.chan2->uniqueid,
         p->refer->refer_to, p->refer->refer_to_context);
      /* Success  - we have a new channel */
      ast_debug(3, "%s transfer succeeded. Telling transferer.\n", p->refer->attendedtransfer? "Attended" : "Blind");
      transmit_notify_with_sipfrag(p, seqno, "200 Ok", TRUE);
      if (p->refer->localtransfer)
         p->refer->status = REFER_200OK;
      if (p->owner)
         p->owner->hangupcause = AST_CAUSE_NORMAL_CLEARING;
      append_history(p, "Xfer", "Refer succeeded.");
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
      /* Do not hangup call, the other side do that when we say 200 OK */
      /* We could possibly implement a timer here, auto congestion */
      res = 0;
   } else {
      ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Don't delay hangup */
      ast_debug(3, "%s transfer failed. Resuming original call.\n", p->refer->attendedtransfer? "Attended" : "Blind");
      append_history(p, "Xfer", "Refer failed.");
      /* Failure of some kind */
      p->refer->status = REFER_FAILED;
      transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable", TRUE);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
      res = -1;
   }
   return res;
}

static int handle_request_register	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct sockaddr_in *	sin,
		char *	e
	)			[static]

Handle incoming REGISTER request.

Definition at line 19498 of file chan_sip.c.

References append_history, ast_debug, ast_inet_ntoa(), ast_log(), AUTH_ACL_FAILED, AUTH_BAD_TRANSPORT, AUTH_NOT_FOUND, AUTH_PEER_NOT_DYNAMIC, AUTH_SECRET_FAILED, AUTH_UNKNOWN_DOMAIN, AUTH_USERNAME_MISMATCH, sip_pvt::callid, check_via(), copy_request(), DEFAULT_TRANS_TIMEOUT, get_header(), sip_pvt::initreq, LOG_NOTICE, sip_request::method, register_verify(), sip_methods, and sip_scheddestroy().

Referenced by handle_incoming().

{
   enum check_auth_result res;

   /* Use this as the basis */
   copy_request(&p->initreq, req);
   if (sipdebug)
      ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   check_via(p, req);
   if ((res = register_verify(p, sin, req, e)) < 0) {
      const char *reason;

      switch (res) {
      case AUTH_SECRET_FAILED:
         reason = "Wrong password";
         break;
      case AUTH_USERNAME_MISMATCH:
         reason = "Username/auth name mismatch";
         break;
      case AUTH_NOT_FOUND:
         reason = "No matching peer found";
         break;
      case AUTH_UNKNOWN_DOMAIN:
         reason = "Not a local domain";
         break;
      case AUTH_PEER_NOT_DYNAMIC:
         reason = "Peer is not supposed to register";
         break;
      case AUTH_ACL_FAILED:
         reason = "Device does not match ACL";
         break;
      case AUTH_BAD_TRANSPORT:
         reason = "Device not configured to use this transport type";
         break;
      default:
         reason = "Unknown failure";
         break;
      }
      ast_log(LOG_NOTICE, "Registration from '%s' failed for '%s' - %s\n",
         get_header(req, "To"), ast_inet_ntoa(sin->sin_addr),
         reason);
      append_history(p, "RegRequest", "Failed : Account %s : %s", get_header(req, "To"), reason);
   } else
      append_history(p, "RegRequest", "Succeeded : Account %s", get_header(req, "To"));

   if (res < 1) {
      /* Destroy the session, but keep us around for just a bit in case they don't
         get our 200 OK */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   }
   return res;
}

static int handle_request_subscribe	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct sockaddr_in *	sin,
		int	seqno,
		char *	e
	)			[static]

Handle incoming SUBSCRIBE request.

Definition at line 19179 of file chan_sip.c.

References add_peer_mwi_subs(), ao2_iterator_init(), ao2_lock(), ao2_t_iterator_next, ao2_t_ref, ao2_unlock(), append_history, ast_debug, AST_EXTENSION_REMOVED, ast_extension_state(), ast_extension_state2str(), ast_extension_state_add(), ast_extension_state_del(), ast_inet_ntoa(), AST_LIST_EMPTY, ast_log(), ast_set_flag, ast_strdupa, ast_string_field_build, ast_strlen_zero(), ast_test_flag, ast_verbose, AUTH_CHALLENGE_SENT, AUTH_FAKE_AUTH, sip_pvt::autokillid, build_contact(), sip_pvt::callid, cb_extensionstate(), check_user_full(), check_via(), sip_pvt::context, copy_request(), CPIM_PIDF_XML, sip_request::debug, DIALOG_INFO_XML, dialog_ref(), dialog_unlink_all(), dialog_unref(), dialogs, sip_pvt::dialogver, sip_pvt::expiry, sip_pvt::exten, FALSE, sip_peer::flags, sip_pvt::flags, get_destination(), get_header(), gettag(), sip_request::headers, sip_request::ignore, sip_pvt::initreq, sip_pvt::lastinvite, sip_pvt::laststate, LOG_NOTICE, LOG_WARNING, sip_peer::mailboxes, make_our_tag(), sip_request::method, sip_peer::mwipvt, sip_peer::name, sip_pvt::needdestroy, NONE, parse_ok_contact(), PIDF_XML, ref_peer(), sip_pvt::relatedpeer, sip_pvt::sa, sip_cancel_destroy(), sip_methods, SIP_PAGE2_ALLOWSUBSCRIBE, SIP_PAGE2_DIALOG_ESTABLISHED, SIP_PAGE2_SUBSCRIBEMWIONLY, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), sip_send_mwi_to_peer(), SIP_SUBSCRIBE, sip_pvt::stateid, sip_pvt::subscribecontext, sip_pvt::subscribed, sip_pvt::subscribeuri, sip_pvt::tag, cfsip_methods::text, transmit_fake_auth_response(), transmit_response(), transmit_response_reliable(), transmit_state_notify(), TRUE, unref_peer(), sip_pvt::useragent, sip_pvt::username, XMIT_UNRELIABLE, and XPIDF_XML.

Referenced by handle_incoming().

{
   int gotdest = 0;
   int res = 0;
   int firststate = AST_EXTENSION_REMOVED;
   struct sip_peer *authpeer = NULL;
   const char *eventheader = get_header(req, "Event");   /* Get Event package name */
   const char *acceptheader = get_header(req, "Accept");
   int resubscribe = (p->subscribed != NONE);
   char *temp, *event;
   struct ao2_iterator i;

   if (p->initreq.headers) {  
      /* We already have a dialog */
      if (p->initreq.method != SIP_SUBSCRIBE) {
         /* This is a SUBSCRIBE within another SIP dialog, which we do not support */
         /* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */
         transmit_response(p, "403 Forbidden (within dialog)", req);
         /* Do not destroy session, since we will break the call if we do */
         ast_debug(1, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text);
         return 0;
      } else if (req->debug) {
         if (resubscribe)
            ast_debug(1, "Got a re-subscribe on existing subscription %s\n", p->callid);
         else
            ast_debug(1, "Got a new subscription %s (possibly with auth)\n", p->callid);
      }
   }

   /* Check if we have a global disallow setting on subscriptions. 
      if so, we don't have to check peer settings after auth, which saves a lot of processing
   */
   if (!global_allowsubscribe) {
      transmit_response(p, "403 Forbidden (policy)", req);
      p->needdestroy = 1;
      return 0;
   }

   if (!req->ignore && !resubscribe) { /* Set up dialog, new subscription */
      const char *to = get_header(req, "To");
      char totag[128];

      /* Check to see if a tag was provided, if so this is actually a resubscription of a dialog we no longer know about */
      if (!ast_strlen_zero(to) && gettag(req, "To", totag, sizeof(totag))) {
         if (req->debug)
            ast_verbose("Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.\n");
         transmit_response(p, "481 Subscription does not exist", req);
         p->needdestroy = 1;
         return 0;
      }

      /* Use this as the basis */
      if (req->debug)
         ast_verbose("Creating new subscription\n");

      copy_request(&p->initreq, req);
      if (sipdebug)
         ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
      check_via(p, req);
   } else if (req->debug && req->ignore)
      ast_verbose("Ignoring this SUBSCRIBE request\n");

   /* Find parameters to Event: header value and remove them for now */
   if (ast_strlen_zero(eventheader)) {
      transmit_response(p, "489 Bad Event", req);
      ast_debug(2, "Received SIP subscribe for unknown event package: <none>\n");
      p->needdestroy = 1;
      return 0;
   }

   if ( (strchr(eventheader, ';'))) {
      event = ast_strdupa(eventheader);   /* Since eventheader is a const, we can't change it */
      temp = strchr(event, ';');       
      *temp = '\0';           /* Remove any options for now */
                     /* We might need to use them later :-) */
   } else
      event = (char *) eventheader;    /* XXX is this legal ? */

   /* Handle authentication */
   res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, sin, &authpeer);
   /* if an authentication response was sent, we are done here */
   if (res == AUTH_CHALLENGE_SENT)  /* authpeer = NULL here */
      return 0;
   if (res < 0) {
      if (res == AUTH_FAKE_AUTH) {
         ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
         transmit_fake_auth_response(p, SIP_SUBSCRIBE, req, XMIT_UNRELIABLE);
      } else {
         ast_log(LOG_NOTICE, "Failed to authenticate device %s for SUBSCRIBE\n", get_header(req, "From"));
         transmit_response_reliable(p, "403 Forbidden", req);
      }
      p->needdestroy = 1;
      return 0;
   }

   /* At this point, authpeer cannot be NULL. Remember we hold a reference,
    * so we must release it when done.
    * XXX must remove all the checks for authpeer == NULL.
    */

   /* Check if this device  is allowed to subscribe at all */
   if (!ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
      transmit_response(p, "403 Forbidden (policy)", req);
      p->needdestroy = 1;
      if (authpeer)
         unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 1)");
      return 0;
   }

   if (strcmp(event, "message-summary")) {
      /* Get destination right away */
      gotdest = get_destination(p, NULL);
   }

   /* Get full contact header - this needs to be used as a request URI in NOTIFY's */
   parse_ok_contact(p, req);

   build_contact(p);
   if (gotdest) {
      transmit_response(p, "404 Not Found", req);
      p->needdestroy = 1;
      if (authpeer)
         unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 2)");
      return 0;
   }

   /* Initialize tag for new subscriptions */   
   if (ast_strlen_zero(p->tag))
      make_our_tag(p->tag, sizeof(p->tag));

   if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
      if (authpeer)  /* We do not need the authpeer any more */
         unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 2)");

      /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
      /* Polycom phones only handle xpidf+xml, even if they say they can
         handle pidf+xml as well
      */
      if (strstr(p->useragent, "Polycom")) {
         p->subscribed = XPIDF_XML;
      } else if (strstr(acceptheader, "application/pidf+xml")) {
         p->subscribed = PIDF_XML;         /* RFC 3863 format */
      } else if (strstr(acceptheader, "application/dialog-info+xml")) {
         p->subscribed = DIALOG_INFO_XML;
         /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
      } else if (strstr(acceptheader, "application/cpim-pidf+xml")) {
         p->subscribed = CPIM_PIDF_XML;    /* RFC 3863 format */
      } else if (strstr(acceptheader, "application/xpidf+xml")) {
         p->subscribed = XPIDF_XML;        /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */
      } else if (ast_strlen_zero(acceptheader)) {
         if (p->subscribed == NONE) { /* if the subscribed field is not already set, and there is no accept header... */
            transmit_response(p, "489 Bad Event", req);
  
            ast_log(LOG_WARNING, "SUBSCRIBE failure: no Accept header: pvt: stateid: %d, laststate: %d, dialogver: %d, subscribecont: '%s', subscribeuri: '%s'\n",
               p->stateid, p->laststate, p->dialogver, p->subscribecontext, p->subscribeuri);
            p->needdestroy = 1;
            return 0;
         }
         /* if p->subscribed is non-zero, then accept is not obligatory; according to rfc 3265 section 3.1.3, at least.
            so, we'll just let it ride, keeping the value from a previous subscription, and not abort the subscription */
      } else {
         /* Can't find a format for events that we know about */
         char mybuf[200];
         snprintf(mybuf, sizeof(mybuf), "489 Bad Event (format %s)", acceptheader);
         transmit_response(p, mybuf, req);
 
         ast_log(LOG_WARNING, "SUBSCRIBE failure: unrecognized format: '%s' pvt: subscribed: %d, stateid: %d, laststate: %d, dialogver: %d, subscribecont: '%s', subscribeuri: '%s'\n",
            acceptheader, (int)p->subscribed, p->stateid, p->laststate, p->dialogver, p->subscribecontext, p->subscribeuri);
         p->needdestroy = 1;
         return 0;
      }
   } else if (!strcmp(event, "message-summary")) { 
      if (!ast_strlen_zero(acceptheader) && strcmp(acceptheader, "application/simple-message-summary")) {
         /* Format requested that we do not support */
         transmit_response(p, "406 Not Acceptable", req);
         ast_debug(2, "Received SIP mailbox subscription for unknown format: %s\n", acceptheader);
         p->needdestroy = 1;
         if (authpeer)
            unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 3)");
         return 0;
      }
      /* Looks like they actually want a mailbox status 
        This version of Asterisk supports mailbox subscriptions
        The subscribed URI needs to exist in the dial plan
        In most devices, this is configurable to the voicemailmain extension you use
      */
      if (!authpeer || AST_LIST_EMPTY(&authpeer->mailboxes)) {
         transmit_response(p, "404 Not found (no mailbox)", req);
         p->needdestroy = 1;
         ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer->name);
         if (authpeer)
            unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 4)");
         return 0;
      }

      p->subscribed = MWI_NOTIFICATION;
      if (ast_test_flag(&authpeer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY)) {
         add_peer_mwi_subs(authpeer);
      }
      if (authpeer->mwipvt && authpeer->mwipvt != p) {   /* Destroy old PVT if this is a new one */
         /* We only allow one subscription per peer */
         dialog_unlink_all(authpeer->mwipvt, TRUE, TRUE);
         authpeer->mwipvt = dialog_unref(authpeer->mwipvt, "unref dialog authpeer->mwipvt");
         /* sip_destroy(authpeer->mwipvt); */
      }
      if (authpeer->mwipvt)
         dialog_unref(authpeer->mwipvt, "Unref previously stored mwipvt dialog pointer");
      authpeer->mwipvt = dialog_ref(p, "setting peers' mwipvt to p");      /* Link from peer to pvt UH- should this be dialog_ref()? */
      if (p->relatedpeer)
         unref_peer(p->relatedpeer, "Unref previously stored relatedpeer ptr");
      p->relatedpeer = ref_peer(authpeer, "setting dialog's relatedpeer pointer");  /* already refcounted...Link from pvt to peer UH- should this be dialog_ref()? */
      /* Do not release authpeer here */
   } else { /* At this point, Asterisk does not understand the specified event */
      transmit_response(p, "489 Bad Event", req);
      ast_debug(2, "Received SIP subscribe for unknown event package: %s\n", event);
      p->needdestroy = 1;
      if (authpeer)
         unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 5)");
      return 0;
   }

   /* Add subscription for extension state from the PBX core */
   if (p->subscribed != MWI_NOTIFICATION && !resubscribe) {
      if (p->stateid > -1) {
         ast_extension_state_del(p->stateid, cb_extensionstate);
         /* we need to dec the refcount, now that the extensionstate is removed */
         dialog_unref(p, "the extensionstate containing this dialog ptr was deleted");
      }
      p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, dialog_ref(p,"copying dialog ptr into extension state struct"));
   }

   if (!req->ignore && p)
      p->lastinvite = seqno;
   if (p && !p->needdestroy) {
      p->expiry = atoi(get_header(req, "Expires"));

      /* check if the requested expiry-time is within the approved limits from sip.conf */
      if (p->expiry > max_expiry)
         p->expiry = max_expiry;
      if (p->expiry < min_expiry && p->expiry > 0)
         p->expiry = min_expiry;

      if (sipdebug) {
         if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer)
            ast_debug(2, "Adding subscription for mailbox notification - peer %s\n", p->relatedpeer->name);
         else
            ast_debug(2, "Adding subscription for extension %s context %s for peer %s\n", p->exten, p->context, p->username);
      }
      if (p->autokillid > -1 && sip_cancel_destroy(p))   /* Remove subscription expiry for renewals */
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      if (p->expiry > 0)
         sip_scheddestroy(p, (p->expiry + 10) * 1000);   /* Set timer for destruction of call at expiration */

      if (p->subscribed == MWI_NOTIFICATION) {
         ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         transmit_response(p, "200 OK", req);
         if (p->relatedpeer) {   /* Send first notification */
            ao2_lock(p->relatedpeer); /* was WRLOCK */
            sip_send_mwi_to_peer(p->relatedpeer, NULL, 0);
            ao2_unlock(p->relatedpeer);
         }
      } else {
         struct sip_pvt *p_old;

         if ((firststate = ast_extension_state(NULL, p->context, p->exten)) < 0) {

            ast_log(LOG_NOTICE, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension.\n", p->exten, p->context, ast_inet_ntoa(p->sa.sin_addr));
            transmit_response(p, "404 Not found", req);
            p->needdestroy = 1;
            return 0;
         }
         ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         transmit_response(p, "200 OK", req);
         transmit_state_notify(p, firststate, 1, FALSE); /* Send first notification */
         append_history(p, "Subscribestatus", "%s", ast_extension_state2str(firststate));
         /* hide the 'complete' exten/context in the refer_to field for later display */
         ast_string_field_build(p, subscribeuri, "%s@%s", p->exten, p->context);

         /* remove any old subscription from this peer for the same exten/context,
         as the peer has obviously forgotten about it and it's wasteful to wait
         for it to expire and send NOTIFY messages to the peer only to have them
         ignored (or generate errors)
         */
         i = ao2_iterator_init(dialogs, 0);

         while ((p_old = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
            if (p_old == p) {
               ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
               continue;
            }
            if (p_old->initreq.method != SIP_SUBSCRIBE) {
               ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
               continue;
            }
            if (p_old->subscribed == NONE) {
               ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
               continue;
            }
            sip_pvt_lock(p_old);
            if (!strcmp(p_old->username, p->username)) {
               if (!strcmp(p_old->exten, p->exten) &&
                   !strcmp(p_old->context, p->context)) {
                  p_old->needdestroy = 1;
                  sip_pvt_unlock(p_old);
                  ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before break");
                  break;
               }
            }
            sip_pvt_unlock(p_old);
            ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next");
         }
      }
      if (!p->expiry)
         p->needdestroy = 1;
   }
   return 1;
}

static void handle_response	(	struct sip_pvt *	p,
		int	resp,
		char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Handle SIP response in dialogue.

Note:

only called by handle_incoming

Note:

SIP is incapable of performing a hairpin call, which is yet another failure of not having a layer 2 (again, YAY IETF for thinking ahead). So we treat this as a call forward and hope we end up at the right place...

Definition at line 16484 of file chan_sip.c.

References __sip_ack(), __sip_semi_ack(), AST_CAUSE_PROTOCOL_ERROR, ast_clear_flag, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_PROGRESS, ast_debug, ast_inet_ntoa(), ast_log(), ast_queue_control(), ast_queue_hangup_with_cause(), ast_set_flag, ast_skip_blanks(), ast_skip_nonblanks(), ast_strdupa, ast_string_field_build, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verb, ast_verbose, ast_channel::call_forward, cb_extensionstate(), sip_pvt::context, sip_request::debug, do_proxy_auth(), sip_pvt::exten, FALSE, find_sdp(), find_sip_method(), sip_pvt::flags, get_header(), gettag(), handle_response_invite(), handle_response_notify(), handle_response_peerpoke(), handle_response_refer(), handle_response_register(), hangup_sip2cause(), ast_channel::hangupcause, sip_request::ignore, sip_pvt::laststate, LOG_NOTICE, LOG_WARNING, MAX_AUTHTRIES, sip_pvt::method, msg, sip_pvt::needdestroy, sip_pvt::owner, parse_moved_contact(), sip_pvt::pendinginvite, process_sdp(), sip_pvt::refer, sip_pvt::relatedpeer, SDP_T38_NONE, SIP_ACK, sip_alreadygone(), SIP_BYE, SIP_CANCEL, sip_cancel_destroy(), SIP_INFO, SIP_INVITE, SIP_MESSAGE, sip_methods, SIP_NOTIFY, SIP_OPTIONS, SIP_OUTGOING, SIP_PAGE2_DIALOG_ESTABLISHED, SIP_PAGE2_STATECHANGEQUEUE, SIP_REFER, SIP_REGISTER, SIP_SUBSCRIBE, stop_media_flows(), sip_pvt::subscribed, text, sip_pvt::theirtag, transmit_request(), and XMIT_UNRELIABLE.

{
   struct ast_channel *owner;
   int sipmethod;
   int res = 1;
   const char *c = get_header(req, "Cseq");
   /* GCC 4.2 complains if I try to cast c as a char * when passing it to ast_skip_nonblanks, so make a copy of it */
   char *c_copy = ast_strdupa(c);
   /* Skip the Cseq and its subsequent spaces */
   const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));

   if (!msg)
      msg = "";

   sipmethod = find_sip_method(msg);

   owner = p->owner;
   if (owner) 
      owner->hangupcause = hangup_sip2cause(resp);

   /* Acknowledge whatever it is destined for */
   if ((resp >= 100) && (resp <= 199))
      __sip_semi_ack(p, seqno, 0, sipmethod);
   else
      __sip_ack(p, seqno, 0, sipmethod);

   /* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */
   if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite) 
      p->pendinginvite = 0;

   /* Get their tag if we haven't already */
   if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
      char tag[128];

      gettag(req, "To", tag, sizeof(tag));
      ast_string_field_set(p, theirtag, tag);
   }
   /* This needs to be configurable on a channel/peer level,
      not mandatory for all communication. Sadly enough, NAT implementations
      are not so stable so we can always rely on these headers. 
      Temporarily disabled, while waiting for fix.
      Fix assigned to Rizzo :-)
   */
   /* check_via_response(p, req); */

   /* RFC 3261 Section 15 specifies that if we receive a 408 or 481
    * in response to a BYE, then we should end the current dialog
    * and session.  It is known that at least one phone manufacturer
    * potentially will send a 404 in response to a BYE, so we'll be
    * liberal in what we accept and end the dialog and session if we
    * receive any of those responses to a BYE.
    */
   if ((resp == 404 || resp == 408 || resp == 481) && sipmethod == SIP_BYE) {
      p->needdestroy = 1;
      return;
   }

   if (p->relatedpeer && p->method == SIP_OPTIONS) {
      /* We don't really care what the response is, just that it replied back. 
         Well, as long as it's not a 100 response...  since we might
         need to hang around for something more "definitive" */
      if (resp != 100)
         handle_response_peerpoke(p, resp, req);
   } else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
      switch(resp) {
      case 100:   /* 100 Trying */
      case 101:   /* 101 Dialog establishment */
         if (sipmethod == SIP_INVITE) 
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 183:   /* 183 Session Progress */
         if (sipmethod == SIP_INVITE) 
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 180:   /* 180 Ringing */
         if (sipmethod == SIP_INVITE) 
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 182:       /* 182 Queued */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 200:   /* 200 OK */
         p->authtries = 0; /* Reset authentication counter */
         if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) {
            /* We successfully transmitted a message 
               or a video update request in INFO */
            /* Nothing happens here - the message is inside a dialog */
         } else if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_NOTIFY) {
            handle_response_notify(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_REGISTER) 
            res = handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_BYE) {    /* Ok, we're ready to go */
            p->needdestroy = 1;
            ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         } else if (sipmethod == SIP_SUBSCRIBE) {
            ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         }
         break;
      case 202:   /* Transfer accepted */
         if (sipmethod == SIP_REFER) 
            handle_response_refer(p, resp, rest, req, seqno);
         break;
      case 401: /* Not www-authorized on SIP method */
      case 407: /* Proxy auth required */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_NOTIFY)
            handle_response_notify(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_REFER)
            handle_response_refer(p, resp, rest, req, seqno);
         else if (p->registry && sipmethod == SIP_REGISTER)
            res = handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_BYE) {
            if (p->options)
               p->options->auth_type = resp;
            if (ast_strlen_zero(p->authname)) {
               ast_log(LOG_WARNING, "Asked to authenticate %s, to %s:%d but we have no matching peer!\n",
                     msg, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
               p->needdestroy = 1;
            } else if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp,  sipmethod, 0)) {
               ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From"));
               p->needdestroy = 1;
            }
         } else {
            ast_log(LOG_WARNING, "Got authentication request (%d) on %s to '%s'\n", resp, sip_methods[sipmethod].text, get_header(req, "To"));
            p->needdestroy = 1;
         }
         break;
      case 403: /* Forbidden - we failed authentication */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (p->registry && sipmethod == SIP_REGISTER) 
            res = handle_response_register(p, resp, rest, req, seqno);
         else {
            ast_log(LOG_WARNING, "Forbidden - maybe wrong password on authentication for %s\n", msg);
            p->needdestroy = 1;
         }
         break;
      case 404: /* Not found */
         if (p->registry && sipmethod == SIP_REGISTER)
            res = handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (owner)
            ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         break;
      case 423: /* Interval too brief */
         if (sipmethod == SIP_REGISTER)
            res = handle_response_register(p, resp, rest, req, seqno);
         break;
      case 408: /* Request timeout - terminate dialog */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_REGISTER) 
            res = handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_BYE) {
            p->needdestroy = 1;
            ast_debug(4, "Got timeout on bye. Thanks for the answer. Now, kill this call\n");
         } else {
            if (owner)
               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
            p->needdestroy = 1;
         }
         break;

      case 422: /* Session-Timers: Session Interval Too Small */
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         }
         break;

      case 481: /* Call leg does not exist */
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_REFER) {
            handle_response_refer(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {
            /* The other side has no transaction to bye,
            just assume it's all right then */
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
         } else if (sipmethod == SIP_CANCEL) {
            /* The other side has no transaction to cancel,
            just assume it's all right then */
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
         } else {
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
            /* Guessing that this is not an important request */
         }
         break;
      case 487:
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 488: /* Not acceptable here - codec error */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 491: /* Pending */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else {
            ast_debug(1, "Got 491 on %s, unspported. Call ID %s\n", sip_methods[sipmethod].text, p->callid);
            p->needdestroy = 1;
         }
         break;
      case 501: /* Not Implemented */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_REFER)
            handle_response_refer(p, resp, rest, req, seqno);
         else
            ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_inet_ntoa(p->sa.sin_addr), msg);
         break;
      case 603:   /* Declined transfer */
         if (sipmethod == SIP_REFER) {
            handle_response_refer(p, resp, rest, req, seqno);
            break;
         }
         /* Fallthrough */
      default:
         if ((resp >= 300) && (resp < 700)) {
            /* Fatal response */
            if ((resp != 487))
               ast_verb(3, "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
   
            if (sipmethod == SIP_INVITE)
               stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */

            /* XXX Locking issues?? XXX */
            switch(resp) {
            case 300: /* Multiple Choices */
            case 301: /* Moved permanently */
            case 302: /* Moved temporarily */
            case 305: /* Use Proxy */
               parse_moved_contact(p, req);
               /* Fall through */
            case 486: /* Busy here */
            case 600: /* Busy everywhere */
            case 603: /* Decline */
               if (p->owner)
                  ast_queue_control(p->owner, AST_CONTROL_BUSY);
               break;
            case 482: /*!
               \note SIP is incapable of performing a hairpin call, which
               is yet another failure of not having a layer 2 (again, YAY
                IETF for thinking ahead).  So we treat this as a call
                forward and hope we end up at the right place... */
               ast_debug(1, "Hairpin detected, setting up call forward for what it's worth\n");
               if (p->owner)
                  ast_string_field_build(p->owner, call_forward,
                               "Local/%s@%s", p->username, p->context);
               /* Fall through */
            case 480: /* Temporarily Unavailable */
            case 404: /* Not Found */
            case 410: /* Gone */
            case 400: /* Bad Request */
            case 500: /* Server error */
               if (sipmethod == SIP_REFER) {
                  handle_response_refer(p, resp, rest, req, seqno);
                  break;
               }
               /* Fall through */
            case 502: /* Bad gateway */
            case 503: /* Service Unavailable */
            case 504: /* Server Timeout */
               if (owner)
                  ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
               break;
            default:
               /* Send hangup */ 
               if (owner && sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO && sipmethod != SIP_BYE)
                  ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
               break;
            }
            /* ACK on invite */
            if (sipmethod == SIP_INVITE) 
               transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
            if (sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO) 
               sip_alreadygone(p);
            if (!p->owner)
               p->needdestroy = 1;
         } else if ((resp >= 100) && (resp < 200)) {
            if (sipmethod == SIP_INVITE) {
               if (!req->ignore && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
               if (find_sdp(req))
                  process_sdp(p, req, SDP_T38_NONE);
               if (p->owner) {
                  /* Queue a progress frame */
                  ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
               }
            }
         } else
            ast_log(LOG_NOTICE, "Dont know how to handle a %d %s response from %s\n", resp, rest, p->owner ? p->owner->name : ast_inet_ntoa(p->sa.sin_addr));
      }
   } else { 
      /* Responses to OUTGOING SIP requests on INCOMING calls 
         get handled here. As well as out-of-call message responses */
      if (req->debug)
         ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);

      if (sipmethod == SIP_INVITE && resp == 200) {
         /* Tags in early session is replaced by the tag in 200 OK, which is 
         the final reply to our INVITE */
         char tag[128];

         gettag(req, "To", tag, sizeof(tag));
         ast_string_field_set(p, theirtag, tag);
      }

      switch(resp) {
      case 200:
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_CANCEL) {
            ast_debug(1, "Got 200 OK on CANCEL\n");

            /* Wait for 487, then destroy */
         } else if (sipmethod == SIP_NOTIFY) {
            /* They got the notify, this is the end */
            if (p->owner) {
               if (p->refer) {
                  ast_debug(1, "Got 200 OK on NOTIFY for transfer\n");
               } else
                  ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
               /* ast_queue_hangup(p->owner); Disabled */
            } else {
               if (!p->subscribed && !p->refer)
                  p->needdestroy = 1;
               if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
                  /* Ready to send the next state we have on queue */
                  ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
                  cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
               }
            }
         } else if (sipmethod == SIP_BYE)
            p->needdestroy = 1;
         else if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO)
            /* We successfully transmitted a message or
               a video update request in INFO */
            ;
         else if (sipmethod == SIP_BYE) 
            /* Ok, we're ready to go */
            p->needdestroy = 1;
         break;
      case 202:   /* Transfer accepted */
         if (sipmethod == SIP_REFER) 
            handle_response_refer(p, resp, rest, req, seqno);
         break;
      case 401:   /* www-auth */
      case 407:
         if (sipmethod == SIP_REFER)
            handle_response_refer(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_INVITE) 
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_BYE) {
            if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, sipmethod, 0)) {
               ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From"));
               p->needdestroy = 1;
            }
         }
         break;
      case 481:   /* Call leg does not exist */
         if (sipmethod == SIP_INVITE) {
            /* Re-invite failed */
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {
            p->needdestroy = 1;
         } else if (sipdebug) {
            ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
         }
         break;
      case 501: /* Not Implemented */
         if (sipmethod == SIP_INVITE) 
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_REFER) 
            handle_response_refer(p, resp, rest, req, seqno);
         break;
      case 603:   /* Declined transfer */
         if (sipmethod == SIP_REFER) {
            handle_response_refer(p, resp, rest, req, seqno);
            break;
         }
         /* Fallthrough */
      default: /* Errors without handlers */
         if ((resp >= 100) && (resp < 200)) {
            if (sipmethod == SIP_INVITE) {   /* re-invite */
               if (!req->ignore && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
            }
         }
         if ((resp >= 300) && (resp < 700)) {
            if ((resp != 487))
               ast_verb(3, "Incoming call: Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
            switch(resp) {
            case 488: /* Not acceptable here - codec error */
            case 603: /* Decline */
            case 500: /* Server error */
            case 502: /* Bad gateway */
            case 503: /* Service Unavailable */
            case 504: /* Server timeout */

               /* re-invite failed */
               if (sipmethod == SIP_INVITE && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
               break;
            }
         }
         break;
      }
   }
}

static void handle_response_invite	(	struct sip_pvt *	p,
		int	resp,
		char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Handle SIP response to INVITE dialogue.

Definition at line 15860 of file chan_sip.c.

References ast_channel::_state, sip_pvt::alreadygone, append_history, AST_CAUSE_NORMAL_CLEARING, AST_CONTROL_ANSWER, AST_CONTROL_CONGESTION, AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, ast_debug, ast_log(), ast_null_frame, ast_queue_control(), ast_queue_frame(), ast_queue_hangup_with_cause(), ast_random(), ast_rtp_set_rtptimers_onhold(), ast_sched_add(), AST_SCHED_DEL_UNREF, ast_set_flag, ast_setstate(), AST_STATE_RINGING, AST_STATE_UP, ast_string_field_set, ast_strlen_zero(), ast_test_flag, sip_invite_param::auth_type, sip_pvt::authtries, build_route(), sip_pvt::callid, change_t38_state(), check_pendings(), DEC_CALL_LIMIT, DEC_CALL_RINGING, DEFAULT_TRANS_TIMEOUT, dialog_ref(), dialog_unref(), do_proxy_auth(), EVENT_FLAG_SYSTEM, FALSE, find_sdp(), sip_pvt::flags, sip_pvt::fullcontact, get_header(), sip_request::ignore, sip_pvt::initid, sip_pvt::initreq, INV_CALLING, INV_CANCELLED, INV_COMPLETED, INV_EARLY_MEDIA, INV_PROCEEDING, INV_TERMINATED, sip_pvt::invitestate, LOG_NOTICE, LOG_WARNING, manager_event, MAX_AUTHTRIES, ast_channel::name, sip_pvt::needdestroy, sip_pvt::options, sip_pvt::outgoing_call, sip_pvt::owner, parse_ok_contact(), parse_session_expires(), sip_pvt::peername, sip_pvt::pendinginvite, proc_422_rsp(), process_sdp(), sched, SDP_T38_ACCEPT, SDP_T38_NONE, SESSION_TIMER_MODE_ORIGINATE, SESSION_TIMER_MODE_REFUSE, SESSION_TIMER_REFRESHER_AUTO, SESSION_TIMER_REFRESHER_UAC, SESSION_TIMER_REFRESHER_UAS, set_address_from_contact(), SIP_ACK, sip_alreadygone(), sip_cancel_destroy(), SIP_INVITE, SIP_OUTGOING, SIP_PAGE2_DIALOG_ESTABLISHED, SIP_PENDINGBYE, sip_reinvite_retry(), sip_scheddestroy(), sip_st_dlg::st_active, sip_st_dlg::st_active_peer_ua, st_get_mode(), sip_st_dlg::st_interval, sip_st_dlg::st_ref, start_session_timer(), t38properties::state, sip_pvt::stimer, sip_pvt::t38, T38_DISABLED, T38_LOCAL_REINVITE, sip_pvt::theirtag, transmit_reinvite_with_sdp(), transmit_request(), TRUE, sip_pvt::udptl, ast_channel::uniqueid, update_call_counter(), sip_pvt::waitid, XMIT_ERROR, and XMIT_UNRELIABLE.

Referenced by handle_response().

{
   int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
   int res = 0;
   int xmitres = 0;
   int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
   char *p_hdrval;
   int rtn;

   if (reinvite)
      ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
   else
      ast_debug(4, "SIP response %d to standard invite\n", resp);

   if (p->alreadygone) { /* This call is already gone */
      ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
      return;
   }

   /* Acknowledge sequence number - This only happens on INVITE from SIP-call */
   /* Don't auto congest anymore since we've gotten something useful back */
   AST_SCHED_DEL_UNREF(sched, p->initid, dialog_unref(p, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));

   /* RFC3261 says we must treat every 1xx response (but not 100)
      that we don't recognize as if it was 183.
   */
   if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 182 && resp != 183)
      resp = 183;

   /* Any response between 100 and 199 is PROCEEDING */
   if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
      p->invitestate = INV_PROCEEDING;
 
   /* Final response, not 200 ? */
   if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
      p->invitestate = INV_COMPLETED;
   
   /* Final response, clear out pending invite */
   if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite)
      p->pendinginvite = 0;

   switch (resp) {
   case 100:   /* Trying */
   case 101:   /* Dialog establishment */
      if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      check_pendings(p);
      break;

   case 180:   /* 180 Ringing */
   case 182:       /* 182 Queued */
      if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      if (!req->ignore && p->owner) {
         ast_queue_control(p->owner, AST_CONTROL_RINGING);
         if (p->owner->_state != AST_STATE_UP) {
            ast_setstate(p->owner, AST_STATE_RINGING);
         }
      }
      if (find_sdp(req)) {
         if (p->invitestate != INV_CANCELLED)
            p->invitestate = INV_EARLY_MEDIA;
         res = process_sdp(p, req, SDP_T38_NONE);
         if (!req->ignore && p->owner) {
            /* Queue a progress frame only if we have SDP in 180 or 182 */
            ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
         }
      }
      check_pendings(p);
      break;

   case 183:   /* Session progress */
      if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      /* Ignore 183 Session progress without SDP */
      if (find_sdp(req)) {
         if (p->invitestate != INV_CANCELLED)
            p->invitestate = INV_EARLY_MEDIA;
         res = process_sdp(p, req, SDP_T38_NONE);
         if (!req->ignore && p->owner) {
            /* Queue a progress frame */
            ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
         }
      }
      check_pendings(p);
      break;

   case 200:   /* 200 OK on invite - someone's answering our call */
      if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      p->authtries = 0;
      if (find_sdp(req)) {
         if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore)
            if (!reinvite)
               /* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
               /* For re-invites, we try to recover */
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
      }

      /* Parse contact header for continued conversation */
      /* When we get 200 OK, we know which device (and IP) to contact for this call */
      /* This is important when we have a SIP proxy between us and the phone */
      if (outgoing) {
         update_call_counter(p, DEC_CALL_RINGING);
         parse_ok_contact(p, req);
         /* Save Record-Route for any later requests we make on this dialogue */
         if (!reinvite)
            build_route(p, req, 1);

         if(set_address_from_contact(p)) {
            /* Bad contact - we don't know how to reach this device */
            /* We need to ACK, but then send a bye */
            if (!p->route && !req->ignore)
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
         } 

      }

      if (!req->ignore && p->owner) {
         if (!reinvite) {
            ast_queue_control(p->owner, AST_CONTROL_ANSWER);
            if (global_callevents)
               manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
                  "Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
                  p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
         } else { /* RE-invite */
            ast_queue_frame(p->owner, &ast_null_frame);
         }
      } else {
          /* It's possible we're getting an 200 OK after we've tried to disconnect
              by sending CANCEL */
         /* First send ACK, then send bye */
         if (!req->ignore)
            ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
      }

      /* Check for Session-Timers related headers */
      if (st_get_mode(p) != SESSION_TIMER_MODE_REFUSE && p->outgoing_call == TRUE && !reinvite) {
         p_hdrval = (char*)get_header(req, "Session-Expires");
            if (!ast_strlen_zero(p_hdrval)) {
            /* UAS supports Session-Timers */
            enum st_refresher tmp_st_ref = SESSION_TIMER_REFRESHER_AUTO;
            int tmp_st_interval = 0;
            rtn = parse_session_expires(p_hdrval, &tmp_st_interval, &tmp_st_ref);
            if (rtn != 0) {
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
            }
            if (tmp_st_ref == SESSION_TIMER_REFRESHER_UAC || 
               tmp_st_ref == SESSION_TIMER_REFRESHER_UAS) {
               p->stimer->st_ref = tmp_st_ref;
            } 
            if (tmp_st_interval) {
               p->stimer->st_interval = tmp_st_interval;
            }
            p->stimer->st_active = TRUE;
            p->stimer->st_active_peer_ua = TRUE;
            start_session_timer(p);
         } else {
            /* UAS doesn't support Session-Timers */
            if (st_get_mode(p) == SESSION_TIMER_MODE_ORIGINATE) {
               p->stimer->st_ref = SESSION_TIMER_REFRESHER_UAC;
               p->stimer->st_active_peer_ua = FALSE;
               start_session_timer(p);
            }
         }
      }


      /* If I understand this right, the branch is different for a non-200 ACK only */
      p->invitestate = INV_TERMINATED;
      ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
      check_pendings(p);
      break;

   case 407: /* Proxy authentication */
   case 401: /* Www auth */
      /* First we ACK */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->options)
         p->options->auth_type = resp;

      /* Then we AUTH */
      ast_string_field_set(p, theirtag, NULL);  /* forget their old tag, so we don't match tags when getting response */
      if (!req->ignore) {
         if (p->authtries < MAX_AUTHTRIES)
            p->invitestate = INV_CALLING;
         if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
            ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
            p->needdestroy = 1;
            sip_alreadygone(p);
            if (p->owner)
               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         }
      }
      break;

   case 403: /* Forbidden */
      /* First we ACK */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", get_header(&p->initreq, "From"));
      if (!req->ignore && p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      p->needdestroy = 1;
      sip_alreadygone(p);
      break;

   case 404: /* Not found */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      sip_alreadygone(p);
      break;

   case 408: /* Request timeout */
   case 481: /* Call leg does not exist */
      /* Could be REFER caused INVITE with replaces */
      ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      break;

   case 422: /* Session-Timers: Session interval too small */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      ast_string_field_set(p, theirtag, NULL);
      proc_422_rsp(p, req);
      break;

   case 487: /* Cancelled transaction */
      /* We have sent CANCEL on an outbound INVITE 
         This transaction is already scheduled to be killed by sip_hangup().
      */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore) {
         ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_CLEARING);
         append_history(p, "Hangup", "Got 487 on CANCEL request from us. Queued AST hangup request");
      } else if (!req->ignore) {
         update_call_counter(p, DEC_CALL_LIMIT);
         append_history(p, "Hangup", "Got 487 on CANCEL request from us on call without owner. Killing this dialog.");
         p->needdestroy = 1;
         sip_alreadygone(p);
      }
      break;
   case 488: /* Not acceptable here */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
         change_t38_state(p, T38_DISABLED);
         /* Try to reset RTP timers */
         ast_rtp_set_rtptimers_onhold(p->rtp);

         /* Trigger a reinvite back to audio */
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      } else {
         /* We can't set up this call, so give up */
         if (p->owner && !req->ignore)
            ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         p->needdestroy = 1;
         /* If there's no dialog to end, then mark p as already gone */
         if (!reinvite)
            sip_alreadygone(p);
      }
      break;
   case 491: /* Pending */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore) {
         if (p->owner->_state != AST_STATE_UP) {
            ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
            p->needdestroy = 1;
         } else {
            /* This is a re-invite that failed. */
            /* Reset the flag after a while 
             */
            int wait;
            /* RFC 3261, if owner of call, wait between 2.1 to 4 seconds,
             * if not owner of call, wait 0 to 2 seconds */
            if (p->outgoing_call) {
               wait = 2100 + ast_random() % 2000;
            } else {
               wait = ast_random() % 2000;
            }
            p->waitid = ast_sched_add(sched, wait, sip_reinvite_retry, dialog_ref(p, "passing dialog ptr into sched structure based on waitid for sip_reinvite_retry."));
            ast_log(LOG_WARNING, "just did sched_add waitid(%d) for sip_reinvite_retry for dialog %s in handle_response_invite\n", p->waitid, p->callid);
            ast_debug(2, "Reinvite race. Waiting %d secs before retry\n", wait);
         }
      }
      break;

   case 501: /* Not implemented */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      break;
   }
   if (xmitres == XMIT_ERROR)
      ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
}

static void handle_response_notify	(	struct sip_pvt *	p,
		int	resp,
		char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Definition at line 16162 of file chan_sip.c.

References AST_CAUSE_NORMAL_UNSPECIFIED, ast_clear_flag, ast_debug, ast_inet_ntoa(), ast_log(), ast_queue_hangup_with_cause(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, sip_pvt::authname, sip_pvt::authtries, cb_extensionstate(), sip_pvt::context, do_proxy_auth(), sip_pvt::exten, sip_pvt::flags, get_header(), sip_pvt::initreq, sip_pvt::laststate, LOG_NOTICE, LOG_WARNING, ast_channel::name, sip_pvt::needdestroy, NONE, sip_pvt::notify_headers, sip_pvt::owner, sip_pvt::recv, sip_pvt::refer, SIP_NOTIFY, SIP_PAGE2_STATECHANGEQUEUE, sip_pvt::subscribed, and sip_pvt::theirtag.

Referenced by handle_response().

{
   switch (resp) {
   case 200:   /* Notify accepted */
      /* They got the notify, this is the end */
      if (p->owner) {
         if (!p->refer) {
            ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
            ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
         } else {
            ast_debug(4, "Got OK on REFER Notify message\n");
         }
      } else {
         if (p->subscribed == NONE) {
            ast_debug(4, "Got 200 accepted on NOTIFY\n");
            p->needdestroy = 1;
         }
         if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
            /* Ready to send the next state we have on queue */
            ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
            cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
         }
      }
      break;
   case 401:   /* Not www-authorized on SIP method */
   case 407:   /* Proxy auth */
      if (!p->notify_headers) {
         break; /* Only device notify can use NOTIFY auth */
      }
      ast_string_field_set(p, theirtag, NULL);
      if (ast_strlen_zero(p->authname)) {
         ast_log(LOG_WARNING, "Asked to authenticate NOTIFY to %s:%d but we have no matching peer or realm auth!\n", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
         p->needdestroy = 1;
      }
      if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_NOTIFY, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on NOTYFY to '%s'\n", get_header(&p->initreq, "From"));
         p->needdestroy = 1;
      }
      break;
   }
}

static void handle_response_peerpoke	(	struct sip_pvt *	p,
		int	resp,
		struct sip_request *	req
	)			[static]

Handle qualification responses (OPTIONS).

Definition at line 16414 of file chan_sip.c.

References ast_check_realtime(), AST_DEVICE_UNKNOWN, ast_devstate_changed(), ast_log(), AST_SCHED_REPLACE_UNREF, ast_tvdiff_ms(), ast_tvnow(), ast_update_realtime(), sip_peer::call, DEFAULT_FREQ_NOTOK, dialog_unref(), EVENT_FLAG_SYSTEM, sip_peer::lastms, LOG_NOTICE, manager_event, sip_peer::maxms, sip_peer::name, sip_pvt::needdestroy, sip_settings::peer_rtupdate, sip_peer::pokeexpire, sip_peer::ps, sip_peer::qualifyfreq, ref_peer(), register_peer_exten(), sip_pvt::relatedpeer, s, sched, SENTINEL, sip_cfg, sip_poke_peer_s(), TRUE, and unref_peer().

Referenced by handle_response().

{
   struct sip_peer *peer = /* ref_peer( */ p->relatedpeer /* , "bump refcount on p, as it is being used in this function(handle_response_peerpoke)")*/ ; /* hope this is already refcounted! */
   int statechanged, is_reachable, was_reachable;
   int pingtime = ast_tvdiff_ms(ast_tvnow(), peer->ps);

   /*
    * Compute the response time to a ping (goes in peer->lastms.)
    * -1 means did not respond, 0 means unknown,
    * 1..maxms is a valid response, >maxms means late response.
    */
   if (pingtime < 1) /* zero = unknown, so round up to 1 */
      pingtime = 1;

   /* Now determine new state and whether it has changed.
    * Use some helper variables to simplify the writing
    * of the expressions.
    */
   was_reachable = peer->lastms > 0 && peer->lastms <= peer->maxms;
   is_reachable = pingtime <= peer->maxms;
   statechanged = peer->lastms == 0 /* yes, unknown before */
      || was_reachable != is_reachable;

   peer->lastms = pingtime;
   peer->call = dialog_unref(peer->call, "unref dialog peer->call");
   if (statechanged) {
      const char *s = is_reachable ? "Reachable" : "Lagged";
      char str_lastms[20];
      snprintf(str_lastms, sizeof(str_lastms), "%d", pingtime);

      ast_log(LOG_NOTICE, "Peer '%s' is now %s. (%dms / %dms)\n",
         peer->name, s, pingtime, peer->maxms);
      ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
      if (sip_cfg.peer_rtupdate) {
         ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", str_lastms, SENTINEL);
      }
      manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
         "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: %s\r\nTime: %d\r\n",
         peer->name, s, pingtime);
      if (is_reachable && global_regextenonqualify)
         register_peer_exten(peer, TRUE);
   }

   p->needdestroy = 1;

   /* Try again eventually */
   AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
         is_reachable ? peer->qualifyfreq : DEFAULT_FREQ_NOTOK,
         sip_poke_peer_s, peer,
         unref_peer(_data, "removing poke peer ref"),
         unref_peer(peer, "removing poke peer ref"),
         ref_peer(peer, "adding poke peer ref"));
}

static void handle_response_refer	(	struct sip_pvt *	p,
		int	resp,
		char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Definition at line 16207 of file chan_sip.c.

References AST_CONTROL_CONGESTION, ast_debug, ast_inet_ntoa(), ast_log(), ast_queue_control(), ast_strlen_zero(), sip_pvt::authname, sip_pvt::authtries, sip_pvt::callid, do_proxy_auth(), get_header(), sip_pvt::initreq, LOG_NOTICE, LOG_WARNING, sip_pvt::needdestroy, sip_pvt::owner, sip_pvt::recv, sip_pvt::refer, REFER_ACCEPTED, REFER_FAILED, REFER_NOAUTH, sip_refer::refer_to, SIP_REFER, and sip_refer::status.

Referenced by handle_response().

{
   /* If no refer structure exists, then do nothing */
   if (!p->refer)
      return;

   switch (resp) {
   case 202:   /* Transfer accepted */
      /* We need  to do something here */
      /* The transferee is now sending INVITE to target */
      p->refer->status = REFER_ACCEPTED;
      /* Now wait for next message */
      ast_debug(3, "Got 202 accepted on transfer\n");
      /* We should hang along, waiting for NOTIFY's here */
      break;

   case 401:   /* Not www-authorized on SIP method */
   case 407:   /* Proxy auth */
      if (ast_strlen_zero(p->authname)) {
         ast_log(LOG_WARNING, "Asked to authenticate REFER to %s:%d but we have no matching peer or realm auth!\n",
            ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
         p->needdestroy = 1;
      }
      if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_REFER, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REFER to '%s'\n", get_header(&p->initreq, "From"));
         p->refer->status = REFER_NOAUTH;
         p->needdestroy = 1;
      }
      break;
   case 481: /* Call leg does not exist */

      /* A transfer with Replaces did not work */
      /* OEJ: We should Set flag, cancel the REFER, go back
      to original call - but right now we can't */
      ast_log(LOG_WARNING, "Remote host can't match REFER request to call '%s'. Giving up.\n", p->callid);
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      p->needdestroy = 1;
      break;

   case 500:   /* Server error */
   case 501:   /* Method not implemented */
      /* Return to the current call onhold */
      /* Status flag needed to be reset */
      ast_log(LOG_NOTICE, "SIP transfer to %s failed, call miserably fails. \n", p->refer->refer_to);
      p->needdestroy = 1;
      p->refer->status = REFER_FAILED;
      break;
   case 603:   /* Transfer declined */
      ast_log(LOG_NOTICE, "SIP transfer to %s declined, call miserably fails. \n", p->refer->refer_to);
      p->refer->status = REFER_FAILED;
      p->needdestroy = 1;
      break;
   }
}

static int handle_response_register	(	struct sip_pvt *	p,
		int	resp,
		char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Handle responses on REGISTER to services.

Definition at line 16264 of file chan_sip.c.

References __get_header(), ast_debug, ast_log(), AST_SCHED_DEL, AST_SCHED_REPLACE_UNREF, ast_strlen_zero(), ast_tvnow(), sip_pvt::authtries, sip_registry::call, sip_pvt::callid, DEFAULT_TRANS_TIMEOUT, dialog_unref(), do_register_auth(), EVENT_FLAG_SYSTEM, sip_registry::expire, sip_registry::expiry, EXPIRY_GUARD_LIMIT, EXPIRY_GUARD_MIN, EXPIRY_GUARD_PCT, EXPIRY_GUARD_SECS, get_header(), sip_registry::hostname, sip_pvt::initreq, LOG_NOTICE, LOG_WARNING, manager_event, MAX, MAX_AUTHTRIES, sip_pvt::needdestroy, sip_pvt::our_contact, sip_pvt::peername, sip_registry::refresh, REG_STATE_NOAUTH, REG_STATE_REGISTERED, REG_STATE_REJECTED, REG_STATE_UNREGISTERED, sip_registry::regattempts, sip_pvt::registry, registry_addref(), registry_unref(), sip_registry::regstate, regstate2str(), sip_registry::regtime, S_OR, sched, SIP_REGISTER, sip_reregister(), sip_scheddestroy(), strcasestr(), sip_registry::timeout, transmit_register(), sip_pvt::username, and sip_registry::username.

Referenced by handle_response().

{
   int expires, expires_ms;
   struct sip_registry *r;
   r=p->registry;
   
   switch (resp) {
   case 401:   /* Unauthorized */
      if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s@%s' (Tries %d)\n", p->registry->username, p->registry->hostname, p->authtries);
         p->needdestroy = 1;
      }
      break;
   case 403:   /* Forbidden */
      ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for REGISTER for '%s' to '%s'\n", p->registry->username, p->registry->hostname);
      AST_SCHED_DEL(sched, r->timeout);
      r->regstate = REG_STATE_NOAUTH;
      p->needdestroy = 1;
      break;
   case 404:   /* Not found */
      ast_log(LOG_WARNING, "Got 404 Not found on SIP register to service %s@%s, giving up\n", p->registry->username, p->registry->hostname);
      p->needdestroy = 1;
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 404");
      r->regstate = REG_STATE_REJECTED;
      AST_SCHED_DEL(sched, r->timeout);
      break;
   case 407:   /* Proxy auth */
      if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", get_header(&p->initreq, "From"), p->authtries);
         p->needdestroy = 1;
      }
      break;
   case 408:   /* Request timeout */
      /* Got a timeout response, so reset the counter of failed responses */
      if (r) {
         r->regattempts = 0;
      } else {
         ast_log(LOG_WARNING, "Got a 408 response to our REGISTER on call %s after we had destroyed the registry object\n", p->callid);
      }
      break;
   case 423:   /* Interval too brief */
      r->expiry = atoi(get_header(req, "Min-Expires"));
      ast_log(LOG_WARNING, "Got 423 Interval too brief for service %s@%s, minimum is %d seconds\n", p->registry->username, p->registry->hostname, r->expiry);
      AST_SCHED_DEL(sched, r->timeout);
      r->timeout = -1;
      if (r->call) {
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 423");
         p->needdestroy = 1;
      }
      if (r->expiry > max_expiry) {
         ast_log(LOG_WARNING, "Required expiration time from %s@%s is too high, giving up\n", p->registry->username, p->registry->hostname);
         r->expiry = default_expiry;
         r->regstate = REG_STATE_REJECTED;
      } else {
         r->regstate = REG_STATE_UNREGISTERED;
         transmit_register(r, SIP_REGISTER, NULL, NULL);
      }
      manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
      break;
   case 479:   /* SER: Not able to process the URI - address is wrong in register*/
      ast_log(LOG_WARNING, "Got error 479 on register to %s@%s, giving up (check config)\n", p->registry->username, p->registry->hostname);
      p->needdestroy = 1;
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 479");
      r->regstate = REG_STATE_REJECTED;
      AST_SCHED_DEL(sched, r->timeout);
      break;
   case 200:   /* 200 OK */
      if (!r) {
         ast_log(LOG_WARNING, "Got 200 OK on REGISTER, but there isn't a registry entry for '%s' (we probably already got the OK)\n", S_OR(p->peername, p->username));
         p->needdestroy = 1;
         return 0;
      }
      
      r->regstate = REG_STATE_REGISTERED;
      r->regtime = ast_tvnow();     /* Reset time of last succesful registration */
      manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate));
      r->regattempts = 0;
      ast_debug(1, "Registration successful\n");
      if (r->timeout > -1) {
         ast_debug(1, "Cancelling timeout %d\n", r->timeout);
      }
      AST_SCHED_DEL(sched, r->timeout);
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 200");
      p->registry = registry_unref(p->registry, "unref registry entry p->registry");
      /* Let this one hang around until we have all the responses */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      /* p->needdestroy = 1; */
      
      /* set us up for re-registering */
      /* figure out how long we got registered for */
      AST_SCHED_DEL(sched, r->expire);
      
      /* according to section 6.13 of RFC, contact headers override
         expires headers, so check those first */
      expires = 0;
      
      /* XXX todo: try to save the extra call */
      if (!ast_strlen_zero(get_header(req, "Contact"))) {
         const char *contact = NULL;
         const char *tmptmp = NULL;
         int start = 0;
         for(;;) {
            contact = __get_header(req, "Contact", &start);
            /* this loop ensures we get a contact header about our register request */
            if(!ast_strlen_zero(contact)) {
               if( (tmptmp=strstr(contact, p->our_contact))) {
                  contact=tmptmp;
                  break;
               }
            } else
               break;
         }
         tmptmp = strcasestr(contact, "expires=");
         if (tmptmp) {
            if (sscanf(tmptmp + 8, "%d;", &expires) != 1)
               expires = 0;
         }
         
      }
      if (!expires) 
         expires=atoi(get_header(req, "expires"));
      if (!expires)
         expires=default_expiry;
      
      expires_ms = expires * 1000;
      if (expires <= EXPIRY_GUARD_LIMIT)
         expires_ms -= MAX((expires_ms * EXPIRY_GUARD_PCT), EXPIRY_GUARD_MIN);
      else
         expires_ms -= EXPIRY_GUARD_SECS * 1000;
      if (sipdebug)
         ast_log(LOG_NOTICE, "Outbound Registration: Expiry for %s is %d sec (Scheduling reregistration in %d s)\n", r->hostname, expires, expires_ms/1000); 
      
      r->refresh= (int) expires_ms / 1000;
      
      /* Schedule re-registration before we expire */
      AST_SCHED_REPLACE_UNREF(r->expire, sched, expires_ms, sip_reregister, r, 
                        registry_unref(_data,"unref in REPLACE del fail"), 
                        registry_unref(r,"unref in REPLACE add fail"), 
                        registry_addref(r,"The Addition side of REPLACE")); 
      /* it is clear that we would not want to destroy the registry entry if we just
         scheduled a callback and recorded it in there! */
      /* since we never bumped the count, we shouldn't decrement it! registry_unref(r, "unref registry ptr r"); if this gets deleted, p->registry will be a bad pointer! */ 
   }
   return 1;
}

static const char * hangup_cause2sip

(

int

cause

)

[static]

Convert Asterisk hangup causes to SIP codes.

 Possible values from causes.h
        AST_CAUSE_NOTDEFINED    AST_CAUSE_NORMAL        AST_CAUSE_BUSY
        AST_CAUSE_FAILURE       AST_CAUSE_CONGESTION    AST_CAUSE_UNALLOCATED

	In addition to these, a lot of PRI codes is defined in causes.h 
	...should we take care of them too ?
	
	Quote RFC 3398

   ISUP Cause value                        SIP response
   ----------------                        ------------
   1  unallocated number                   404 Not Found
   2  no route to network                  404 Not found
   3  no route to destination              404 Not found
   16 normal call clearing                 --- (*)
   17 user busy                            486 Busy here
   18 no user responding                   408 Request Timeout
   19 no answer from the user              480 Temporarily unavailable
   20 subscriber absent                    480 Temporarily unavailable
   21 call rejected                        403 Forbidden (+)
   22 number changed (w/o diagnostic)      410 Gone
   22 number changed (w/ diagnostic)       301 Moved Permanently
   23 redirection to new destination       410 Gone
   26 non-selected user clearing           404 Not Found (=)
   27 destination out of order             502 Bad Gateway
   28 address incomplete                   484 Address incomplete
   29 facility rejected                    501 Not implemented
   31 normal unspecified                   480 Temporarily unavailable

Definition at line 5076 of file chan_sip.c.

References AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_CALL_REJECTED, AST_CAUSE_CHAN_NOT_IMPLEMENTED, AST_CAUSE_CONGESTION, AST_CAUSE_DESTINATION_OUT_OF_ORDER, AST_CAUSE_FACILITY_REJECTED, AST_CAUSE_FAILURE, AST_CAUSE_INVALID_NUMBER_FORMAT, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NO_ROUTE_TRANSIT_NET, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_NORMAL_UNSPECIFIED, AST_CAUSE_NOTDEFINED, AST_CAUSE_NUMBER_CHANGED, AST_CAUSE_SWITCH_CONGESTION, AST_CAUSE_UNALLOCATED, AST_CAUSE_UNREGISTERED, AST_CAUSE_USER_BUSY, and ast_debug.

Referenced by sip_hangup().

{
   switch (cause) {
      case AST_CAUSE_UNALLOCATED:      /* 1 */
      case AST_CAUSE_NO_ROUTE_DESTINATION:   /* 3 IAX2: Can't find extension in context */
      case AST_CAUSE_NO_ROUTE_TRANSIT_NET:   /* 2 */
         return "404 Not Found";
      case AST_CAUSE_CONGESTION:    /* 34 */
      case AST_CAUSE_SWITCH_CONGESTION:   /* 42 */
         return "503 Service Unavailable";
      case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
         return "408 Request Timeout";
      case AST_CAUSE_NO_ANSWER:     /* 19 */
      case AST_CAUSE_UNREGISTERED:        /* 20 */
         return "480 Temporarily unavailable";
      case AST_CAUSE_CALL_REJECTED:    /* 21 */
         return "403 Forbidden";
      case AST_CAUSE_NUMBER_CHANGED:      /* 22 */
         return "410 Gone";
      case AST_CAUSE_NORMAL_UNSPECIFIED:  /* 31 */
         return "480 Temporarily unavailable";
      case AST_CAUSE_INVALID_NUMBER_FORMAT:
         return "484 Address incomplete";
      case AST_CAUSE_USER_BUSY:
         return "486 Busy here";
      case AST_CAUSE_FAILURE:
         return "500 Server internal failure";
      case AST_CAUSE_FACILITY_REJECTED:   /* 29 */
         return "501 Not Implemented";
      case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
         return "503 Service Unavailable";
      /* Used in chan_iax2 */
      case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
         return "502 Bad Gateway";
      case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
         return "488 Not Acceptable Here";
         
      case AST_CAUSE_NOTDEFINED:
      default:
         ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause);
         return NULL;
   }

   /* Never reached */
   return 0;
}

static int hangup_sip2cause

(

int

cause

)

[static]

Convert SIP hangup causes to Asterisk hangup causes.

Definition at line 4964 of file chan_sip.c.

References AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_BUSY, AST_CAUSE_CALL_REJECTED, AST_CAUSE_CONGESTION, AST_CAUSE_DESTINATION_OUT_OF_ORDER, AST_CAUSE_FACILITY_REJECTED, AST_CAUSE_FAILURE, AST_CAUSE_INTERWORKING, AST_CAUSE_INVALID_NUMBER_FORMAT, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_NORMAL, AST_CAUSE_NORMAL_TEMPORARY_FAILURE, AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, AST_CAUSE_UNALLOCATED, and AST_CAUSE_USER_BUSY.

Referenced by handle_response().

{
   /* Possible values taken from causes.h */

   switch(cause) {
      case 401:   /* Unauthorized */
         return AST_CAUSE_CALL_REJECTED;
      case 403:   /* Not found */
         return AST_CAUSE_CALL_REJECTED;
      case 404:   /* Not found */
         return AST_CAUSE_UNALLOCATED;
      case 405:   /* Method not allowed */
         return AST_CAUSE_INTERWORKING;
      case 407:   /* Proxy authentication required */
         return AST_CAUSE_CALL_REJECTED;
      case 408:   /* No reaction */
         return AST_CAUSE_NO_USER_RESPONSE;
      case 409:   /* Conflict */
         return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
      case 410:   /* Gone */
         return AST_CAUSE_UNALLOCATED;
      case 411:   /* Length required */
         return AST_CAUSE_INTERWORKING;
      case 413:   /* Request entity too large */
         return AST_CAUSE_INTERWORKING;
      case 414:   /* Request URI too large */
         return AST_CAUSE_INTERWORKING;
      case 415:   /* Unsupported media type */
         return AST_CAUSE_INTERWORKING;
      case 420:   /* Bad extension */
         return AST_CAUSE_NO_ROUTE_DESTINATION;
      case 480:   /* No answer */
         return AST_CAUSE_NO_ANSWER;
      case 481:   /* No answer */
         return AST_CAUSE_INTERWORKING;
      case 482:   /* Loop detected */
         return AST_CAUSE_INTERWORKING;
      case 483:   /* Too many hops */
         return AST_CAUSE_NO_ANSWER;
      case 484:   /* Address incomplete */
         return AST_CAUSE_INVALID_NUMBER_FORMAT;
      case 485:   /* Ambiguous */
         return AST_CAUSE_UNALLOCATED;
      case 486:   /* Busy everywhere */
         return AST_CAUSE_BUSY;
      case 487:   /* Request terminated */
         return AST_CAUSE_INTERWORKING;
      case 488:   /* No codecs approved */
         return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
      case 491:   /* Request pending */
         return AST_CAUSE_INTERWORKING;
      case 493:   /* Undecipherable */
         return AST_CAUSE_INTERWORKING;
      case 500:   /* Server internal failure */
         return AST_CAUSE_FAILURE;
      case 501:   /* Call rejected */
         return AST_CAUSE_FACILITY_REJECTED;
      case 502:   
         return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
      case 503:   /* Service unavailable */
         return AST_CAUSE_CONGESTION;
      case 504:   /* Gateway timeout */
         return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
      case 505:   /* SIP version not supported */
         return AST_CAUSE_INTERWORKING;
      case 600:   /* Busy everywhere */
         return AST_CAUSE_USER_BUSY;
      case 603:   /* Decline */
         return AST_CAUSE_CALL_REJECTED;
      case 604:   /* Does not exist anywhere */
         return AST_CAUSE_UNALLOCATED;
      case 606:   /* Not acceptable */
         return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
      default:
         return AST_CAUSE_NORMAL;
   }
   /* Never reached */
   return 0;
}

static int init_req	(	struct sip_request *	req,
		int	sipmethod,
		const char *	recip
	)			[static]

Initialize SIP request.

Definition at line 7889 of file chan_sip.c.

References ast_str_create(), ast_str_set(), sip_methods, SIP_MIN_PACKET, and cfsip_methods::text.

{
   /* Initialize a request */
   memset(req, 0, sizeof(*req));
   if (!(req->data = ast_str_create(SIP_MIN_PACKET)))
      return -1;
   req->method = sipmethod;
   req->header[0] = 0;
   ast_str_set(&req->data, 0, "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip);
   req->len = req->data->used;
   req->headers++;
   return 0;
}

static int init_resp	(	struct sip_request *	resp,
		const char *	msg
	)			[static]

Initialize SIP response, based on SIP request.

Definition at line 7874 of file chan_sip.c.

References ast_str_create(), ast_str_set(), SIP_MIN_PACKET, and SIP_RESPONSE.

{
   /* Initialize a response */
   memset(resp, 0, sizeof(*resp));
   resp->method = SIP_RESPONSE;
   if (!(resp->data = ast_str_create(SIP_MIN_PACKET)))
      return -1;
   resp->header[0] = 0;
   ast_str_set(&resp->data, 0, "SIP/2.0 %s\r\n", msg);
   resp->len = resp->data->used;
   resp->headers++;
   return 0;
}

static void initialize_initreq	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Initialize the initital request packet in the pvt structure. This packet is used for creating replies and future requests in a dialog.

Definition at line 2629 of file chan_sip.c.

References ast_debug, ast_verbose, sip_pvt::callid, copy_request(), sip_request::debug, sip_request::headers, sip_pvt::initreq, sip_request::lines, sip_request::method, parse_request(), and sip_methods.

Referenced by transmit_invite(), transmit_notify_custom(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), transmit_register(), and transmit_reinvite_with_sdp().

{
   if (p->initreq.headers)
      ast_debug(1, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
   else
      ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   /* Use this as the basis */
   copy_request(&p->initreq, req);
   parse_request(&p->initreq);
   if (req->debug)
      ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}

static void initreqprep	(	struct sip_request *	req,
		struct sip_pvt *	p,
		int	sipmethod
	)			[static]

Initiate new SIP request to peer/user.

Todo:

Need to add back the VXML URL here at some point, possibly use build_string for all this junk

Definition at line 9234 of file chan_sip.c.

References add_header(), AST_DIGIT_ANYNUM, ast_inet_ntoa(), AST_PRES_ALLOWED, AST_PRES_RESTRICTION, ast_str_alloca, ast_str_append(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_encode(), build_contact(), build_rpid(), CALLERID_UNKNOWN, sip_pvt::callid, sip_pvt::callingpres, ast_channel::cid, ast_callerid::cid_name, ast_callerid::cid_num, DEFAULT_MAX_FORWARDS, exten, sip_pvt::flags, sip_pvt::fromdomain, sip_pvt::fromname, sip_pvt::fromuser, sip_pvt::fullcontact, init_req(), sip_pvt::lastmsg, sip_pvt::ocseq, sip_pvt::options, sip_pvt::our_contact, sip_pvt::ourip, ourport, sip_pvt::owner, sip_pvt::rpid, s, S_OR, sip_pvt::sa, SIP_INVITE, sip_methods, SIP_NOTIFY, SIP_SENDRPID, sip_standard_port(), SIP_USEREQPHONE, SIPBUFSIZE, sip_pvt::socket, STANDARD_SIP_PORT, ast_str::str, sip_pvt::tag, cfsip_methods::text, sip_pvt::theirtag, sip_pvt::todnid, sip_pvt::tohost, sip_socket::type, sip_pvt::uri, sip_invite_param::uri_options, sip_pvt::username, sip_pvt::via, and sip_invite_param::vxml_url.

Referenced by transmit_invite(), transmit_notify_custom(), and transmit_notify_with_mwi().

{
   struct ast_str *invite = ast_str_alloca(256);
   char from[256];
   char to[256];
   char tmp_n[SIPBUFSIZE/2];  /* build a local copy of 'n' if needed */
   char tmp_l[SIPBUFSIZE/2];  /* build a local copy of 'l' if needed */
   const char *l = NULL;   /* XXX what is this, exactly ? */
   const char *n = NULL;   /* XXX what is this, exactly ? */
   const char *urioptions = "";
   int ourport;

   if (ast_test_flag(&p->flags[0], SIP_USEREQPHONE)) {
      const char *s = p->username;  /* being a string field, cannot be NULL */

      /* Test p->username against allowed characters in AST_DIGIT_ANY
         If it matches the allowed characters list, then sipuser = ";user=phone"
         If not, then sipuser = ""
      */
      /* + is allowed in first position in a tel: uri */
      if (*s == '+')
         s++;
      for (; *s; s++) {
         if (!strchr(AST_DIGIT_ANYNUM, *s) )
            break;
      }
      /* If we have only digits, add ;user=phone to the uri */
      if (!*s)
         urioptions = ";user=phone";
   }


   snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text);

   if (p->owner) {
      l = p->owner->cid.cid_num;
      n = p->owner->cid.cid_name;
   }
   /* if we are not sending RPID and user wants his callerid restricted */
   if (!ast_test_flag(&p->flags[0], SIP_SENDRPID) &&
       ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED)) {
      l = CALLERID_UNKNOWN;
      n = l;
   }
   if (ast_strlen_zero(l))
      l = default_callerid;
   if (ast_strlen_zero(n))
      n = l;
   /* Allow user to be overridden */
   if (!ast_strlen_zero(p->fromuser))
      l = p->fromuser;
   else /* Save for any further attempts */
      ast_string_field_set(p, fromuser, l);

   /* Allow user to be overridden */
   if (!ast_strlen_zero(p->fromname))
      n = p->fromname;
   else /* Save for any further attempts */
      ast_string_field_set(p, fromname, n);

   if (pedanticsipchecking) {
      ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
      n = tmp_n;
      ast_uri_encode(l, tmp_l, sizeof(tmp_l), 0);
      l = tmp_l;
   }

   ourport = ntohs(p->ourip.sin_port);
   if (!sip_standard_port(p->socket.type, ourport) && ast_strlen_zero(p->fromdomain))
      snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, ast_inet_ntoa(p->ourip.sin_addr), ourport, p->tag);
   else
      snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr)), p->tag);

   /* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */
   if (!ast_strlen_zero(p->fullcontact)) {
      /* If we have full contact, trust it */
      ast_str_append(&invite, 0, "%s", p->fullcontact);
   } else {
      /* Otherwise, use the username while waiting for registration */
      ast_str_append(&invite, 0, "sip:");
      if (!ast_strlen_zero(p->username)) {
         n = p->username;
         if (pedanticsipchecking) {
            ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
            n = tmp_n;
         }
         ast_str_append(&invite, 0, "%s@", n);
      }
      ast_str_append(&invite, 0, "%s", p->tohost);
      if (ntohs(p->sa.sin_port) != STANDARD_SIP_PORT)
         ast_str_append(&invite, 0, ":%d", ntohs(p->sa.sin_port));
      ast_str_append(&invite, 0, "%s", urioptions);
   }

   /* If custom URI options have been provided, append them */
   if (p->options && !ast_strlen_zero(p->options->uri_options))
      ast_str_append(&invite, 0, ";%s", p->options->uri_options);
   
   /* This is the request URI, which is the next hop of the call
      which may or may not be the destination of the call
   */
   ast_string_field_set(p, uri, invite->str);
  
   if (!ast_strlen_zero(p->todnid)) {
      /*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
      if (!strchr(p->todnid, '@')) {
         /* We have no domain in the dnid */
         snprintf(to, sizeof(to), "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
      } else {
         snprintf(to, sizeof(to), "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
      }
   } else {
      if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) { 
         /* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
         snprintf(to, sizeof(to), "<%s%s>;tag=%s", (!strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
      } else if (p->options && p->options->vxml_url) {
         /* If there is a VXML URL append it to the SIP URL */
         snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
      } else 
         snprintf(to, sizeof(to), "<%s>", p->uri);
   }

   init_req(req, sipmethod, p->uri);
   /* now tmp_n is available so reuse it to build the CSeq */
   snprintf(tmp_n, sizeof(tmp_n), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);

   add_header(req, "Via", p->via);
   add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
   /* SLD: FIXME?: do Route: here too?  I think not cos this is the first request.
    * OTOH, then we won't have anything in p->route anyway */

   /* Build Remote Party-ID and From */
   if (ast_test_flag(&p->flags[0], SIP_SENDRPID) && (sipmethod == SIP_INVITE)) {
      build_rpid(p);
      add_header(req, "From", p->rpid_from);
   } else 
      add_header(req, "From", from);
   add_header(req, "To", to);
   ast_string_field_set(p, exten, l);
   build_contact(p);
   add_header(req, "Contact", p->our_contact);
   add_header(req, "Call-ID", p->callid);
   add_header(req, "CSeq", tmp_n);
   if (!ast_strlen_zero(global_useragent))
      add_header(req, "User-Agent", global_useragent);
   if (!ast_strlen_zero(p->rpid))
      add_header(req, "Remote-Party-ID", p->rpid);
}

static const char * insecure2str

(

int

mode

)

const [static]

Convert Insecure setting to printable string.

Definition at line 13201 of file chan_sip.c.

References insecurestr, and map_x_s().

Referenced by _sip_show_peer().

{
   return map_x_s(insecurestr, mode, "<error>");
}

static void list_route

(

struct sip_route *

route

)

[static]

List all routes - mostly for debugging.

Definition at line 10769 of file chan_sip.c.

References ast_verbose, sip_route::hop, and sip_route::next.

Referenced by build_route().

{
   if (!route)
      ast_verbose("list_route: no route\n");
   else {
      for (;route; route = route->next)
         ast_verbose("list_route: hop: <%s>\n", route->hop);
   }
}

static int load_module

(

void

)

[static]

PBX load module - initialization.

Todo:

Fix this XXX This must be all wrong XXXX

Definition at line 23386 of file chan_sip.c.

References ao2_t_container_alloc, ast_channel_register(), ast_check_realtime(), ast_cli_register_multiple(), ast_custom_function_register, ast_log(), ast_manager_register2(), AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_SUCCESS, ast_realtime_require_field(), ast_register_application, ast_rtp_proto_register(), ast_udptl_proto_register(), ast_verbose, ASTOBJ_CONTAINER_INIT, CHANNEL_MODULE_LOAD, checksipdomain_function, cli_sip, dialog_cmp_cb(), dialog_hash_cb(), dialogs, EVENT_FLAG_REPORTING, EVENT_FLAG_SYSTEM, hash_dialog_size, hash_peer_size, io, io_context_create(), io_context_destroy(), LOG_ERROR, manager_show_registry(), manager_sip_qualify_peer(), manager_sip_show_peer(), manager_sip_show_peers(), manager_sipnotify(), mandescr_show_peer, mandescr_show_peers, mandescr_show_registry, mandescr_sipnotify, peer_cmp_cb(), peer_hash_cb(), peer_ipcmp_cb(), peer_iphash_cb(), peers_by_ip, regl, reload_config(), restart_monitor(), RQ_CHAR, RQ_INTEGER4, RQ_UINTEGER2, sched, sched_context_create(), sched_context_destroy(), ast_channel_tech::send_digit_begin, SENTINEL, sip_addheader(), sip_dtmfmode(), sip_header_function, sip_poke_all_peers(), sip_rtp, sip_send_all_registers(), sip_tech, sip_tech_info, sip_udptl, sipchaninfo_function, and sippeer_function.

{
   ast_verbose("SIP channel loading...\n");
   /* the fact that ao2_containers can't resize automatically is a major worry! */
   /* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */
   peers = ao2_t_container_alloc(hash_peer_size, peer_hash_cb, peer_cmp_cb, "allocate peers");
   peers_by_ip = ao2_t_container_alloc(hash_peer_size, peer_iphash_cb, peer_ipcmp_cb, "allocate peers_by_ip");
   dialogs = ao2_t_container_alloc(hash_dialog_size, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs");
   
   ASTOBJ_CONTAINER_INIT(&regl); /* Registry object list -- not searched for anything */

   if (!(sched = sched_context_create())) {
      ast_log(LOG_ERROR, "Unable to create scheduler context\n");
      return AST_MODULE_LOAD_FAILURE;
   }

   if (!(io = io_context_create())) {
      ast_log(LOG_ERROR, "Unable to create I/O context\n");
      sched_context_destroy(sched);
      return AST_MODULE_LOAD_FAILURE;
   }

   sip_reloadreason = CHANNEL_MODULE_LOAD;

   if(reload_config(sip_reloadreason)) /* Load the configuration from sip.conf */
      return AST_MODULE_LOAD_DECLINE;

   /* Prepare the version that does not require DTMF BEGIN frames.
    * We need to use tricks such as memcpy and casts because the variable
    * has const fields.
    */
   memcpy(&sip_tech_info, &sip_tech, sizeof(sip_tech));
   memset((void *) &sip_tech_info.send_digit_begin, 0, sizeof(sip_tech_info.send_digit_begin));

   /* Make sure we can register our sip channel type */
   if (ast_channel_register(&sip_tech)) {
      ast_log(LOG_ERROR, "Unable to register channel type 'SIP'\n");
      io_context_destroy(io);
      sched_context_destroy(sched);
      return AST_MODULE_LOAD_FAILURE;
   }

   /* Register all CLI functions for SIP */
   ast_cli_register_multiple(cli_sip, sizeof(cli_sip)/ sizeof(struct ast_cli_entry));

   /* Tell the RTP subdriver that we're here */
   ast_rtp_proto_register(&sip_rtp);

   /* Tell the UDPTL subdriver that we're here */
   ast_udptl_proto_register(&sip_udptl);

   /* Register dialplan applications */
   ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode);
   ast_register_application(app_sipaddheader, sip_addheader, synopsis_sipaddheader, descrip_sipaddheader);

   /* Register dialplan functions */
   ast_custom_function_register(&sip_header_function);
   ast_custom_function_register(&sippeer_function);
   ast_custom_function_register(&sipchaninfo_function);
   ast_custom_function_register(&checksipdomain_function);

   /* Register manager commands */
   ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peers,
         "List SIP peers (text format)", mandescr_show_peers);
   ast_manager_register2("SIPshowpeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peer,
         "Show SIP peer (text format)", mandescr_show_peer);
   ast_manager_register2("SIPqualifypeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_qualify_peer,
         "Show SIP peer (text format)", mandescr_show_peer);   /*! \todo Fix this XXX This must be all wrong XXXX */
   ast_manager_register2("SIPshowregistry", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_show_registry,
         "Show SIP registrations (text format)", mandescr_show_registry);
   ast_manager_register2("SIPnotify", EVENT_FLAG_SYSTEM, manager_sipnotify,
         "Send a SIP notify", mandescr_sipnotify);
   sip_poke_all_peers();   
   sip_send_all_registers();
   
   /* And start the monitor for the first time */
   restart_monitor();

   ast_realtime_require_field(ast_check_realtime("sipregs") ? "sipregs" : "sippeers",
      "name", RQ_CHAR, 10,
      "ipaddr", RQ_CHAR, 15,
      "port", RQ_UINTEGER2, 5,
      "regseconds", RQ_INTEGER4, 11,
      "defaultuser", RQ_CHAR, 10,
      "fullcontact", RQ_CHAR, 35,
      "regserver", RQ_CHAR, 20,
      "useragent", RQ_CHAR, 20,
      "lastms", RQ_INTEGER4, 11,
      SENTINEL);

   return AST_MODULE_LOAD_SUCCESS;
}

static int local_attended_transfer	(	struct sip_pvt *	transferer,
		struct sip_dual *	current,
		struct sip_request *	req,
		int	seqno
	)			[static]

Find all call legs and bridge transferee with target called from handle_request_refer.

Definition at line 18451 of file chan_sip.c.

References ast_channel::_state, ao2_t_ref, append_history, ast_bridged_channel(), ast_channel_unlock, ast_clear_flag, ast_debug, ast_set_flag, ast_state2str(), AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, attempt_transfer(), sip_pvt::callid, sip_dual::chan1, sip_dual::chan2, EVENT_FLAG_CALL, sip_pvt::flags, get_sip_pvt_byid_locked(), sip_refer::localtransfer, manager_event, ast_channel::name, sip_pvt::owner, sip_pvt::refer, REFER_200OK, REFER_FAILED, sip_refer::replaces_callid, sip_refer::replaces_callid_fromtag, sip_refer::replaces_callid_totag, SIP_DEFER_BYE_ON_TRANSFER, SIP_GOTREFER, sip_pvt_unlock, sip_refer::status, transmit_notify_with_sipfrag(), transmit_response(), TRUE, and ast_channel::uniqueid.

Referenced by handle_request_refer().

{
   struct sip_dual target;    /* Chan 1: Call from tranferer to Asterisk */
               /* Chan 2: Call from Asterisk to target */
   int res = 0;
   struct sip_pvt *targetcall_pvt;

   /* Check if the call ID of the replaces header does exist locally */
   if (!(targetcall_pvt = get_sip_pvt_byid_locked(transferer->refer->replaces_callid, transferer->refer->replaces_callid_totag, 
      transferer->refer->replaces_callid_fromtag))) {
      if (transferer->refer->localtransfer) {
         /* We did not find the refered call. Sorry, can't accept then */
         transmit_response(transferer, "202 Accepted", req);
         /* Let's fake a response from someone else in order
            to follow the standard */
         transmit_notify_with_sipfrag(transferer, seqno, "481 Call leg/transaction does not exist", TRUE);
         append_history(transferer, "Xfer", "Refer failed");
         ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);  
         transferer->refer->status = REFER_FAILED;
         return -1;
      }
      /* Fall through for remote transfers that we did not find locally */
      ast_debug(3, "SIP attended transfer: Not our call - generating INVITE with replaces\n");
      return 0;
   }

   /* Ok, we can accept this transfer */
   transmit_response(transferer, "202 Accepted", req);
   append_history(transferer, "Xfer", "Refer accepted");
   if (!targetcall_pvt->owner) { /* No active channel */
      ast_debug(4, "SIP attended transfer: Error: No owner of target call\n");
      /* Cancel transfer */
      transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
      append_history(transferer, "Xfer", "Refer failed");
      ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
      transferer->refer->status = REFER_FAILED;
      sip_pvt_unlock(targetcall_pvt);
      ast_channel_unlock(current->chan1);
      if (targetcall_pvt)
         ao2_t_ref(targetcall_pvt, -1, "Drop targetcall_pvt pointer");
      return -1;
   }

   /* We have a channel, find the bridge */
   target.chan1 = targetcall_pvt->owner;           /* Transferer to Asterisk */
   target.chan2 = ast_bridged_channel(targetcall_pvt->owner);  /* Asterisk to target */

   if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP || target.chan2->_state == AST_STATE_RINGING) ) {
      /* Wrong state of new channel */
      if (target.chan2) 
         ast_debug(4, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state));
      else if (target.chan1->_state != AST_STATE_RING)
         ast_debug(4, "SIP attended transfer: Error: No target channel\n");
      else
         ast_debug(4, "SIP attended transfer: Attempting transfer in ringing state\n");
   }

   /* Transfer */
   if (sipdebug) {
      if (current->chan2)  /* We have two bridges */
         ast_debug(4, "SIP attended transfer: trying to bridge %s and %s\n", target.chan1->name, current->chan2->name);
      else        /* One bridge, propably transfer of IVR/voicemail etc */
         ast_debug(4, "SIP attended transfer: trying to make %s take over (masq) %s\n", target.chan1->name, current->chan1->name);
   }

   ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);   /* Delay hangup */

   /* Perform the transfer */
   manager_event(EVENT_FLAG_CALL, "Transfer", "TransferMethod: SIP\r\nTransferType: Attended\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\n",
      transferer->owner->name,
      transferer->owner->uniqueid,
      transferer->callid,
      target.chan1->name,
      target.chan1->uniqueid);
   res = attempt_transfer(current, &target);
   sip_pvt_unlock(targetcall_pvt);
   if (res) {
      /* Failed transfer */
      transmit_notify_with_sipfrag(transferer, seqno, "486 Busy Here", TRUE);
      append_history(transferer, "Xfer", "Refer failed");
      if (targetcall_pvt->owner)
         ast_channel_unlock(targetcall_pvt->owner);
      ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
   } else {
      /* Transfer succeeded! */

      /* Tell transferer that we're done. */
      transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE);
      append_history(transferer, "Xfer", "Refer succeeded");
      transferer->refer->status = REFER_200OK;
      if (targetcall_pvt->owner) {
         ast_debug(1, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name);
         ast_channel_unlock(targetcall_pvt->owner);
      }
   }
   if (targetcall_pvt)
      ao2_t_ref(targetcall_pvt, -1, "drop targetcall_pvt");
   return 1;
}

static int lws2sws	(	char *	msgbuf,
		int	len
	)			[static]

Parse multiline SIP headers into one header This is enabled if pedanticsipchecking is enabled.

Definition at line 6609 of file chan_sip.c.

Referenced by handle_request_do().

{
   int h = 0, t = 0; 
   int lws = 0; 

   for (; h < len;) { 
      /* Eliminate all CRs */ 
      if (msgbuf[h] == '\r') { 
         h++; 
         continue; 
      } 
      /* Check for end-of-line */ 
      if (msgbuf[h] == '\n') { 
         /* Check for end-of-message */ 
         if (h + 1 == len) 
            break; 
         /* Check for a continuation line */ 
         if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') { 
            /* Merge continuation line */ 
            h++; 
            continue; 
         } 
         /* Propagate LF and start new line */ 
         msgbuf[t++] = msgbuf[h++]; 
         lws = 0;
         continue; 
      } 
      if (msgbuf[h] == ' ' || msgbuf[h] == '\t') { 
         if (lws) { 
            h++; 
            continue; 
         } 
         msgbuf[t++] = msgbuf[h++]; 
         lws = 1; 
         continue; 
      } 
      msgbuf[t++] = msgbuf[h++]; 
      if (lws) 
         lws = 0; 
   } 
   msgbuf[t] = '\0'; 
   return t; 
}

static void make_our_tag	(	char *	tagbuf,
		size_t	len
	)			[static]

Make our SIP dialog tag.

Definition at line 6184 of file chan_sip.c.

References ast_random().

Referenced by handle_request_invite(), handle_request_subscribe(), sip_alloc(), transmit_register(), and transmit_response_using_temp().

{
   snprintf(tagbuf, len, "as%08lx", ast_random());
}

static int manager_show_registry	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Show SIP registrations in the manager API.

Definition at line 12845 of file chan_sip.c.

References ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_listack(), ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_RDLOCK, ASTOBJ_UNLOCK, regl, regstate2str(), s, STANDARD_SIP_PORT, and total.

Referenced by load_module().

{
   const char *id = astman_get_header(m, "ActionID");
   char idtext[256] = "";
   int total = 0;

   if (!ast_strlen_zero(id))
      snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);

   astman_send_listack(s, m, "Registrations will follow", "start");

   ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
      ASTOBJ_RDLOCK(iterator);
      astman_append(s,
         "Event: RegistryEntry\r\n"
         "Host: %s\r\n"
         "Port: %d\r\n"
         "Username: %s\r\n"
         "Refresh: %d\r\n"
         "State: %s\r\n"
         "RegistrationTime: %ld\r\n"
         "\r\n", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT,
                 iterator->username, iterator->refresh, regstate2str(iterator->regstate), (long) iterator->regtime.tv_sec);
      ASTOBJ_UNLOCK(iterator);
      total++;
   } while(0));

   astman_append(s,
      "Event: RegistrationsComplete\r\n"
      "EventList: Complete\r\n"
      "ListItems: %d\r\n"
      "%s"
      "\r\n", total, idtext);
   
   return 0;
}

static int manager_sip_qualify_peer	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Qualify SIP peers in the manager API.

Definition at line 13569 of file chan_sip.c.

References _sip_qualify_peer(), ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_error(), and s.

Referenced by load_module().

{
   const char *a[4];
   const char *peer;

   peer = astman_get_header(m, "Peer");
   if (ast_strlen_zero(peer)) {
      astman_send_error(s, m, "Peer: <name> missing.");
      return 0;
   }
   a[0] = "sip";
   a[1] = "qualify";
   a[2] = "peer";
   a[3] = peer;

   _sip_qualify_peer(1, -1, s, m, 4, a);
   astman_append(s, "\r\n\r\n" );
   return 0;
}

static int manager_sip_show_peer	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Show SIP peers in the manager API.

Definition at line 13510 of file chan_sip.c.

References _sip_show_peer(), ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_error(), and s.

Referenced by load_module().

{
   const char *a[4];
   const char *peer;

   peer = astman_get_header(m, "Peer");
   if (ast_strlen_zero(peer)) {
      astman_send_error(s, m, "Peer: <name> missing.");
      return 0;
   }
   a[0] = "sip";
   a[1] = "show";
   a[2] = "peer";
   a[3] = peer;

   _sip_show_peer(1, -1, s, m, 4, a);
   astman_append(s, "\r\n\r\n" );
   return 0;
}

static int manager_sip_show_peers	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Show SIP peers in the manager API.

Definition at line 12891 of file chan_sip.c.

References _sip_show_peers(), ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_listack(), s, and total.

Referenced by load_module().

{
   const char *id = astman_get_header(m, "ActionID");
   const char *a[] = {"sip", "show", "peers"};
   char idtext[256] = "";
   int total = 0;

   if (!ast_strlen_zero(id))
      snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);

   astman_send_listack(s, m, "Peer status list will follow", "start");
   /* List the peers in separate manager events */
   _sip_show_peers(-1, &total, s, m, 3, a);
   /* Send final confirmation */
   astman_append(s,
   "Event: PeerlistComplete\r\n"
   "EventList: Complete\r\n"
   "ListItems: %d\r\n"
   "%s"
   "\r\n", total, idtext);
   return 0;
}

static int manager_sipnotify	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Definition at line 9786 of file chan_sip.c.

References ao2_t_link, ao2_t_unlink, ast_set_flag, ast_sip_ouraddrfor(), ast_strlen_zero(), ast_variables_destroy(), astman_get_header(), astman_get_variables(), astman_send_ack(), astman_send_error(), build_callid_pvt(), build_via(), create_addr(), dialog_ref(), dialog_unlink_all(), dialog_unref(), dialogs, sip_pvt::flags, sip_pvt::ourip, s, sip_pvt::sa, sip_alloc(), SIP_NOTIFY, SIP_OUTGOING, sip_scheddestroy(), SIP_TRANS_TIMEOUT, transmit_notify_custom(), and TRUE.

Referenced by load_module().

{
   const char *channame = astman_get_header(m, "Channel");
   struct ast_variable *vars = astman_get_variables(m);
   struct sip_pvt *p;

   if (ast_strlen_zero(channame)) {
      astman_send_error(s, m, "SIPNotify requires a channel name");
      return 0;
   }

   if (!strncasecmp(channame, "sip/", 4)) {
      channame += 4;
   }

   if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY))) {
      astman_send_error(s, m, "Unable to build sip pvt data for notify (memory/socket error)");
      return 0;
   }

   if (create_addr(p, channame, NULL, 0)) {
      /* Maybe they're not registered, etc. */
      dialog_unlink_all(p, TRUE, TRUE);
      dialog_unref(p, "unref dialog inside for loop" );
      /* sip_destroy(p); */
      astman_send_error(s, m, "Could not create address");
      return 0;
   }

   /* Notify is outgoing call */
   ast_set_flag(&p->flags[0], SIP_OUTGOING);

   /* Recalculate our side, and recalculate Call ID */
   ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
   build_via(p);
   ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
   build_callid_pvt(p);
   ao2_t_link(dialogs, p, "Linking in new name");
   dialog_ref(p, "bump the count of p, which transmit_sip_request will decrement.");
   sip_scheddestroy(p, SIP_TRANS_TIMEOUT);

   if (!transmit_notify_custom(p, vars)) {
      astman_send_ack(s, m, "Notify Sent");
   } else {
      astman_send_error(s, m, "Unable to send notify");
   }
   ast_variables_destroy(vars);
   return 0;
}

static int map_s_x	(	const struct _map_x_s *	table,
		const char *	s,
		int	errorvalue
	)			[static]

map from a string to an integer value, case insensitive. If no match is found, return errorvalue.

Definition at line 2337 of file chan_sip.c.

References _map_x_s::s, table, and _map_x_s::x.

Referenced by str2dtmfmode(), str2stmode(), and str2strefresher().

{
   const struct _map_x_s *cur;

   for (cur = table; cur->s; cur++)
      if (!strcasecmp(cur->s, s))
         return cur->x;
   return errorvalue;
}

static const char* map_x_s	(	const struct _map_x_s *	table,
		int	x,
		const char *	errorstring
	)			[static]

map from an integer value to a string. If no match is found, return errorstring

Definition at line 2324 of file chan_sip.c.

References _map_x_s::s, table, and _map_x_s::x.

Referenced by dtmfmode2str(), insecure2str(), nat2str(), referstatus2str(), regstate2str(), stmode2str(), and strefresher2str().

{
   const struct _map_x_s *cur;

   for (cur = table; cur->s; cur++)
      if (cur->x == x)
         return cur->s;
   return errorstring;
}

static int method_match	(	enum sipmethod	id,
		const char *	name
	)			[static]

returns true if 'name' (with optional trailing whitespace) matches the sip method 'id'. Strictly speaking, SIP methods are case SENSITIVE, but we do a case-insensitive comparison to be more tolerant. following Jon Postel's rule: Be gentle in what you accept, strict with what you send

Definition at line 2706 of file chan_sip.c.

References sip_request::len, sip_methods, and text.

Referenced by __sip_semi_ack(), and find_sip_method().

{
   int len = strlen(sip_methods[id].text);
   int l_name = name ? strlen(name) : 0;
   /* true if the string is long enough, and ends with whitespace, and matches */
   return (l_name >= len && name[len] < 33 &&
      !strncasecmp(sip_methods[id].text, name, len));
}

static void mwi_event_cb	(	const struct ast_event *	,
		void *
	)			[static]

Receive MWI events that we have subscribed to.

Definition at line 11083 of file chan_sip.c.

References ao2_lock(), ao2_unlock(), and sip_send_mwi_to_peer().

{
   struct sip_peer *peer = userdata;

   ao2_lock(peer);
   sip_send_mwi_to_peer(peer, event, 0);
   ao2_unlock(peer);
}

static const char * nat2str

(

int

nat

)

const [static]

Convert NAT setting to text string.

Definition at line 12629 of file chan_sip.c.

References map_x_s(), and natmodes.

Referenced by _sip_show_peer(), sip_show_channel(), sip_show_settings(), and sip_show_users().

{
   return map_x_s(natmodes, nat, "Unknown");
}

static struct sip_proxy* obproxy_get	(	struct sip_pvt *	dialog,
		struct sip_peer *	peer
	)			[static]

Get default outbound proxy or global proxy.

Definition at line 2681 of file chan_sip.c.

References append_history, ast_debug, global_outboundproxy, sip_proxy::name, and sip_peer::outboundproxy.

Referenced by create_addr(), create_addr_from_peer(), and transmit_register().

{
   if (peer && peer->outboundproxy) {
      if (sipdebug)
         ast_debug(1, "OBPROXY: Applying peer OBproxy to this call\n");
      append_history(dialog, "OBproxy", "Using peer obproxy %s", peer->outboundproxy->name);
      return peer->outboundproxy;
   }
   if (global_outboundproxy.name[0]) {
      if (sipdebug)
         ast_debug(1, "OBPROXY: Applying global OBproxy to this call\n");
      append_history(dialog, "OBproxy", "Using global obproxy %s", global_outboundproxy.name);
      return &global_outboundproxy;
   }
   if (sipdebug)
      ast_debug(1, "OBPROXY: Not applying OBproxy to this call\n");
   return NULL;
}

static void parse_copy	(	struct sip_request *	dst,
		const struct sip_request *	src
	)			[static]

Copy SIP request, parse it.

Definition at line 3413 of file chan_sip.c.

References copy_request(), and parse_request().

Referenced by send_request(), and send_response().

{
   copy_request(dst, src);
   parse_request(dst);
}

int parse_minse	(	const char *	p_hdrval,
		int *const	p_interval
	)			[static]

Session-Timers: Function for parsing Min-SE header.

Definition at line 20571 of file chan_sip.c.

References ast_debug, ast_log(), ast_skip_blanks(), ast_strlen_zero(), and LOG_WARNING.

Referenced by handle_request_invite(), and proc_422_rsp().

{
   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "Null Min-SE header\n");
      return -1;
   }

   *p_interval = 0;
   p_hdrval = ast_skip_blanks(p_hdrval);
   if (!sscanf(p_hdrval, "%d", p_interval)) {
      ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
      return -1;
   }

   ast_debug(2, "Received Min-SE: %d\n", *p_interval);
   return 0;
}

static void parse_moved_contact	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Parse 302 Moved temporalily response.

Definition at line 15723 of file chan_sip.c.

References ao2_ref, ast_copy_string(), ast_debug, ast_string_field_build, ast_string_field_set, ast_test_flag, sip_socket::fd, sip_pvt::flags, get_header(), get_in_brackets(), get_transport(), sip_pvt::owner, pbx_builtin_setvar_helper(), remove_uri_parameters(), s, SIP_PROMISCREDIR, SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, SIP_TRANSPORT_UDP, SIPBUFSIZE, sip_pvt::socket, strcasestr(), sip_socket::tcptls_session, and sip_socket::type.

Referenced by handle_response().

{
   char tmp[SIPBUFSIZE];
   char *s, *e, *t, *trans;
   char *domain;
   enum sip_transport transport = SIP_TRANSPORT_UDP;

   ast_copy_string(tmp, get_header(req, "Contact"), sizeof(tmp));
   if ((t = strchr(tmp, ',')))
      *t = '\0';

   s = get_in_brackets(tmp);
   if ((trans = strcasestr(s, ";transport="))) do {
      trans += 11;

      if ((e = strchr(trans, ';')))
         *e = '\0';

      if (!strncasecmp(trans, "tcp", 3))
         transport = SIP_TRANSPORT_TCP;
      else if (!strncasecmp(trans, "tls", 3))
         transport = SIP_TRANSPORT_TLS;
      else {
         if (strncasecmp(trans, "udp", 3))
            ast_debug(1, "received contact with an invalid transport, '%s'\n", s);
         transport = SIP_TRANSPORT_UDP;
      }
   } while(0);
   s = remove_uri_parameters(s);

   if (p->socket.tcptls_session) {
      ao2_ref(p->socket.tcptls_session, -1);
      p->socket.tcptls_session = NULL;
   }

   p->socket.fd = -1;
   p->socket.type = transport;

   if (ast_test_flag(&p->flags[0], SIP_PROMISCREDIR)) {
      char *host = NULL;
      if (!strncasecmp(s, "sip:", 4))
         s += 4;
      else if (!strncasecmp(s, "sips:", 5))
         s += 5;
      e = strchr(s, '/');
      if (e)
         *e = '\0';
      if ((host = strchr(s, '@'))) {
         *host++ = '\0';
         ast_debug(2, "Found promiscuous redirection to 'SIP/%s::::%s@%s'\n", s, get_transport(transport), host);
         if (p->owner)
            ast_string_field_build(p->owner, call_forward, "SIP/%s::::%s@%s", s, get_transport(transport), host);
      } else {
         ast_debug(2, "Found promiscuous redirection to 'SIP/::::%s@%s'\n", get_transport(transport), s);
         if (p->owner)
            ast_string_field_build(p->owner, call_forward, "SIP/::::%s@%s", get_transport(transport), s);
      }
   } else {
      e = strchr(tmp, '@');
      if (e) {
         *e++ = '\0';
         domain = e;
      } else {
         /* No username part */
         domain = tmp;
      }
      e = strchr(tmp, '/');   /* WHEN do we hae a forward slash in the URI? */
      if (e)
         *e = '\0';

      if (!strncasecmp(s, "sip:", 4))
         s += 4;
      else if (!strncasecmp(s, "sips:", 5))
         s += 5;
      e = strchr(s, ';');  /* And username ; parameters? */
      if (e)
         *e = '\0';  
      ast_debug(2, "Received 302 Redirect to extension '%s' (domain %s)\n", s, domain);
      if (p->owner) {
         pbx_builtin_setvar_helper(p->owner, "SIPDOMAIN", domain);
         ast_string_field_set(p->owner, call_forward, s);
      }
   }
}

static int parse_ok_contact	(	struct sip_pvt *	pvt,
		struct sip_request *	req
	)			[static]

Save contact header for 200 OK on INVITE.

Definition at line 10504 of file chan_sip.c.

References ast_copy_string(), ast_string_field_set, get_header(), get_in_brackets(), SIPBUFSIZE, and TRUE.

Referenced by handle_request_invite(), handle_request_subscribe(), and handle_response_invite().

{
   char contact[SIPBUFSIZE]; 
   char *c;

   /* Look for brackets */
   ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
   c = get_in_brackets(contact);

   /* Save full contact to call pvt for later bye or re-invite */
   ast_string_field_set(pvt, fullcontact, c);

   /* Save URI for later ACKs, BYE or RE-invites */
   ast_string_field_set(pvt, okcontacturi, c);

   /* We should return false for URI:s we can't handle,
      like tel:, mailto:,ldap: etc */
   return TRUE;      
}

static enum parse_register_result parse_register_contact	(	struct sip_pvt *	pvt,
		struct sip_peer *	p,
		struct sip_request *	req
	)			[static]

Parse contact header and save registration (peer registration).

Definition at line 10582 of file chan_sip.c.

References sip_peer::addr, ahp, ao2_t_link, ao2_t_unlink, ast_apply_ha(), ast_copy_string(), ast_db_put(), ast_gethostbyname(), ast_inet_ntoa(), ast_log(), ast_sched_add(), AST_SCHED_DEL_UNREF, ast_sched_when(), AST_SENSE_ALLOW, ast_strdupa, ast_string_field_build, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verb, ast_verbose, sip_peer::contactha, copy_socket_data(), sip_request::data, destroy_association(), EVENT_FLAG_SYSTEM, sip_peer::expire, expire_register(), sip_pvt::expiry, FALSE, sip_peer::flags, sip_peer::fullcontact, get_header(), get_in_brackets(), global_contact_ha, hp, inaddrcmp(), sip_peer::is_realtime, sip_peer::lastms, LOG_NOTICE, LOG_WARNING, manager_event, sip_peer::name, PARSE_REGISTER_FAILED, PARSE_REGISTER_QUERY, PARSE_REGISTER_UPDATE, parse_uri(), peers_by_ip, sip_pvt::recv, ref_peer(), register_peer_exten(), sip_peer::rt_fromcontact, s, sched, SIP_NAT_ROUTE, SIP_PAGE2_RTCACHEFRIENDS, sip_poke_peer(), SIP_TRANSPORT_TLS, SIP_TRANSPORT_UDP, SIPBUFSIZE, sip_pvt::sipoptions, sip_peer::sipoptions, sip_pvt::socket, sip_request::socket, sip_peer::socket, STANDARD_SIP_PORT, STANDARD_TLS_PORT, strcasestr(), strsep(), sip_socket::type, unref_peer(), sip_peer::useragent, sip_peer::username, VERBOSE_PREFIX_3, and VERBOSITY_ATLEAST.

Referenced by register_verify().

{
   char contact[SIPBUFSIZE]; 
   char data[SIPBUFSIZE];
   const char *expires = get_header(req, "Expires");
   int expire = atoi(expires);
   char *curi, *host, *pt, *curi2;
   int port;
   const char *useragent;
   struct hostent *hp;
   struct ast_hostent ahp;
   struct sockaddr_in oldsin, testsin;

   ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));

   if (ast_strlen_zero(expires)) {  /* No expires header, try look in Contact: */
      char *s = strcasestr(contact, ";expires=");
      if (s) {
         expires = strsep(&s, ";"); /* trim ; and beyond */
         if (sscanf(expires + 9, "%d", &expire) != 1)
            expire = default_expiry;
      } else {
         /* Nothing has been specified */
         expire = default_expiry;
      }
   }

   if (peer->socket.type == req->socket.type)
      copy_socket_data(&peer->socket, &req->socket);
   copy_socket_data(&pvt->socket, &req->socket);

   /* Look for brackets */
   curi = contact;
   if (strchr(contact, '<') == NULL)   /* No <, check for ; and strip it */
      strsep(&curi, ";");  /* This is Header options, not URI options */
   curi = get_in_brackets(contact);
   curi2 = ast_strdupa(curi);

   /* if they did not specify Contact: or Expires:, they are querying
      what we currently have stored as their contact address, so return
      it
   */
   if (ast_strlen_zero(curi) && ast_strlen_zero(expires)) {
      /* If we have an active registration, tell them when the registration is going to expire */
      if (peer->expire > -1 && !ast_strlen_zero(peer->fullcontact))
         pvt->expiry = ast_sched_when(sched, peer->expire);
      return PARSE_REGISTER_QUERY;
   } else if (!strcasecmp(curi, "*") || !expire) { /* Unregister this peer */
      /* This means remove all registrations and return OK */
      memset(&peer->addr, 0, sizeof(peer->addr));

      AST_SCHED_DEL_UNREF(sched, peer->expire,
            unref_peer(peer, "remove register expire ref"));

      destroy_association(peer);
      
      register_peer_exten(peer, FALSE);   /* Remove extension from regexten= setting in sip.conf */
      peer->fullcontact[0] = '\0';
      peer->useragent[0] = '\0';
      peer->sipoptions = 0;
      peer->lastms = 0;
      pvt->expiry = 0;

      ast_verb(3, "Unregistered SIP '%s'\n", peer->name);

      manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\n", peer->name);
      return PARSE_REGISTER_UPDATE;
   }

   /* Store whatever we got as a contact from the client */
   ast_copy_string(peer->fullcontact, curi, sizeof(peer->fullcontact));

   /* For the 200 OK, we should use the received contact */
   ast_string_field_build(pvt, our_contact, "<%s>", curi);

   /* Make sure it's a SIP URL */
   if (pvt->socket.type == SIP_TRANSPORT_TLS) {
      if (parse_uri(curi, "sips:", &curi, NULL, &host, &pt, NULL)) {
         if (parse_uri(curi2, "sip:", &curi, NULL, &host, &pt, NULL))
            ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
      }
      port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
   } else {
      if (parse_uri(curi, "sip:", &curi, NULL, &host, &pt, NULL))
         ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
      port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
   }

   oldsin = peer->addr;

   /* If we were already linked into the peers_by_ip container unlink ourselves so nobody can find us */
   if (peer->addr.sin_addr.s_addr) {
      ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
   }

   /* Check that they're allowed to register at this IP */
   /* XXX This could block for a long time XXX */
   hp = ast_gethostbyname(host, &ahp);
   if (!hp)  {
      ast_log(LOG_WARNING, "Invalid host '%s'\n", host);
      *peer->fullcontact = '\0';
      ast_string_field_set(pvt, our_contact, "");
      return PARSE_REGISTER_FAILED;
   }
   memcpy(&testsin.sin_addr, hp->h_addr, sizeof(testsin.sin_addr));
   if (  ast_apply_ha(global_contact_ha, &testsin) != AST_SENSE_ALLOW ||
         ast_apply_ha(peer->contactha, &testsin) != AST_SENSE_ALLOW) {
      ast_log(LOG_WARNING, "Host '%s' disallowed by rule\n", host);
      *peer->fullcontact = '\0';
      ast_string_field_set(pvt, our_contact, "");
      return PARSE_REGISTER_FAILED;
   }

   if (!ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE)) {
      peer->addr.sin_family = AF_INET;
      memcpy(&peer->addr.sin_addr, hp->h_addr, sizeof(peer->addr.sin_addr));
      peer->addr.sin_port = htons(port);
   } else {
      /* Don't trust the contact field.  Just use what they came to us
         with */
      peer->addr = pvt->recv;
   }

   /* Now that our address has been updated put ourselves back into the container for lookups */
   ao2_t_link(peers_by_ip, peer, "ao2_link into peers_by_ip table");

   /* Save SIP options profile */
   peer->sipoptions = pvt->sipoptions;

   if (!ast_strlen_zero(curi) && ast_strlen_zero(peer->username))
      ast_copy_string(peer->username, curi, sizeof(peer->username));

   AST_SCHED_DEL_UNREF(sched, peer->expire,
         unref_peer(peer, "remove register expire ref"));

   if (expire > max_expiry)
      expire = max_expiry;
   if (expire < min_expiry)
      expire = min_expiry;
   if (peer->is_realtime && !ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
      peer->expire = -1;
   } else {
      peer->expire = ast_sched_add(sched, (expire + 10) * 1000, expire_register, 
            ref_peer(peer, "add registration ref"));
      if (peer->expire == -1) {
         unref_peer(peer, "remote registration ref");
      }
   }
   pvt->expiry = expire;
   snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port), expire, peer->username, peer->fullcontact);
   /* Saving TCP connections is useless, we won't be able to reconnect 
      XXX WHY???? XXX
      \todo check this
   */
   if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP)) 
      ast_db_put("SIP/Registry", peer->name, data);
   manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\nPort: %d\r\n", peer->name,  ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port));

   /* Is this a new IP address for us? */
   if (VERBOSITY_ATLEAST(2) && inaddrcmp(&peer->addr, &oldsin)) {
      ast_verbose(VERBOSE_PREFIX_3 "Registered SIP '%s' at %s port %d\n", peer->name, ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port));
   }
   sip_poke_peer(peer, 0);
   register_peer_exten(peer, 1);
   
   /* Save User agent */
   useragent = get_header(req, "User-Agent");
   if (strcasecmp(useragent, peer->useragent)) {   /* XXX copy if they are different ? */
      ast_copy_string(peer->useragent, useragent, sizeof(peer->useragent));
      ast_verb(4, "Saved useragent \"%s\" for peer %s\n", peer->useragent, peer->name);
   }
   return PARSE_REGISTER_UPDATE;
}

static int parse_request

(

struct sip_request *

req

)

[static]

Parse a SIP message.

Note:

this function is used both on incoming and outgoing packets

Definition at line 6656 of file chan_sip.c.

References ast_debug, ast_strlen_zero(), sip_request::data, sip_request::header, sip_request::headers, sip_request::line, SIP_MAX_HEADERS, SIP_MAX_LINES, and ast_str::str.

Referenced by _sip_tcp_helper_thread(), handle_request_do(), initialize_initreq(), and parse_copy().

{
   char *c = req->data->str;
   ptrdiff_t *dst = req->header;
   int i = 0, lim = SIP_MAX_HEADERS - 1;
   unsigned int skipping_headers = 0;
   ptrdiff_t current_header_offset = 0;
   char *previous_header = "";

   req->header[0] = 0;
   req->headers = -1;   /* mark that we are working on the header */
   for (; *c; c++) {
      if (*c == '\r') {    /* remove \r */
         *c = '\0';
      } else if (*c == '\n') {   /* end of this line */
         *c = '\0';
         current_header_offset = (c + 1) - req->data->str;
         previous_header = req->data->str + dst[i];
         if (skipping_headers) {
            /* check to see if this line is blank; if so, turn off
               the skipping flag, so the next line will be processed
               as a body line */
            if (ast_strlen_zero(previous_header)) {
               skipping_headers = 0;
            }
            dst[i] = current_header_offset; /* record start of next line */
            continue;
         }
         if (sipdebug) {
            ast_debug(4, "%7s %2d [%3d]: %s\n",
                 req->headers < 0 ? "Header" : "Body",
                 i, (int) strlen(previous_header), previous_header);
         }
         if (ast_strlen_zero(previous_header) && req->headers < 0) {
            req->headers = i; /* record number of header lines */
            dst = req->line;  /* start working on the body */
            i = 0;
            lim = SIP_MAX_LINES - 1;
         } else { /* move to next line, check for overflows */
            if (i++ == lim) {
               /* if we're processing headers, then skip any remaining
                  headers and move on to processing the body, otherwise
                  we're done */
               if (req->headers != -1) {
                  break;
               } else {
                  req->headers = i;
                  dst = req->line;
                  i = 0;
                  lim = SIP_MAX_LINES - 1;
                  skipping_headers = 1;
               }
            }
         }
         dst[i] = current_header_offset; /* record start of next line */
      }
   }

   /* Check for last header or body line without CRLF. The RFC for SDP requires CRLF,
      but since some devices send without, we'll be generous in what we accept. However,
      if we've already reached the maximum number of lines for portion of the message
      we were parsing, we can't accept any more, so just ignore it.
   */
   previous_header = req->data->str + dst[i];
   if ((i < lim) && !ast_strlen_zero(previous_header)) {
      if (sipdebug) {
         ast_debug(4, "%7s %2d [%3d]: %s\n",
              req->headers < 0 ? "Header" : "Body",
              i, (int) strlen(previous_header), previous_header );
      }
      i++;
   }

   /* update count of header or body lines */
   if (req->headers >= 0) {   /* we are in the body */
      req->lines = i;
   } else {       /* no body */
      req->headers = i;
      req->lines = 0;
      /* req->data->used will be a NULL byte */
      req->line[0] = req->data->used;
   }

   if (*c) {
      ast_log(LOG_WARNING, "Too many lines, skipping <%s>\n", c);
   }

   /* Split up the first line parts */
   return determine_firstline_parts(req);
}

int parse_session_expires	(	const char *	p_hdrval,
		int *const	p_interval,
		enum st_refresher *const	p_ref
	)			[static]

Session-Timers: Function for parsing Session-Expires header.

Definition at line 20591 of file chan_sip.c.

References ast_debug, ast_log(), ast_skip_blanks(), ast_strdupa, ast_strlen_zero(), LOG_WARNING, SESSION_TIMER_REFRESHER_AUTO, SESSION_TIMER_REFRESHER_UAC, SESSION_TIMER_REFRESHER_UAS, and strsep().

Referenced by handle_request_invite(), and handle_response_invite().

{
   char *p_token;
   int  ref_idx;
   char *p_se_hdr;

   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "Null Session-Expires header\n");
      return -1;
   }

   *p_ref = SESSION_TIMER_REFRESHER_AUTO;
   *p_interval = 0;

   p_se_hdr = ast_strdupa(p_hdrval);
   p_se_hdr = ast_skip_blanks(p_se_hdr);

   while ((p_token = strsep(&p_se_hdr, ";"))) {
      p_token = ast_skip_blanks(p_token);
      if (!sscanf(p_token, "%d", p_interval)) {
         ast_log(LOG_WARNING, "Parsing of Session-Expires failed\n");
         return -1;
      }

      ast_debug(2, "Session-Expires: %d\n", *p_interval);

      if (!p_se_hdr)
         continue;
      
      ref_idx = strlen("refresher=");
      if (!strncasecmp(p_se_hdr, "refresher=", ref_idx)) {
         p_se_hdr += ref_idx;
         p_se_hdr = ast_skip_blanks(p_se_hdr);

         if (!strncasecmp(p_se_hdr, "uac", strlen("uac"))) {
            *p_ref = SESSION_TIMER_REFRESHER_UAC;
            ast_debug(2, "Refresher: UAC\n");
         } else if (!strncasecmp(p_se_hdr, "uas", strlen("uas"))) {
            *p_ref = SESSION_TIMER_REFRESHER_UAS;
            ast_debug(2, "Refresher: UAS\n");
         } else {
            ast_log(LOG_WARNING, "Invalid refresher value %s\n", p_se_hdr);
            return -1;
         }
         break;
      }
   }
   return 0;
}

static unsigned int parse_sip_options	(	struct sip_pvt *	pvt,
		const char *	supported
	)			[static]

Parse supported header in incoming packet.

Definition at line 2730 of file chan_sip.c.

References ARRAY_LEN, ast_debug, ast_skip_blanks(), ast_strdupa, ast_strlen_zero(), FALSE, cfsip_options::id, sip_request::next, SIP_OPT_UNKNOWN, sip_options, sip_pvt::sipoptions, text, and TRUE.

Referenced by handle_request_invite().

{
   char *next, *sep;
   char *temp;
   unsigned int profile = 0;
   int i, found;

   if (ast_strlen_zero(supported) )
      return 0;
   temp = ast_strdupa(supported);

   if (sipdebug)
      ast_debug(3, "Begin: parsing SIP \"Supported: %s\"\n", supported);

   for (next = temp; next; next = sep) {
      found = FALSE;
      if ( (sep = strchr(next, ',')) != NULL)
         *sep++ = '\0';
      next = ast_skip_blanks(next);
      if (sipdebug)
         ast_debug(3, "Found SIP option: -%s-\n", next);
      for (i = 0; i < ARRAY_LEN(sip_options); i++) {
         if (!strcasecmp(next, sip_options[i].text)) {
            profile |= sip_options[i].id;
            found = TRUE;
            if (sipdebug)
               ast_debug(3, "Matched SIP option: %s\n", next);
            break;
         }
      }

      /* This function is used to parse both Suported: and Require: headers.
      Let the caller of this function know that an unknown option tag was 
      encountered, so that if the UAC requires it then the request can be 
      rejected with a 420 response. */
      if (!found)
         profile |= SIP_OPT_UNKNOWN;

      if (!found && sipdebug) {
         if (!strncasecmp(next, "x-", 2))
            ast_debug(3, "Found private SIP option, not supported: %s\n", next);
         else
            ast_debug(3, "Found no match for SIP option: %s (Please file bug report!)\n", next);
      }
   }

   if (pvt)
      pvt->sipoptions = profile;
   return profile;
}

static int parse_uri	(	char *	uri,
		char *	scheme,
		char **	ret_name,
		char **	pass,
		char **	domain,
		char **	port,
		char **	options
	)			[static]

* parses a URI in its components.

Note:

• If scheme is specified, drop it from the top.
• If a component is not requested, do not split around it.

This means that if we don't have domain, we cannot split name:pass and domain:port. It is safe to call with ret_name, pass, domain, port pointing all to the same place. Init pointers to empty string so we never get NULL dereferencing. Overwrites the string. return 0 on success, other values on error.

 * general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...] 
 * 

Definition at line 3621 of file chan_sip.c.

References ast_debug, name, and strsep().

Referenced by __set_address_from_contact(), check_user_full(), and parse_register_contact().

{
   char *name = NULL;
   int error = 0;

   /* init field as required */
   if (pass)
      *pass = "";
   if (port)
      *port = "";
   if (scheme) {
      int l = strlen(scheme);
      if (!strncasecmp(uri, scheme, l))
         uri += l;
      else {
         ast_debug(1, "Missing scheme '%s' in '%s'\n", scheme, uri);
         error = -1;
      }
   }
   if (!domain) {
      /* if we don't want to split around domain, keep everything as a name,
       * so we need to do nothing here, except remember why.
       */
   } else {
      /* store the result in a temp. variable to avoid it being
       * overwritten if arguments point to the same place.
       */
      char *c, *dom = "";

      if ((c = strchr(uri, '@')) == NULL) {
         /* domain-only URI, according to the SIP RFC. */
         dom = uri;
         name = "";
      } else {
         *c++ = '\0';
         dom = c;
         name = uri;
      }

      /* Remove options in domain and name */
      dom = strsep(&dom, ";");
      name = strsep(&name, ";");

      if (port && (c = strchr(dom, ':'))) { /* Remove :port */
         *c++ = '\0';
         *port = c;
      }
      if (pass && (c = strchr(name, ':'))) { /* user:password */
         *c++ = '\0';
         *pass = c;
      }
      *domain = dom;
   }
   if (ret_name)  /* same as for domain, store the result only at the end */
      *ret_name = name;
   if (options)
      *options = uri ? uri : "";

   return error;
}

static int peer_cmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Note:

The only member of the peer used here is the name field

Definition at line 1690 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, and sip_peer::name.

{
   struct sip_peer *peer = obj, *peer2 = arg;

   return !strcasecmp(peer->name, peer2->name) ? CMP_MATCH | CMP_STOP : 0;
}

static int peer_dump_func	(	void *	userobj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 13114 of file chan_sip.c.

References ao2_t_ref, ast_cli(), and sip_peer::name.

Referenced by sip_show_objects().

{
   struct sip_peer *peer = userobj;
   int refc = ao2_t_ref(userobj, 0, "");
   int *fd = arg;
   
   ast_cli(*fd, "name: %s\ntype: peer\nobjflags: %d\nrefcount: %d\n\n", 
          peer->name, 0, refc);
   return 0;
}

static int peer_hash_cb	(	const void *	obj,
		const int	flags
	)			[static]

Note:

The only member of the peer used here is the name field

Definition at line 1680 of file chan_sip.c.

References ast_str_case_hash(), and sip_peer::name.

{
   const struct sip_peer *peer = obj;

   return ast_str_case_hash(peer->name);
}

static int peer_ipcmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Note:

the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.

Definition at line 1717 of file chan_sip.c.

References sip_peer::addr, ast_test_flag, CMP_MATCH, CMP_STOP, sip_peer::flags, and SIP_INSECURE_PORT.

Referenced by load_module().

{
   struct sip_peer *peer = obj, *peer2 = arg;

   if (peer->addr.sin_addr.s_addr != peer2->addr.sin_addr.s_addr) {
      return 0;
   }
   
   if (!ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT) && !ast_test_flag(&peer2->flags[0], SIP_INSECURE_PORT)) {
      if (peer->addr.sin_port == peer2->addr.sin_port) {
         return CMP_MATCH | CMP_STOP;
      } else {
         return 0;
      }
   }

   return CMP_MATCH | CMP_STOP;
}

static int peer_iphash_cb	(	const void *	obj,
		const int	flags
	)			[static]

Note:

the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.

Definition at line 1700 of file chan_sip.c.

References sip_peer::addr, ast_test_flag, sip_peer::flags, and SIP_INSECURE_PORT.

Referenced by load_module().

{
   const struct sip_peer *peer = obj;
   int ret1 = peer->addr.sin_addr.s_addr;
   if (ret1 < 0)
      ret1 = -ret1;
   
   if (ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT)) {
      return ret1;
   } else {
      return ret1 + peer->addr.sin_port;
   }
}

static int peer_is_marked	(	void *	peerobj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 13304 of file chan_sip.c.

References CMP_MATCH, and sip_peer::the_mark.

Referenced by sip_do_reload(), and sip_prune_realtime().

{
   struct sip_peer *peer = peerobj;
   return peer->the_mark ? CMP_MATCH : 0;
}

static void peer_mailboxes_to_str	(	struct ast_str **	mailbox_str,
		struct sip_peer *	peer
	)			[static]

list peer mailboxes to CLI

Definition at line 13607 of file chan_sip.c.

References AST_LIST_NEXT, AST_LIST_TRAVERSE, ast_str_append(), ast_strlen_zero(), sip_mailbox::entry, mailbox, sip_peer::mailboxes, and S_OR.

Referenced by _sip_show_peer(), function_sippeer(), show_channels_cb(), and sip_send_mwi_to_peer().

{
   struct sip_mailbox *mailbox;

   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      ast_str_append(mailbox_str, 0, "%s%s%s%s",
         mailbox->mailbox,
         ast_strlen_zero(mailbox->context) ? "" : "@",
         S_OR(mailbox->context, ""),
         AST_LIST_NEXT(mailbox, entry) ? "," : "");
   }
}

static int peer_markall_func	(	void *	device,
		void *	arg,
		int	flags
	)			[static]

Definition at line 22038 of file chan_sip.c.

References sip_peer::the_mark.

Referenced by reload_config().

{
   struct sip_peer *peer = device;
   peer->the_mark = 1;
   return 0;
}

static int peer_status	(	struct sip_peer *	peer,
		char *	status,
		int	statuslen
	)			[static]

Definition at line 12696 of file chan_sip.c.

References ast_copy_string(), sip_peer::lastms, and sip_peer::maxms.

{
   int res = 0;
   if (peer->maxms) {
      if (peer->lastms < 0) {
         ast_copy_string(status, "UNREACHABLE", statuslen);
      } else if (peer->lastms > peer->maxms) {
         snprintf(status, statuslen, "LAGGED (%d ms)", peer->lastms);
         res = 1;
      } else if (peer->lastms) {
         snprintf(status, statuslen, "OK (%d ms)", peer->lastms);
         res = 1;
      } else {
         ast_copy_string(status, "UNKNOWN", statuslen);
      }
   } else { 
      ast_copy_string(status, "Unmonitored", statuslen);
      /* Checking if port is 0 */
      res = -1;
   }
   return res;
}

int peercomparefunc	(	const void *	a,
		const void *	b
	)

Definition at line 12934 of file chan_sip.c.

Referenced by _sip_show_peers().

{
   struct sip_peer **ap = (struct sip_peer **)a;
   struct sip_peer **bp = (struct sip_peer **)b;
   return strcmp((*ap)->name, (*bp)->name);
}

static void print_codec_to_cli	(	int	fd,
		struct ast_codec_pref *	pref
	)			[static]

Print codec list from preference to CLI/manager.

Definition at line 13439 of file chan_sip.c.

References ast_cli(), ast_codec_pref_index(), ast_getformatname(), and ast_codec_pref::framing.

Referenced by _sip_show_peer(), handle_skinny_show_line(), sip_show_settings(), and sip_show_user().

{
   int x, codec;

   for(x = 0; x < 32 ; x++) {
      codec = ast_codec_pref_index(pref, x);
      if (!codec)
         break;
      ast_cli(fd, "%s", ast_getformatname(codec));
      ast_cli(fd, ":%d", pref->framing[x]);
      if (x < 31 && ast_codec_pref_index(pref, x + 1))
         ast_cli(fd, ",");
   }
   if (!x)
      ast_cli(fd, "none");
}

static void print_group	(	int	fd,
		ast_group_t	group,
		int	crlf
	)			[static]

Print call group and pickup group.

Definition at line 13164 of file chan_sip.c.

References ast_cli(), ast_print_group(), and buf.

Referenced by _sip_show_peer(), and sip_show_user().

{
   char buf[256];
   ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_group(buf, sizeof(buf), group) );
}

static void proc_422_rsp	(	struct sip_pvt *	p,
		struct sip_request *	rsp
	)			[static]

Handle 422 response to INVITE with session-timer requested.

Session-Timers: An INVITE originated by Asterisk that asks for session-timers support from the UAS can result into a 422 response. This is how a UAS or an intermediary proxy server tells Asterisk that the session refresh interval offered by Asterisk is too low for them. The proc_422_rsp() function handles a 422 response. It extracts the Min-SE header that comes back in 422 and sends a new INVITE accordingly.

Definition at line 20649 of file chan_sip.c.

References ast_log(), ast_strlen_zero(), get_header(), LOG_WARNING, parse_minse(), SIP_INVITE, sip_st_dlg::st_interval, sip_pvt::stimer, and transmit_invite().

Referenced by handle_response_invite().

{
   int rtn;
   const char *p_hdrval;
   int minse;

   p_hdrval = get_header(rsp, "Min-SE");
   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "422 response without a Min-SE header %s\n", p_hdrval);
      return;
   }
   rtn = parse_minse(p_hdrval, &minse);
   if (rtn != 0) {
      ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
      return;
   }
   p->stimer->st_interval = minse;
   transmit_invite(p, SIP_INVITE, 1, 2); 
}

static int proc_session_timer

(

const void *

vp

)

[static]

Session-Timers: Process session refresh timeout event.

Definition at line 20499 of file chan_sip.c.

References ast_channel::_state, ast_channel_trylock, ast_channel_unlock, ast_debug, ast_log(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), AST_STATE_UP, sip_pvt::callid, dialog_unref(), FALSE, LOG_ERROR, LOG_WARNING, sip_pvt::outgoing_call, sip_pvt::owner, SESSION_TIMER_REFRESHER_UAC, SESSION_TIMER_REFRESHER_UAS, sip_pvt_lock, sip_pvt_unlock, sip_st_dlg::st_active, sip_st_dlg::st_expirys, sip_st_dlg::st_ref, sip_st_dlg::st_schedid, sip_pvt::stimer, stop_session_timer(), transmit_reinvite_with_sdp(), and TRUE.

Referenced by start_session_timer().

{
   struct sip_pvt *p = (struct sip_pvt *) vp;
   int sendreinv = FALSE;
   int res = 0;

   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in proc_session_timer - %s\n", p->callid);
      goto return_unref;
   }

   ast_debug(2, "Session timer expired: %d - %s\n", p->stimer->st_schedid, p->callid);

   if (!p->owner) {
      goto return_unref;
   }

   if ((p->stimer->st_active != TRUE) || (p->owner->_state != AST_STATE_UP)) {
      goto return_unref;
   }

   switch (p->stimer->st_ref) {
   case SESSION_TIMER_REFRESHER_UAC:
      if (p->outgoing_call == TRUE) {
         sendreinv = TRUE;
      }
      break;
   case SESSION_TIMER_REFRESHER_UAS:
      if (p->outgoing_call != TRUE) {
         sendreinv = TRUE;
      }
      break;
   default:
      ast_log(LOG_ERROR, "Unknown session refresher %d\n", p->stimer->st_ref);
      goto return_unref;
   }

   if (sendreinv == TRUE) {
      res = 1;
      transmit_reinvite_with_sdp(p, FALSE, TRUE);
   } else {
      p->stimer->st_expirys++;
      if (p->stimer->st_expirys >= 2) {
         ast_log(LOG_WARNING, "Session-Timer expired - %s\n", p->callid);

         while (p->owner && ast_channel_trylock(p->owner)) {
            sip_pvt_unlock(p);
            usleep(1);
            sip_pvt_lock(p);
         }

         ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
         ast_channel_unlock(p->owner);
      }
   }

return_unref:
   if (!res) {
      /* An error occurred.  Stop session timer processing */
      p->stimer->st_schedid = -1;
      stop_session_timer(p);
      
      /* If we are not asking to be rescheduled, then we need to release our
       * reference to the dialog. */
      dialog_unref(p, "removing session timer ref");
   }

   return res;
}

static void process_request_queue	(	struct sip_pvt *	p,
		int *	recount,
		int *	nounlock
	)			[static]

Definition at line 19774 of file chan_sip.c.

References ast_free, AST_LIST_REMOVE_HEAD, ast_log(), handle_incoming(), LOG_DEBUG, sip_request::next, option_debug, and sip_pvt::request_queue.

Referenced by handle_request_do().

{
   struct sip_request *req;

   while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
      if (handle_incoming(p, req, &p->recv, recount, nounlock) == -1) {
         /* Request failed */
         if (option_debug) {
            ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
         }
      }
      ast_free(req);
   }
}

static int process_sdp	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	t38action
	)			[static]

Process SIP SDP offer, select formats and activate RTP channels If offer is rejected, we will not change any properties of the call Return 0 on success, a negative value on errors. Must be called after find_sdp().

< RTP Audio port number

< RTP Video port number

< RTP Text port number

< media socket address

< Video socket address

< Text socket address

< RTP Audio host IP

< RTP video host IP

< RTP text host IP

Definition at line 6852 of file chan_sip.c.

References append_history, ast_clear_flag, ast_codec_choose(), ast_codec_pref_setsize(), AST_CONTROL_HOLD, AST_CONTROL_UNHOLD, ast_debug, AST_FORMAT_AUDIO_MASK, AST_FORMAT_T140RED, ast_getformatname_multiple(), ast_gethostbyname(), ast_inet_ntoa(), ast_log(), ast_null_frame, ast_queue_control(), ast_queue_control_data(), ast_queue_frame(), ast_rtp_alloc_size(), ast_rtp_codec_getformat(), ast_rtp_codec_getpref(), ast_rtp_codec_setpref(), AST_RTP_DTMF, ast_rtp_get_current_formats(), ast_rtp_get_peer(), ast_rtp_lookup_mime_multiple(), ast_rtp_new_init(), AST_RTP_OPT_G726_NONSTANDARD, ast_rtp_pt_clear(), ast_rtp_pt_copy(), ast_rtp_set_m_type(), ast_rtp_set_peer(), ast_rtp_set_rtpmap_type(), ast_rtp_setdtmf(), ast_rtp_setdtmfcompensate(), ast_rtp_stop(), ast_rtp_unset_m_type(), ast_set_flag, ast_set_read_format(), ast_set_write_format(), ast_skip_blanks(), ast_strdupa, ast_strlen_zero(), ast_test_flag, ast_threadstorage_get(), ast_udptl_set_error_correction_scheme(), ast_udptl_set_far_max_datagram(), ast_udptl_set_local_max_datagram(), ast_udptl_set_peer(), ast_udptl_stop(), ast_verbose, sip_pvt::autoframing, buf, sip_pvt::callid, sip_pvt::capability, t38properties::capability, change_t38_state(), sip_request::data, sip_request::debug, EVENT_FLAG_CALL, FALSE, sip_pvt::flags, format, get_sdp(), get_sdp_iterate(), hp, sip_pvt::jointcapability, t38properties::jointcapability, sip_pvt::jointnoncodeccapability, sip_pvt::lastinvite, sip_pvt::lastrtprx, sip_pvt::lastrtptx, sip_request::len, LOG_DEBUG, LOG_ERROR, LOG_NOTICE, LOG_WARNING, manager_event, MAX_RTP_PT, sip_pvt::mohsuggest, ast_channel::name, ast_channel::nativeformats, sip_pvt::noncodeccapability, sip_pvt::notext, sip_pvt::novideo, option_debug, sip_pvt::owner, sip_pvt::peercapability, t38properties::peercapability, sip_registry::portno, sip_pvt::prefs, ast_channel::readformat, sip_pvt::red, RED_MAX_GENERATION, sip_pvt::rtp, rtp_red_init(), s, S_OR, SDP_MAX_RTPMAP_CODECS, sip_request::sdp_start, SDP_T38_ACCEPT, SDP_T38_INITIATE, sip_pvt::session_modify, sip_pvt::sessionversion_remote, sip_debug_test_pvt(), SIP_DTMF, SIP_DTMF_AUTO, SIP_DTMF_INBAND, SIP_DTMF_RFC2833, SIP_G726_NONSTANDARD, SIP_NAT, SIP_PAGE2_CALL_ONHOLD, SIP_PAGE2_CALL_ONHOLD_ACTIVE, SIP_PAGE2_CALL_ONHOLD_INACTIVE, SIP_PAGE2_CALL_ONHOLD_ONEDIR, SIP_PAGE2_RFC2833_COMPENSATE, SIP_PAGE2_UDPTL_DESTINATION, sip_peer_hold(), SIPBUFSIZE, t38properties::state, ast_str::str, strsep(), sip_pvt::t38, T38_DISABLED, T38_ENABLED, T38_LOCAL_REINVITE, T38_PEER_DIRECT, T38_PEER_REINVITE, T38FAX_FILL_BIT_REMOVAL, T38FAX_RATE_12000, T38FAX_RATE_14400, T38FAX_RATE_2400, T38FAX_RATE_4800, T38FAX_RATE_7200, T38FAX_RATE_9600, T38FAX_RATE_MANAGEMENT_LOCAL_TCF, T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF, T38FAX_TRANSCODING_JBIG, T38FAX_TRANSCODING_MMR, T38FAX_UDP_EC_FEC, T38FAX_UDP_EC_NONE, T38FAX_UDP_EC_REDUNDANCY, T38FAX_VERSION_0, T38FAX_VERSION_1, text, sip_pvt::trtp, TRUE, sip_pvt::udptl, UDPTL_ERROR_CORRECTION_FEC, UDPTL_ERROR_CORRECTION_NONE, UDPTL_ERROR_CORRECTION_REDUNDANCY, ast_channel::uniqueid, sip_pvt::vrtp, and ast_channel::writeformat.

{
   const char *m;    /* SDP media offer */
   const char *c;
   const char *a;
   const char *o;    /* Pointer to o= line */
   char *o_copy;     /* Copy of o= line */
   char *token;
   char host[258];
   int len = -1;
   int portno = -1;     /*!< RTP Audio port number */
   int vportno = -1;    /*!< RTP Video port number */
   int tportno = -1;    /*!< RTP Text port number */
   int udptlportno = -1;
   int peert38capability = 0;
   char s[256];
   int old = 0;

   /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */ 
   int peercapability = 0, peernoncodeccapability = 0;
   int vpeercapability = 0, vpeernoncodeccapability = 0;
   int tpeercapability = 0, tpeernoncodeccapability = 0;
   struct sockaddr_in sin;    /*!< media socket address */
   struct sockaddr_in vsin;   /*!< Video socket address */
   struct sockaddr_in tsin;   /*!< Text socket address */

   const char *codecs;
   struct hostent *hp;     /*!< RTP Audio host IP */
   struct hostent *vhp = NULL;   /*!< RTP video host IP */
   struct hostent *thp = NULL;   /*!< RTP text host IP */
   struct ast_hostent audiohp;
   struct ast_hostent videohp;
   struct ast_hostent texthp;
   int codec;
   int destiterator = 0;
   int iterator;
   int sendonly = -1;
   int numberofports;
   struct ast_rtp *newaudiortp, *newvideortp, *newtextrtp;  /* Buffers for codec handling */
   int newjointcapability;          /* Negotiated capability */
   int newpeercapability;
   int newnoncodeccapability;
   int numberofmediastreams = 0;
   int debug = sip_debug_test_pvt(p);
      
   int found_rtpmap_codecs[SDP_MAX_RTPMAP_CODECS];
   int last_rtpmap_codec=0;

   char buf[SIPBUFSIZE];
   uint64_t rua_version;
   
   int red_data_pt[10];
   int red_num_gen = 0;
   int red_pt = 0;

   char *red_cp;           /* For T.140 red */
   char red_fmtp[100] = "empty";    /* For T.140 red */

   if (!p->rtp) {
      ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
      return -1;
   }

   /* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
#ifdef LOW_MEMORY
   newaudiortp = ast_threadstorage_get(&ts_audio_rtp, ast_rtp_alloc_size());
#else
   newaudiortp = alloca(ast_rtp_alloc_size());
#endif
   memset(newaudiortp, 0, ast_rtp_alloc_size());
   ast_rtp_new_init(newaudiortp);
   ast_rtp_pt_clear(newaudiortp);

#ifdef LOW_MEMORY
   newvideortp = ast_threadstorage_get(&ts_video_rtp, ast_rtp_alloc_size());
#else
   newvideortp = alloca(ast_rtp_alloc_size());
#endif
   memset(newvideortp, 0, ast_rtp_alloc_size());
   ast_rtp_new_init(newvideortp);
   ast_rtp_pt_clear(newvideortp);

#ifdef LOW_MEMORY
   newtextrtp = ast_threadstorage_get(&ts_text_rtp, ast_rtp_alloc_size());
#else
   newtextrtp = alloca(ast_rtp_alloc_size());
#endif
   memset(newtextrtp, 0, ast_rtp_alloc_size());
   ast_rtp_new_init(newtextrtp);
   ast_rtp_pt_clear(newtextrtp);

   /* Update our last rtprx when we receive an SDP, too */
   p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */

   /* Store the SDP version number of remote UA. This will allow us to 
   distinguish between session modifications and session refreshes. If 
   the remote UA does not send an incremented SDP version number in a 
   subsequent RE-INVITE then that means its not changing media session. 
   The RE-INVITE may have been sent to update connected party, remote  
   target or to refresh the session (Session-Timers).  Asterisk must not 
   change media session and increment its own version number in answer 
   SDP in this case. */ 
   
   o = get_sdp(req, "o");
   if (ast_strlen_zero(o)) {
      ast_log(LOG_WARNING, "SDP sytax error. SDP without an o= line\n");
      return -1;
   }

   o_copy = ast_strdupa(o);
   token = strsep(&o_copy, " ");  /* Skip username   */
   if (!o_copy) { 
      ast_log(LOG_WARNING, "SDP sytax error in o= line username\n");
      return -1;
   }
   token = strsep(&o_copy, " ");  /* Skip session-id */
   if (!o_copy) { 
      ast_log(LOG_WARNING, "SDP sytax error in o= line session-id\n");
      return -1;
   }
   token = strsep(&o_copy, " ");  /* Version         */
   if (!o_copy) { 
      ast_log(LOG_WARNING, "SDP sytax error in o= line\n");
      return -1;
   }
   if (!sscanf(token, "%" SCNu64, &rua_version)) {
      ast_log(LOG_WARNING, "SDP sytax error in o= line version\n");
      return -1;
   }

   if (p->sessionversion_remote < 0 || p->sessionversion_remote != rua_version) {
      p->sessionversion_remote = rua_version;
      p->session_modify = TRUE;
   } else if (p->sessionversion_remote == rua_version) {
      p->session_modify = FALSE;
      ast_debug(2, "SDP version number same as previous SDP\n");
      return 0;
   } 

   /* Try to find first media stream */
   m = get_sdp(req, "m");
   destiterator = req->sdp_start;
   c = get_sdp_iterate(&destiterator, req, "c");
   if (ast_strlen_zero(m) || ast_strlen_zero(c)) {
      ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c);
      return -1;
   }

   /* Check for IPv4 address (not IPv6 yet) */
   if (sscanf(c, "IN IP4 %256s", host) != 1) {
      ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
      return -1;
   }

   /* XXX This could block for a long time, and block the main thread! XXX */
   hp = ast_gethostbyname(host, &audiohp);
   if (!hp) {
      ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c);
      return -1;
   }
   vhp = hp;   /* Copy to video address as default too */
   thp = hp;   /* Copy to text address as default too */
   
   iterator = req->sdp_start;
   /* default: novideo and notext set */
   p->novideo = TRUE;
   p->notext = TRUE;

   if (p->vrtp)
      ast_rtp_pt_clear(newvideortp);  /* Must be cleared in case no m=video line exists */
 
   if (p->trtp)
      ast_rtp_pt_clear(newtextrtp);  /* Must be cleared in case no m=text line exists */

   /* Find media streams in this SDP offer */
   while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
      int x;
      int audio = FALSE;
      int video = FALSE;
      int text = FALSE;

      numberofports = 1;
      len = -1;
      if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
          (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
         audio = TRUE;
         numberofmediastreams++;
         /* Found audio stream in this media definition */
         portno = x;
         /* Scan through the RTP payload types specified in a "m=" line: */
         for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
            if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
               ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
               return -1;
            }
            if (debug)
               ast_verbose("Found RTP audio format %d\n", codec);
            ast_rtp_set_m_type(newaudiortp, codec);
         }
      } else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
          (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
         video = TRUE;
         p->novideo = FALSE;
         numberofmediastreams++;
         vportno = x;
         /* Scan through the RTP payload types specified in a "m=" line: */
         for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
            if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
               ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
               return -1;
            }
            if (debug)
               ast_verbose("Found RTP video format %d\n", codec);
            ast_rtp_set_m_type(newvideortp, codec);
         }
      } else if ((sscanf(m, "text %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
          (sscanf(m, "text %d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
         text = TRUE;
         p->notext = FALSE;
         numberofmediastreams++;
         tportno = x;
         /* Scan through the RTP payload types specified in a "m=" line: */
         for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
            if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
               ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
               return -1;
            }
            if (debug)
               ast_verbose("Found RTP text format %d\n", codec);
            ast_rtp_set_m_type(newtextrtp, codec);
         }
      } else if (p->udptl && ( (sscanf(m, "image %d udptl t38%n", &x, &len) == 1 && len > 0) || 
         (sscanf(m, "image %d UDPTL t38%n", &x, &len) == 1 && len > 0) )) {
         if (debug)
            ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
         udptlportno = x;
         numberofmediastreams++;
      } else 
         ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
      if (numberofports > 1)
         ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
      

      /* Check for Media-description-level-address for audio */
      c = get_sdp_iterate(&destiterator, req, "c");
      if (!ast_strlen_zero(c)) {
         if (sscanf(c, "IN IP4 %256s", host) != 1) {
            ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
         } else {
            /* XXX This could block for a long time, and block the main thread! XXX */
            if (audio) {
               if ( !(hp = ast_gethostbyname(host, &audiohp))) {
                  ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c);
                  return -2;
               }
            } else if (video) {
               if (!(vhp = ast_gethostbyname(host, &videohp))) {
                  ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
                  return -2;
               }
            } else if (text) {
               if (!(thp = ast_gethostbyname(host, &texthp))) {
                  ast_log(LOG_WARNING, "Unable to lookup RTP text host in secondary c= line, '%s'\n", c);
                  return -2;
               }
            }
         }

      }
   }
   if (portno == -1 && vportno == -1 && udptlportno == -1  && tportno == -1)
      /* No acceptable offer found in SDP  - we have no ports */
      /* Do not change RTP or VRTP if this is a re-invite */
      return -2;

   if (numberofmediastreams > 3)
      /* We have too many fax, audio and/or video and/or text media streams, fail this offer */
      return -3;

   /* RTP addresses and ports for audio and video */
   sin.sin_family = AF_INET;
   vsin.sin_family = AF_INET;
   tsin.sin_family = AF_INET;
   memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
   if (vhp)
      memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
   if (thp)
      memcpy(&tsin.sin_addr, thp->h_addr, sizeof(tsin.sin_addr));

   /* Setup UDPTL port number */
   if (p->udptl) {
      if (udptlportno > 0) {
         sin.sin_port = htons(udptlportno);
         if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
            struct sockaddr_in peer;
            ast_rtp_get_peer(p->rtp, &peer);
            if (peer.sin_addr.s_addr) {
               memcpy(&sin.sin_addr, &peer.sin_addr, sizeof(sin.sin_addr));
               if (debug) {
                  ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr));
               }
            }
         }
         ast_udptl_set_peer(p->udptl, &sin);
         if (debug)
            ast_debug(1, "Peer T.38 UDPTL is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
      } else {
         ast_udptl_stop(p->udptl);
         if (debug)
            ast_debug(1, "Peer doesn't provide T.38 UDPTL\n");
      }
   }

      
   if (p->rtp) {
      if (portno > 0) {
         sin.sin_port = htons(portno);
         ast_rtp_set_peer(p->rtp, &sin);
         if (debug)
            ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
      } else {
         if (udptlportno > 0) {
            if (debug)
               ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid %s\n", p->callid);
         } else {
            ast_rtp_stop(p->rtp);
            if (debug)
               ast_verbose("Peer doesn't provide audio. Callid %s\n", p->callid);
         }
      }
   }
   /* Setup video port number, assumes we have audio */
   if (vportno != -1)
      vsin.sin_port = htons(vportno);

   /* Setup text port number, assumes we have audio */
   if (tportno != -1)
      tsin.sin_port = htons(tportno);

   /* Next, scan through each "a=xxxx:" line, noting each
    * specified RTP payload type (with corresponding MIME subtype):
    */
   /* XXX This needs to be done per media stream, since it's media stream specific */
   iterator = req->sdp_start;
   while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
      char mimeSubtype[128];
      if (option_debug > 1) {
         int breakout = FALSE;
      
         /* If we're debugging, check for unsupported sdp options */
         if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
            if (debug)
               ast_verbose("Got unsupported a:rtcp in SDP offer \n");
            breakout = TRUE;
         } else if (!strncasecmp(a, "fmtp:", (size_t) 5)) {
            /* Format parameters:  Not supported */
            /* Note: This is used for codec parameters, like bitrate for
               G722 and video formats for H263 and H264 
               See RFC2327 for an example */
            if (debug)
               ast_verbose("Got unsupported a:fmtp in SDP offer \n");
            breakout = TRUE;
         } else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
            /* Video stuff:  Not supported */
            if (debug)
               ast_verbose("Got unsupported a:framerate in SDP offer \n");
            breakout = TRUE;
         } else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
            /* Video stuff:  Not supported */
            if (debug)
               ast_verbose("Got unsupported a:maxprate in SDP offer \n");
            breakout = TRUE;
         } else if (!strncasecmp(a, "crypto:", (size_t) 7)) {
            /* SRTP stuff, not yet supported */
            if (debug)
               ast_verbose("Got unsupported a:crypto in SDP offer \n");
            breakout = TRUE;
         }
         if (breakout)  /* We have a match, skip to next header */
            continue;
      }
      if (!strcasecmp(a, "sendonly")) {
         if (sendonly == -1)
            sendonly = 1;
         continue;
      } else if (!strcasecmp(a, "inactive")) {
         if (sendonly == -1)
            sendonly = 2;
         continue;
      }  else if (!strcasecmp(a, "sendrecv")) {
         if (sendonly == -1)
            sendonly = 0;
         continue;
      } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) {
         char *tmp = strrchr(a, ':');
         long int framing = 0;
         if (tmp) {
            tmp++;
            framing = strtol(tmp, NULL, 10);
            if (framing == LONG_MIN || framing == LONG_MAX) {
               framing = 0;
               ast_debug(1, "Can't read framing from SDP: %s\n", a);
            }
         }
         if (framing && p->autoframing) {
            struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
            int codec_n;
            int format = 0;
            for (codec_n = 0; codec_n < MAX_RTP_PT; codec_n++) {
               format = ast_rtp_codec_getformat(codec_n);
               if (!format)   /* non-codec or not found */
                  continue;
               if (option_debug)
                  ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
               ast_codec_pref_setsize(pref, format, framing);
            }
            ast_rtp_codec_setpref(p->rtp, pref);
         }
         continue;
         
      } else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
         /* count numbers of generations in fmtp */
         red_cp = &red_fmtp[strlen(red_fmtp)];
         strncpy(red_fmtp, a, 100);

         sscanf(red_cp, "%u", &red_data_pt[red_num_gen]);
         red_cp = strtok(red_cp, "/");
         while (red_cp && red_num_gen++ < RED_MAX_GENERATION) {
            sscanf(red_cp, "%u", &red_data_pt[red_num_gen]);
            red_cp = strtok(NULL, "/");
         }
         red_cp = red_fmtp;

      } else if (sscanf(a, "rtpmap: %u %127[^/]/", &codec, mimeSubtype) == 2) {
         /* We have a rtpmap to handle */

         if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
            /* Note: should really look at the 'freq' and '#chans' params too */
            /* Note: This should all be done in the context of the m= above */
            if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {         /* Video */
               if(ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
                  if (debug)
                     ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
                  found_rtpmap_codecs[last_rtpmap_codec] = codec;
                  last_rtpmap_codec++;
               } else {
                  ast_rtp_unset_m_type(newvideortp, codec);
                  if (debug) 
                     ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
               }
            } else if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
               if (p->trtp) {
                  /* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
                  ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
               }
            } else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
               if (p->trtp) {
                  ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
                  red_pt = codec;
                  sprintf(red_fmtp, "fmtp:%d ", red_pt); 

                  if (debug)
                     ast_verbose("Red submimetype has payload type: %d\n", red_pt);
               }
            } else {                                          /* Must be audio?? */
               if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
                           ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
                  if (debug)
                     ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
                  found_rtpmap_codecs[last_rtpmap_codec] = codec;
                  last_rtpmap_codec++;
               } else {
                  ast_rtp_unset_m_type(newaudiortp, codec);
                  if (debug) 
                     ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
               }
            }
         } else {
            if (debug)
               ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
         }

      }
   }
   
   if (udptlportno != -1) {
      int found = 0, x;
      
      old = 0;
      
      /* Scan trough the a= lines for T38 attributes and set apropriate fileds */
      iterator = req->sdp_start;
      while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
         if ((sscanf(a, "T38FaxMaxBuffer:%d", &x) == 1)) {
            found = 1;
            ast_debug(3, "MaxBufferSize:%d\n", x);
         } else if ((sscanf(a, "T38MaxBitRate:%d", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%d", &x) == 1)) {
            found = 1;
            ast_debug(3, "T38MaxBitRate: %d\n", x);
            switch (x) {
            case 14400:
               peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
               break;
            case 12000:
               peert38capability |= T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
               break;
            case 9600:
               peert38capability |= T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
               break;
            case 7200:
               peert38capability |= T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
               break;
            case 4800:
               peert38capability |= T38FAX_RATE_4800 | T38FAX_RATE_2400;
               break;
            case 2400:
               peert38capability |= T38FAX_RATE_2400;
               break;
            }
         } else if ((sscanf(a, "T38FaxVersion:%d", &x) == 1)) {
            found = 1;
            ast_debug(3, "FaxVersion: %d\n", x);
            if (x == 0)
               peert38capability |= T38FAX_VERSION_0;
            else if (x == 1)
               peert38capability |= T38FAX_VERSION_1;
         } else if ((sscanf(a, "T38FaxMaxDatagram:%d", &x) == 1) || (sscanf(a, "T38MaxDatagram:%d", &x) == 1)) {
            found = 1;
            ast_debug(3, "FaxMaxDatagram: %d\n", x);
            ast_udptl_set_far_max_datagram(p->udptl, x);
            ast_udptl_set_local_max_datagram(p->udptl, x);
         } else if ((strncmp(a, "T38FaxFillBitRemoval", 20) == 0)) {
            found = 1;
            if(sscanf(a, "T38FaxFillBitRemoval:%d", &x) == 1) {
                ast_debug(3, "FillBitRemoval: %d\n", x);
                if(x == 1)
               peert38capability |= T38FAX_FILL_BIT_REMOVAL;
            } else {
                ast_debug(3, "FillBitRemoval\n");
                peert38capability |= T38FAX_FILL_BIT_REMOVAL;
            }
         } else if ((strncmp(a, "T38FaxTranscodingMMR", 20) == 0)) {
            found = 1;
            if(sscanf(a, "T38FaxTranscodingMMR:%d", &x) == 1) {
                ast_debug(3, "Transcoding MMR: %d\n", x);
                if(x == 1)
               peert38capability |= T38FAX_TRANSCODING_MMR;
            } else {
                ast_debug(3, "Transcoding MMR\n");
                peert38capability |= T38FAX_TRANSCODING_MMR;
            }
         } else if ((strncmp(a, "T38FaxTranscodingJBIG", 21) == 0)) {
            found = 1;
            if(sscanf(a, "T38FaxTranscodingJBIG:%d", &x) == 1) {
                ast_debug(3, "Transcoding JBIG: %d\n", x);
                if(x == 1)
               peert38capability |= T38FAX_TRANSCODING_JBIG;
            } else {
                ast_debug(3, "Transcoding JBIG\n");
                peert38capability |= T38FAX_TRANSCODING_JBIG;
            }
         } else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) {
            found = 1;
            ast_debug(3, "RateManagement: %s\n", s);
            if (!strcasecmp(s, "localTCF"))
               peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF;
            else if (!strcasecmp(s, "transferredTCF"))
               peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
         } else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) {
            found = 1;
            ast_debug(3, "UDP EC: %s\n", s);
            if (!strcasecmp(s, "t38UDPRedundancy")) {
               peert38capability |= T38FAX_UDP_EC_REDUNDANCY;
               ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
            } else if (!strcasecmp(s, "t38UDPFEC")) {
               peert38capability |= T38FAX_UDP_EC_FEC;
               ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
            } else {
               peert38capability |= T38FAX_UDP_EC_NONE;
               ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
            }
         }
      }
      if (found) { /* Some cisco equipment returns nothing beside c= and m= lines in 200 OK T38 SDP */
         p->t38.peercapability = peert38capability;
         p->t38.jointcapability = (peert38capability & 255); /* Put everything beside supported speeds settings */
         peert38capability &= (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400);
         p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */
      }
      if (debug)
         ast_debug(1, "Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n",
            p->t38.capability,
            p->t38.peercapability,
            p->t38.jointcapability);

      /* Remote party offers T38, we need to update state */
      if (t38action == SDP_T38_ACCEPT) {
         if (p->t38.state == T38_LOCAL_REINVITE)
            change_t38_state(p, T38_ENABLED);
      } else if (t38action == SDP_T38_INITIATE) {
         if (p->owner && p->lastinvite) {
            change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
         } else {
            change_t38_state(p, T38_PEER_DIRECT); /* T38 Offered directly from peer in first invite */
         }
      }
   } else {
      change_t38_state(p, T38_DISABLED);
   }

   /* Now gather all of the codecs that we are asked for: */
   ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
   ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
   ast_rtp_get_current_formats(newtextrtp, &tpeercapability, &tpeernoncodeccapability);
 
   newjointcapability = p->capability & (peercapability | vpeercapability | tpeercapability);
   newpeercapability = (peercapability | vpeercapability | tpeercapability);
   newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
      
      
   if (debug) {
      /* shame on whoever coded this.... */
      char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE];

      ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n",
             ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability),
             ast_getformatname_multiple(s2, SIPBUFSIZE, peercapability),
             ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
             ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability),
             ast_getformatname_multiple(s5, SIPBUFSIZE, newjointcapability));

      ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
             ast_rtp_lookup_mime_multiple(s1, SIPBUFSIZE, p->noncodeccapability, 0, 0),
             ast_rtp_lookup_mime_multiple(s2, SIPBUFSIZE, peernoncodeccapability, 0, 0),
             ast_rtp_lookup_mime_multiple(s3, SIPBUFSIZE, newnoncodeccapability, 0, 0));
   }
   if (!newjointcapability) {
      /* If T.38 was not negotiated either, totally bail out... */
      if (!p->t38.jointcapability || !udptlportno) {
         ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
         /* Do NOT Change current setting */
         return -1;
      } else {
         ast_debug(3, "Have T.38 but no audio codecs, accepting offer anyway\n");
         return 0;
      }
   }

   /* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
      they are acceptable */
   p->jointcapability = newjointcapability;          /* Our joint codec profile for this call */
   p->peercapability = newpeercapability;            /* The other sides capability in latest offer */
   p->jointnoncodeccapability = newnoncodeccapability;   /* DTMF capabilities */

   if (p->jointcapability & AST_FORMAT_T140RED) {
      p->red = 1; 
      rtp_red_init(p->trtp, 300, red_data_pt, 2);
   } else {
      p->red = 0; 
   }

   ast_rtp_pt_copy(p->rtp, newaudiortp);
   if (p->vrtp)
      ast_rtp_pt_copy(p->vrtp, newvideortp);
   if (p->trtp)
      ast_rtp_pt_copy(p->trtp, newtextrtp);

   if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      if (newnoncodeccapability & AST_RTP_DTMF) {
         /* XXX Would it be reasonable to drop the DSP at this point? XXX */
         ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
         /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
         ast_rtp_setdtmf(p->rtp, 1);
         ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
      } else {
         ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
      }
   }

   /* Setup audio port number */
   if (p->rtp && sin.sin_port) {
      ast_rtp_set_peer(p->rtp, &sin);
      if (debug)
         ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
   }

   /* Setup video port number */
   if (p->vrtp && vsin.sin_port) {
      ast_rtp_set_peer(p->vrtp, &vsin);
      if (debug) 
         ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
   }

   /* Setup text port number */
   if (p->trtp && tsin.sin_port) {
      ast_rtp_set_peer(p->trtp, &tsin);
      if (debug) 
         ast_verbose("Peer text RTP is at port %s:%d\n", ast_inet_ntoa(tsin.sin_addr), ntohs(tsin.sin_port));
   }

   /* Ok, we're going with this offer */
   ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));

   if (!p->owner)    /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
      return 0;

   ast_debug(4, "We have an owner, now see if we need to change this call\n");

   if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
      if (debug) {
         char s1[SIPBUFSIZE], s2[SIPBUFSIZE];
         ast_debug(1, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", 
            ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcapability),
            ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats));
      }
      p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability) | (p->capability & tpeercapability);
      ast_set_read_format(p->owner, p->owner->readformat);
      ast_set_write_format(p->owner, p->owner->writeformat);
   }
   
   if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && sin.sin_addr.s_addr && (!sendonly || sendonly == -1)) {
      ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
      /* Activate a re-invite */
      ast_queue_frame(p->owner, &ast_null_frame);
      /* Queue Manager Unhold event */
      append_history(p, "Unhold", "%s", req->data->str);
      if (global_callevents)
         manager_event(EVENT_FLAG_CALL, "Hold",
                  "Status: Off\r\n"
                  "Channel: %s\r\n"
                  "Uniqueid: %s\r\n",
                  p->owner->name,
                  p->owner->uniqueid);
      if (global_notifyhold)
         sip_peer_hold(p, FALSE);
      ast_clear_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */
   } else if (!sin.sin_addr.s_addr || (sendonly && sendonly != -1)) {
      int already_on_hold = ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD);
      ast_queue_control_data(p->owner, AST_CONTROL_HOLD, 
                   S_OR(p->mohsuggest, NULL),
                   !ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
      if (sendonly)
         ast_rtp_stop(p->rtp);
      /* RTCP needs to go ahead, even if we're on hold!!! */
      /* Activate a re-invite */
      ast_queue_frame(p->owner, &ast_null_frame);
      /* Queue Manager Hold event */
      append_history(p, "Hold", "%s", req->data->str);
      if (global_callevents && !ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
         manager_event(EVENT_FLAG_CALL, "Hold",
                  "Status: On\r\n"
                  "Channel: %s\r\n"
                  "Uniqueid: %s\r\n",
                  p->owner->name, 
                  p->owner->uniqueid);
      }
      if (sendonly == 1)   /* One directional hold (sendonly/recvonly) */
         ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR);
      else if (sendonly == 2) /* Inactive stream */
         ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE);
      else
         ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ACTIVE);
      if (global_notifyhold && !already_on_hold)
         sip_peer_hold(p, TRUE);
   }
   
   return 0;
}

static struct sip_proxy* proxy_allocate	(	char *	name,
		char *	port,
		int	force
	)			[static]

Allocate and initialize sip proxy.

Definition at line 2667 of file chan_sip.c.

References ao2_alloc, ast_copy_string(), ast_strlen_zero(), proxy_update(), and STANDARD_SIP_PORT.

Referenced by build_peer().

{
   struct sip_proxy *proxy;
   proxy = ao2_alloc(sizeof(*proxy), NULL);
   if (!proxy)
      return NULL;
   proxy->force = force;
   ast_copy_string(proxy->name, name, sizeof(proxy->name));
   proxy->ip.sin_port = htons((!ast_strlen_zero(port) ? atoi(port) : STANDARD_SIP_PORT));
   proxy_update(proxy);
   return proxy;
}

static int proxy_update

(

struct sip_proxy *

proxy

)

[static]

Resolve DNS srv name or host name in a sip_proxy structure

Definition at line 2650 of file chan_sip.c.

References ast_get_ip_or_srv(), ast_log(), FALSE, inet_aton(), sip_proxy::ip, sip_proxy::last_dnsupdate, LOG_WARNING, sip_proxy::name, and TRUE.

Referenced by proxy_allocate().

{
   /* if it's actually an IP address and not a name,
           there's no need for a managed lookup */
   if (!inet_aton(proxy->name, &proxy->ip.sin_addr)) {
      /* Ok, not an IP address, then let's check if it's a domain or host */
      /* XXX Todo - if we have proxy port, don't do SRV */
      if (ast_get_ip_or_srv(&proxy->ip, proxy->name, global_srvlookup ? "_sip._udp" : NULL) < 0) {
         ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
         return FALSE;
      }
   }
   proxy->last_dnsupdate = time(NULL);
   return TRUE;
}

static int queue_request	(	struct sip_pvt *	p,
		const struct sip_request *	req
	)			[static]

Definition at line 19844 of file chan_sip.c.

References ast_calloc, AST_LIST_INSERT_TAIL, ast_sched_add(), copy_request(), dialog_ref(), dialog_unref(), sip_request::next, sip_pvt::request_queue, sip_pvt::request_queue_sched_id, sched, and scheduler_process_request_queue().

Referenced by handle_request_do().

{
   struct sip_request *newreq;

   if (!(newreq = ast_calloc(1, sizeof(*newreq)))) {
      return -1;
   }

   copy_request(newreq, req);
   AST_LIST_INSERT_TAIL(&p->request_queue, newreq, next);
   if (p->request_queue_sched_id == -1) {
      if ((p->request_queue_sched_id = ast_sched_add(sched, 10, scheduler_process_request_queue, dialog_ref(p, "Increment refcount to pass dialog pointer to sched callback"))) == -1) {
         dialog_unref(p, "Decrement refcount due to sched_add failure");
      }
   }

   return 0;
}

static struct sip_peer * realtime_peer	(	const char *	peername,
		struct sockaddr_in *	sin,
		int	devstate_only
	)			[static]

realtime_peer: Get peer from realtime storage Checks the "sippeers" realtime family from extconfig.conf Checks the "sipregs" realtime family from extconfig.conf if it's configured. This returns a pointer to a peer and because we use build_peer, we can rest assured that the refcount is bumped.

Note:

If this one loaded something, then we need to ensure that the host field matched. The only reason why we can't have this as a criteria is because we only have the IP address and the host field might be set as a name (and the reverse PTR might not match).

Definition at line 3945 of file chan_sip.c.

References ahp, ast_check_realtime(), ast_gethostbyname(), ast_inet_ntoa(), ast_load_realtime(), ast_variables_destroy(), hp, ipaddr, ast_variable::name, ast_variable::next, SENTINEL, ast_variable::value, and var.

{
   struct sip_peer *peer;
   struct ast_variable *var = NULL;
   struct ast_variable *varregs = NULL;
   struct ast_variable *tmp;
   struct ast_config *peerlist = NULL;
   char ipaddr[INET_ADDRSTRLEN];
   char portstring[6]; /*up to 5 digits plus null terminator*/
   char *cat = NULL;
   unsigned short portnum;
   int realtimeregs = ast_check_realtime("sipregs");

   /* First check on peer name */
   if (newpeername) {
      if (realtimeregs)
         varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);

      var = ast_load_realtime("sippeers", "name", newpeername, "host", "dynamic", SENTINEL);
      if (!var && sin)
         var = ast_load_realtime("sippeers", "name", newpeername, "host", ast_inet_ntoa(sin->sin_addr), SENTINEL);
      if (!var) {
         var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
         /*!\note
          * If this one loaded something, then we need to ensure that the host
          * field matched.  The only reason why we can't have this as a criteria
          * is because we only have the IP address and the host field might be
          * set as a name (and the reverse PTR might not match).
          */
         if (var && sin) {
            for (tmp = var; tmp; tmp = tmp->next) {
               if (!strcasecmp(tmp->name, "host")) {
                  struct hostent *hp;
                  struct ast_hostent ahp;
                  if (!(hp = ast_gethostbyname(tmp->value, &ahp)) || (memcmp(&hp->h_addr, &sin->sin_addr, sizeof(hp->h_addr)))) {
                     /* No match */
                     ast_variables_destroy(var);
                     var = NULL;
                  }
                  break;
               }
            }
         }
      }
   }

   if (!var && sin) {   /* Then check on IP address for dynamic peers */
      ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
      portnum = ntohs(sin->sin_port);
      sprintf(portstring, "%u", portnum);
      var = ast_load_realtime("sippeers", "host", ipaddr, "port", portstring, SENTINEL);  /* First check for fixed IP hosts */
      if (var) {
         if (realtimeregs) {
            newpeername = get_name_from_variable(var, newpeername);
            varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
         }
      } else {
         if (realtimeregs)
            varregs = ast_load_realtime("sipregs", "ipaddr", ipaddr, "port", portstring, SENTINEL); /* Then check for registered hosts */
         else
            var = ast_load_realtime("sippeers", "ipaddr", ipaddr, "port", portstring, SENTINEL); /* Then check for registered hosts */
         if (varregs) {
            newpeername = get_name_from_variable(varregs, newpeername);
            var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
         }
      }
      if (!var) { /*We couldn't match on ipaddress and port, so we need to check if port is insecure*/
         peerlist = ast_load_realtime_multientry("sippeers", "host", ipaddr, SENTINEL);
         if (peerlist) {
            var = get_insecure_variable_from_config(peerlist);
            if(var) {
               if (realtimeregs) {
                  newpeername = get_name_from_variable(var, newpeername);
                  varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
               }
            } else { /*var wasn't found in the list of "hosts", so try "ipaddr"*/
               peerlist = NULL;
               cat = NULL;
               peerlist = ast_load_realtime_multientry("sippeers", "ipaddr", ipaddr, SENTINEL);
               if(peerlist) {
                  var = get_insecure_variable_from_config(peerlist);
                  if(var) {
                     if (realtimeregs) {
                        newpeername = get_name_from_variable(var, newpeername);
                        varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
                     }
                  }
               }
            }
         } else {
            if (realtimeregs) {
               peerlist = ast_load_realtime_multientry("sipregs", "ipaddr", ipaddr, SENTINEL);
               if (peerlist) {
                  varregs = get_insecure_variable_from_config(peerlist);
                  if (varregs) {
                     newpeername = get_name_from_variable(varregs, newpeername);
                     var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
                  }
               }
            } else {
               peerlist = ast_load_realtime_multientry("sippeers", "ipaddr", ipaddr, SENTINEL);
               if (peerlist) {
                  var = get_insecure_variable_from_config(peerlist);
                  if (var) {
                     newpeername = get_name_from_variable(var, newpeername);
                     varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
                  }
               }
            }
         }
      }
   }

   if (!var) {
      if (peerlist)
         ast_config_destroy(peerlist);
      return NULL;
   }

   for (tmp = var; tmp; tmp = tmp->next) {
      /* If this is type=user, then skip this object. */
      if (!strcasecmp(tmp->name, "type") &&
          !strcasecmp(tmp->value, "user")) {
         if(peerlist)
            ast_config_destroy(peerlist);
         else {
            ast_variables_destroy(var);
            ast_variables_destroy(varregs);
         }
         return NULL;
      } else if (!newpeername && !strcasecmp(tmp->name, "name")) {
         newpeername = tmp->value;
      }
   }
   
   if (!newpeername) {  /* Did not find peer in realtime */
      ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", ipaddr);
      if(peerlist)
         ast_config_destroy(peerlist);
      else
         ast_variables_destroy(var);
      return NULL;
   }


   /* Peer found in realtime, now build it in memory */
   peer = build_peer(newpeername, var, varregs, TRUE);
   if (!peer) {
      if(peerlist)
         ast_config_destroy(peerlist);
      else {
         ast_variables_destroy(var);
         ast_variables_destroy(varregs);
      }
      return NULL;
   }

   ast_debug(3, "-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs);

   if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && !devstate_only) {
      /* Cache peer */
      ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
      if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
         AST_SCHED_REPLACE_UNREF(peer->expire, sched, global_rtautoclear * 1000, expire_register, peer,
               unref_peer(_data, "remove registration ref"),
               unref_peer(peer, "remove registration ref"),
               ref_peer(peer, "add registration ref"));
      }
      ao2_t_link(peers, peer, "link peer into peers table");
      if (peer->addr.sin_addr.s_addr) {
         ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
      }
   }
   peer->is_realtime = 1;
   if (peerlist)
      ast_config_destroy(peerlist);
   else {
      ast_variables_destroy(var);
      ast_variables_destroy(varregs);
   }

   return peer;
}

static void realtime_update_peer	(	const char *	peername,
		struct sockaddr_in *	sin,
		const char *	username,
		const char *	fullcontact,
		const char *	useragent,
		int	expirey,
		int	deprecated_username,
		int	lastms
	)			[static]

Update peer object in realtime storage If the Asterisk system name is set in asterisk.conf, we will use that name and store that in the "regserver" field in the sippeers table to facilitate multi-server setups.

Definition at line 3749 of file chan_sip.c.

References ast_check_realtime(), ast_config_AST_SYSTEM_NAME, ast_copy_string(), ast_inet_ntoa(), ast_strlen_zero(), ast_update_realtime(), ipaddr, sip_settings::rtsave_sysname, SENTINEL, and sip_cfg.

{
   char port[10];
   char ipaddr[INET_ADDRSTRLEN];
   char regseconds[20];
   char *tablename = NULL;
   char str_lastms[20];

   const char *sysname = ast_config_AST_SYSTEM_NAME;
   char *syslabel = NULL;

   time_t nowtime = time(NULL) + expirey;
   const char *fc = fullcontact ? "fullcontact" : NULL;

   int realtimeregs = ast_check_realtime("sipregs");

   tablename = realtimeregs ? "sipregs" : "sippeers";
   

   snprintf(str_lastms, sizeof(str_lastms), "%d", lastms);
   snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime);  /* Expiration time */
   ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
   snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
   
   if (ast_strlen_zero(sysname)) /* No system name, disable this */
      sysname = NULL;
   else if (sip_cfg.rtsave_sysname)
      syslabel = "regserver";

   if (fc) {
      ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
         "port", port, "regseconds", regseconds,
         deprecated_username ? "username" : "defaultuser", defaultuser,
         "useragent", useragent, "lastms", str_lastms,
         fc, fullcontact, syslabel, sysname, SENTINEL); /* note fc and syslabel _can_ be NULL */
   } else {
      ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
         "port", port, "regseconds", regseconds,
         "useragent", useragent, "lastms", str_lastms,
         deprecated_username ? "username" : "defaultuser", defaultuser,
         syslabel, sysname, SENTINEL); /* note syslabel _can_ be NULL */
   }
}

static void receive_message	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Receive SIP MESSAGE method messages.

Note:

We only handle messages within current calls currently Reference: RFC 3428

Definition at line 12516 of file chan_sip.c.

References AST_FRAME_TEXT, ast_log(), ast_queue_frame(), ast_verbose, buf, sip_pvt::callid, DEFAULT_TRANS_TIMEOUT, f, FALSE, get_header(), get_msg_text(), LOG_WARNING, sip_pvt::owner, sip_debug_test_pvt(), sip_scheddestroy(), and transmit_response().

Referenced by handle_request_message().

{
   char buf[1400];   
   struct ast_frame f;
   const char *content_type = get_header(req, "Content-Type");

   if (strncmp(content_type, "text/plain", strlen("text/plain"))) { /* No text/plain attachment */
      transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
      if (!p->owner)
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   }

   if (get_msg_text(buf, sizeof(buf), req, FALSE)) {
      ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid);
      transmit_response(p, "202 Accepted", req);
      if (!p->owner)
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   }

   if (p->owner) {
      if (sip_debug_test_pvt(p))
         ast_verbose("SIP Text message received: '%s'\n", buf);
      memset(&f, 0, sizeof(f));
      f.frametype = AST_FRAME_TEXT;
      f.subclass = 0;
      f.offset = 0;
      f.data.ptr = buf;
      f.datalen = strlen(buf);
      ast_queue_frame(p->owner, &f);
      transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */
      return;
   }

   /* Message outside of a call, we do not support that */
   ast_log(LOG_WARNING, "Received message to %s from %s, dropped it...\n  Content-Type:%s\n  Message: %s\n", get_header(req, "To"), get_header(req, "From"), content_type, buf);
   transmit_response(p, "405 Method Not Allowed", req);
   sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   return;
}

static struct sip_peer* ref_peer	(	struct sip_peer *	peer,
		char *	tag
	)			[static]

Definition at line 2497 of file chan_sip.c.

References ao2_t_ref.

Referenced by build_peer(), handle_request_subscribe(), handle_response_peerpoke(), parse_register_contact(), reg_source_db(), sip_poke_all_peers(), sip_poke_noanswer(), sip_poke_peer(), and sip_unregister().

{
   ao2_t_ref(peer, 1, tag);
   return peer;
}

static struct sip_proxy* ref_proxy	(	struct sip_pvt *	pvt,
		struct sip_proxy *	proxy
	)			[static]

maintain proper refcounts for a sip_pvt's outboundproxy

This function sets pvt's outboundproxy pointer to the one referenced by the proxy parameter. Because proxy may be a refcounted object, and because pvt's old outboundproxy may also be a refcounted object, we need to maintain the proper refcounts.

Parameters:

	pvt	The sip_pvt for which we wish to set the outboundproxy
	proxy	The sip_proxy which we will point pvt towards.

Returns:

Returns void

Definition at line 2514 of file chan_sip.c.

References ao2_ref, global_outboundproxy, and sip_pvt::outboundproxy.

Referenced by __sip_ack(), create_addr(), create_addr_from_peer(), and transmit_register().

{
   struct sip_proxy *old_obproxy = pvt->outboundproxy;
   /* Cool, now get the refs correct */
   if (proxy && proxy != &global_outboundproxy) {
      ao2_ref(proxy, +1);
   }
   pvt->outboundproxy = proxy;
   if (old_obproxy && old_obproxy != &global_outboundproxy) {
      ao2_ref(old_obproxy, -1);
   }
   return proxy;
}

static const char * referstatus2str

(

enum referstatus

rstatus

)

[static]

Convert transfer status to string.

Definition at line 2621 of file chan_sip.c.

References map_x_s(), and referstatusstrings.

Referenced by show_channels_cb().

{
   return map_x_s(referstatusstrings, rstatus, "");
}

static void reg_source_db

(

struct sip_peer *

peer

)

[static]

Get registration details from Asterisk DB.

Definition at line 10442 of file chan_sip.c.

References sip_peer::addr, ast_copy_string(), ast_db_get(), ast_debug, ast_inet_ntoa(), ast_random(), AST_SCHED_REPLACE_UNREF, sip_request::data, sip_peer::expire, expire_register(), sip_peer::fullcontact, inet_aton(), sip_peer::name, sip_peer::pokeexpire, ref_peer(), register_peer_exten(), sip_peer::rt_fromcontact, sched, sip_poke_peer(), sip_poke_peer_s(), sipsock, strsep(), TRUE, unref_peer(), and sip_peer::username.

{
   char data[256];
   struct in_addr in;
   int expire;
   int port;
   char *scan, *addr, *port_str, *expiry_str, *username, *contact;

   if (peer->rt_fromcontact) 
      return;
   if (ast_db_get("SIP/Registry", peer->name, data, sizeof(data)))
      return;

   scan = data;
   addr = strsep(&scan, ":");
   port_str = strsep(&scan, ":");
   expiry_str = strsep(&scan, ":");
   username = strsep(&scan, ":");
   contact = scan;   /* Contact include sip: and has to be the last part of the database entry as long as we use : as a separator */

   if (!inet_aton(addr, &in))
      return;

   if (port_str)
      port = atoi(port_str);
   else
      return;

   if (expiry_str)
      expire = atoi(expiry_str);
   else
      return;

   if (username)
      ast_copy_string(peer->username, username, sizeof(peer->username));
   if (contact)
      ast_copy_string(peer->fullcontact, contact, sizeof(peer->fullcontact));

   ast_debug(2, "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n",
       peer->name, peer->username, ast_inet_ntoa(in), port, expire);

   memset(&peer->addr, 0, sizeof(peer->addr));
   peer->addr.sin_family = AF_INET;
   peer->addr.sin_addr = in;
   peer->addr.sin_port = htons(port);
   if (sipsock < 0) {
      /* SIP isn't up yet, so schedule a poke only, pretty soon */
      AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, ast_random() % 5000 + 1, sip_poke_peer_s, peer,
            unref_peer(_data, "removing poke peer ref"),
            unref_peer(peer, "removing poke peer ref"),
            ref_peer(peer, "adding poke peer ref"));
   } else {
      sip_poke_peer(peer, 0);
   }
   AST_SCHED_REPLACE_UNREF(peer->expire, sched, (expire + 10) * 1000, expire_register, peer,
         unref_peer(_data, "remove registration ref"),
         unref_peer(peer, "remove registration ref"),
         ref_peer(peer, "add registration ref"));
   register_peer_exten(peer, TRUE);
}

static void register_peer_exten	(	struct sip_peer *	peer,
		int	onoff
	)			[static]

Automatically add peer extension to dial plan.

Definition at line 3794 of file chan_sip.c.

References ast_add_extension(), ast_context_find(), ast_context_remove_extension(), ast_copy_string(), ast_exists_extension(), ast_free_ptr, ast_log(), ast_strdup, ast_strlen_zero(), context, E_MATCH, ext, LOG_WARNING, sip_peer::name, pbx_find_extension(), sip_peer::regexten, S_OR, pbx_find_info::stacklen, and strsep().

{
   char multi[256];
   char *stringp, *ext, *context;
   struct pbx_find_info q = { .stacklen = 0 };

   /* XXX note that global_regcontext is both a global 'enable' flag and
    * the name of the global regexten context, if not specified
    * individually.
    */
   if (ast_strlen_zero(global_regcontext))
      return;

   ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
   stringp = multi;
   while ((ext = strsep(&stringp, "&"))) {
      if ((context = strchr(ext, '@'))) {
         *context++ = '\0';   /* split ext@context */
         if (!ast_context_find(context)) {
            ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
            continue;
         }
      } else {
         context = global_regcontext;
      }
      if (onoff) {
         if (!ast_exists_extension(NULL, context, ext, 1, NULL)) {
            ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
                ast_strdup(peer->name), ast_free_ptr, "SIP");
         }
      } else if (pbx_find_extension(NULL, NULL, &q, context, ext, 1, NULL, "", E_MATCH)) {
         ast_context_remove_extension(context, ext, 1, NULL);
      }
   }
}

static enum check_auth_result register_verify	(	struct sip_pvt *	p,
		struct sockaddr_in *	sin,
		struct sip_request *	req,
		char *	uri
	)			[static]

Verify registration of user

• Registration is done in several steps, first a REGISTER without auth to get a challenge (nonce) then a second one with auth
• Registration requests are only matched with peers that are marked as "dynamic".

Definition at line 11253 of file chan_sip.c.

References sip_peer::addr, ao2_t_link, ast_apply_ha(), ast_copy_flags, ast_copy_string(), AST_DEVICE_UNKNOWN, ast_devstate_changed(), ast_inet_ntoa(), AST_LIST_EMPTY, ast_log(), ast_rtp_codec_setpref(), ast_set_flag, ast_string_field_set, ast_test_flag, ast_uri_decode(), AUTH_ACL_FAILED, AUTH_BAD_TRANSPORT, AUTH_NOT_FOUND, AUTH_PEER_NOT_DYNAMIC, AUTH_SECRET_FAILED, AUTH_UNKNOWN_DOMAIN, AUTH_USERNAME_MISMATCH, sip_peer::autoframing, sip_pvt::autoframing, build_contact(), check_auth(), check_request_transport, check_sip_domain(), EVENT_FLAG_SYSTEM, exten, FALSE, find_peer(), FINDPEERS, sip_peer::flags, sip_pvt::flags, get_header(), get_in_brackets(), sip_peer::ha, sip_peer::host_dynamic, sip_request::ignore, sip_pvt::initreq, sip_peer::lastmsgssent, LOG_ERROR, LOG_NOTICE, LOG_WARNING, manager_event, sip_peer::md5secret, sip_peer::name, name, parse_register_contact(), PARSE_REGISTER_FAILED, PARSE_REGISTER_QUERY, PARSE_REGISTER_UPDATE, peers_by_ip, sip_peer::prefs, remove_uri_parameters(), sip_pvt::rtp, sip_peer::secret, sip_cancel_destroy(), SIP_NAT, SIP_PAGE2_REGISTERTRYING, SIP_PAGE2_SUBSCRIBEMWIONLY, SIP_PENDINGBYE, SIP_REGISTER, temp_peer(), terminate_uri(), transmit_fake_auth_response(), transmit_response(), transmit_response_with_date(), TRUE, unref_peer(), update_peer(), and XMIT_UNRELIABLE.

{
   enum check_auth_result res = AUTH_NOT_FOUND;
   struct sip_peer *peer;
   char tmp[256];
   char *name, *c;
   char *domain;

   terminate_uri(uri);  /* warning, overwrite the string */

   ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp));
   if (pedanticsipchecking)
      ast_uri_decode(tmp);

   c = get_in_brackets(tmp);
   c = remove_uri_parameters(c);

   if (!strncasecmp(c, "sip:", 4)) {
      name = c + 4;
   } else if (!strncasecmp(c, "sips:", 5)) {
      name = c + 5;
   } else {
      name = c;
      ast_log(LOG_NOTICE, "Invalid to address: '%s' from %s (missing sip:) trying to use anyway...\n", c, ast_inet_ntoa(sin->sin_addr));
   }

   /* XXX here too we interpret a missing @domain as a name-only
    * URI, whereas the RFC says this is a domain-only uri.
    */
   /* Strip off the domain name */
   if ((c = strchr(name, '@'))) {
      *c++ = '\0';
      domain = c;
      if ((c = strchr(domain, ':')))   /* Remove :port */
         *c = '\0';
      if (!AST_LIST_EMPTY(&domain_list)) {
         if (!check_sip_domain(domain, NULL, 0)) {
            transmit_response(p, "404 Not found (unknown domain)", &p->initreq);
            return AUTH_UNKNOWN_DOMAIN;
         }
      }
   }
   c = strchr(name, ';');  /* Remove any Username parameters */
   if (c)
      *c = '\0';

   ast_string_field_set(p, exten, name);
   build_contact(p);
   peer = find_peer(name, NULL, TRUE, FINDPEERS, FALSE);
   if (!(peer && ast_apply_ha(peer->ha, sin))) {
      /* Peer fails ACL check */
      if (peer) {
         unref_peer(peer, "register_verify: unref_peer: from find_peer operation");
         peer = NULL;
         res = AUTH_ACL_FAILED;
      } else
         res = AUTH_NOT_FOUND;
   }

   if (peer) {
      /* Set Frame packetization */
      if (p->rtp) {
         ast_rtp_codec_setpref(p->rtp, &peer->prefs);
         p->autoframing = peer->autoframing;
      }
      if (!peer->host_dynamic) {
         ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
         res = AUTH_PEER_NOT_DYNAMIC;
      } else {
         ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT);
         if (ast_test_flag(&p->flags[1], SIP_PAGE2_REGISTERTRYING))
            transmit_response(p, "100 Trying", req);
         if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, XMIT_UNRELIABLE, req->ignore))) {
            if (sip_cancel_destroy(p))
               ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");

            if (check_request_transport(peer, req)) {
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
               transmit_response_with_date(p, "403 Forbidden", req);
               res = AUTH_BAD_TRANSPORT;
            } else {

               /* We have a successful registration attempt with proper authentication,
                  now, update the peer */
               switch (parse_register_contact(p, peer, req)) {
               case PARSE_REGISTER_FAILED:
                  ast_log(LOG_WARNING, "Failed to parse contact info\n");
                  transmit_response_with_date(p, "400 Bad Request", req);
                  peer->lastmsgssent = -1;
                  res = 0;
                  break;
               case PARSE_REGISTER_QUERY:
                  transmit_response_with_date(p, "200 OK", req);
                  peer->lastmsgssent = -1;
                  res = 0;
                  break;
               case PARSE_REGISTER_UPDATE:
                  update_peer(peer, p->expiry);
                  /* Say OK and ask subsystem to retransmit msg counter */
                  transmit_response_with_date(p, "200 OK", req);
                  if (!ast_test_flag((&peer->flags[1]), SIP_PAGE2_SUBSCRIBEMWIONLY))
                     peer->lastmsgssent = -1;
                  res = 0;
                  break;
               }
            }

         } 
      }
   }
   if (!peer && autocreatepeer) {
      /* Create peer if we have autocreate mode enabled */
      peer = temp_peer(name);
      if (peer) {
         ao2_t_link(peers, peer, "link peer into peer table");
         if (peer->addr.sin_addr.s_addr) {
            ao2_t_link(peers_by_ip, peer, "link peer into peers-by-ip table");
         }
         
         if (sip_cancel_destroy(p))
            ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
         switch (parse_register_contact(p, peer, req)) {
         case PARSE_REGISTER_FAILED:
            ast_log(LOG_WARNING, "Failed to parse contact info\n");
            transmit_response_with_date(p, "400 Bad Request", req);
            peer->lastmsgssent = -1;
            res = 0;
            break;
         case PARSE_REGISTER_QUERY:
            transmit_response_with_date(p, "200 OK", req);
            peer->lastmsgssent = -1;
            res = 0;
            break;
         case PARSE_REGISTER_UPDATE:
            /* Say OK and ask subsystem to retransmit msg counter */
            transmit_response_with_date(p, "200 OK", req);
            manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\nPort: %d\r\n", peer->name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
            peer->lastmsgssent = -1;
            res = 0;
            break;
         }
      }
   }
   if (!peer && global_alwaysauthreject) {
      /* If we found a peer, we transmit a 100 Trying.  Therefore, if we're
       * trying to avoid leaking information, we MUST also transmit the same
       * response when we DON'T find a peer. */
      transmit_response(p, "100 Trying", req);
      /* Insert a fake delay between the 100 and the subsequent failure. */
      sched_yield();
   }
   if (!res) {
      ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
   }
   if (res < 0) {
      switch (res) {
      case AUTH_SECRET_FAILED:
         /* Wrong password in authentication. Go away, don't try again until you fixed it */
         transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq);
         if (global_authfailureevents)
            manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: AUTH_SECRET_FAILED\r\nAddress: %s\r\nPort: %d\r\n", 
               name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
         break;
      case AUTH_USERNAME_MISMATCH:
         /* Username and digest username does not match.
            Asterisk uses the From: username for authentication. We need the
            devices to use the same authentication user name until we support
            proper authentication by digest auth name */
         transmit_response(p, "403 Authentication user name does not match account name", &p->initreq);
         if (global_authfailureevents)
            manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: AUTH_USERNAME_MISMATCH\r\nAddress: %s\r\nPort: %d\r\n", 
               name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
         break;
      case AUTH_NOT_FOUND:
      case AUTH_PEER_NOT_DYNAMIC:
      case AUTH_ACL_FAILED:
         if (global_alwaysauthreject) {
            transmit_fake_auth_response(p, SIP_REGISTER, &p->initreq, XMIT_UNRELIABLE);
         } else {
            /* URI not found */
            if (res == AUTH_PEER_NOT_DYNAMIC) {
               transmit_response(p, "403 Forbidden", &p->initreq);
               if (global_authfailureevents)
                  manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: AUTH_PEER_NOT_DYNAMIC\r\nAddress: %s\r\nPort: %d\r\n", 
                     name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
               }
            else
               transmit_response(p, "404 Not found", &p->initreq);
               if (global_authfailureevents)
                  manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: URI_NOT_FOUND\r\nAddress: %s\r\nPort: %d\r\n", 
                     name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
         }
         break;
      case AUTH_BAD_TRANSPORT:
      default:
         break;
      }
   }
   if (peer)
      unref_peer(peer, "register_verify: unref_peer: tossing stack peer pointer at end of func");

   return res;
}

static struct sip_registry* registry_addref	(	struct sip_registry *	reg,
		char *	tag
	)			[static]

Add object reference to SIP registry.

Definition at line 2603 of file chan_sip.c.

References ast_debug, ASTOBJ_REF, sip_registry::hostname, and sip_registry::refcount.

Referenced by handle_response_register(), sip_send_all_registers(), and transmit_register().

{
   ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1);
   return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
}

static void * registry_unref	(	struct sip_registry *	reg,
		char *	tag
	)			[static]

Definition at line 2595 of file chan_sip.c.

References ast_debug, ASTOBJ_UNREF, sip_registry::hostname, sip_registry::refcount, and sip_registry_destroy().

Referenced by __sip_destroy(), dialog_unlink_all(), handle_response_register(), sip_reg_timeout(), sip_register(), sip_registry_destroy(), sip_reregister(), sip_send_all_registers(), and transmit_register().

{
   ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
   ASTOBJ_UNREF(reg, sip_registry_destroy);
   return NULL;
}

static const char * regstate2str

(

enum sipregistrystate

regstate

)

const [static]

Convert registration state status to string.

Definition at line 9858 of file chan_sip.c.

References map_x_s(), and regstatestrings.

{
   return map_x_s(regstatestrings, regstate, "Unknown");
}

static int reload

(

void

)

[static]

Part of Asterisk module interface.

Definition at line 23351 of file chan_sip.c.

References sip_reload().

{
   if (sip_reload(0, 0, NULL))
      return 0;
   return 1;
}

static int reload_config

(

enum channelreloadreason

reason

)

[static]

Re-read SIP.conf config file.

Note:

This function reloads all config data, except for active peers (with registrations). They will only change configuration data at restart, not at reload. SIP debug and recordhistory state will not change

< Type of address: IPv4

< Don't force proxy usage, use route: headers

< Keep track of hold status for a peer

< Match auth username if available instead of From: Default off.

< Default DTMF setting: RFC2833

< NAT support if requested by device with rport

< Allow re-invites

Definition at line 22051 of file chan_sip.c.

References ao2_t_callback, ast_clear_flag, ast_config_load, ast_debug, ast_free_ha(), ast_log(), AST_MAX_CONTEXT, ast_unload_realtime(), ASTOBJ_CONTAINER_DESTROYALL, ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_RDLOCK, ASTOBJ_UNLOCK, authl, CHANNEL_MODULE_LOAD, clear_realm_authentication(), clear_sip_domains(), CONFIG_FLAG_FILEUNCHANGED, config_flags, CONFIG_STATUS_FILEUNCHANGED, context, default_tls_cfg, dialog_unlink_all(), dialog_unref(), dummy(), ast_tls_config::enabled, FALSE, global_contact_ha, ast_tcptls_session_args::local_address, LOG_NOTICE, OBJ_NODATA, peer_markall_func(), regl, sip_registry_destroy(), sip_tcp_desc, sip_tls_desc, STANDARD_SIP_PORT, STANDARD_TLS_PORT, and TRUE.

{
   struct ast_config *cfg, *ucfg;
   struct ast_variable *v;
   struct sip_peer *peer;
   char *cat, *stringp, *context, *oldregcontext;
   char newcontexts[AST_MAX_CONTEXT], oldcontexts[AST_MAX_CONTEXT];
   struct ast_flags dummy[2];
   struct ast_flags config_flags = { reason == CHANNEL_MODULE_LOAD ? 0 : CONFIG_FLAG_FILEUNCHANGED };
   int auto_sip_domains = FALSE;
   struct sockaddr_in old_bindaddr = bindaddr;
   int registry_count = 0, peer_count = 0;
   time_t run_start, run_end;
   
   run_start = time(0);
   ast_unload_realtime("sipregs");     
   ast_unload_realtime("sippeers");
   cfg = ast_config_load(config, config_flags);

   /* We *must* have a config file otherwise stop immediately */
   if (!cfg) {
      ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
      return -1;
   } else if (cfg == CONFIG_STATUS_FILEUNCHANGED) {
      ucfg = ast_config_load("users.conf", config_flags);
      if (ucfg == CONFIG_STATUS_FILEUNCHANGED)
         return 1;
      /* Must reread both files, because one changed */
      ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
      cfg = ast_config_load(config, config_flags);
   } else {
      ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
      ucfg = ast_config_load("users.conf", config_flags);
   }

   /* Initialize tcp sockets */
   memset(&sip_tcp_desc.local_address, 0, sizeof(sip_tcp_desc.local_address));
   memset(&sip_tls_desc.local_address, 0, sizeof(sip_tls_desc.local_address));

   ast_free_ha(global_contact_ha);
   global_contact_ha = NULL;

   default_tls_cfg.enabled = FALSE;    /* Default: Disable TLS */

   sip_tcp_desc.local_address.sin_port = htons(STANDARD_SIP_PORT);
   sip_tls_desc.local_address.sin_port = htons(STANDARD_TLS_PORT);

   if (reason != CHANNEL_MODULE_LOAD) {
      ast_debug(4, "--------------- SIP reload started\n");

      clear_realm_authentication(authl);
      clear_sip_domains();
      authl = NULL;

      /* First, destroy all outstanding registry calls */
      /* This is needed, since otherwise active registry entries will not be destroyed */
      ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {  /* regl is locked */
            
            /* avoid a deadlock in the unlink_all call, if iterator->call's (a dialog) registry entry
               is this registry entry. In other words, if the dialog we are pointing to points back to 
               us, then if we get a lock on this object, and try to UNREF it, we will deadlock, because
               we already ... NO. This is not the problem. */
            ASTOBJ_RDLOCK(iterator); /* now regl is locked, and the object is also locked */
            if (iterator->call) {
               ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", iterator->username, iterator->hostname);
               /* This will also remove references to the registry */
               dialog_unlink_all(iterator->call, TRUE, TRUE);
               iterator->call = dialog_unref(iterator->call, "remove iterator->call from registry traversal");
               /* iterator->call = sip_destroy(iterator->call); */
            }
            ASTOBJ_UNLOCK(iterator);
            
      } while(0));

      /* Then, actually destroy users and registry */
      ASTOBJ_CONTAINER_DESTROYALL(&regl, sip_registry_destroy);
      ast_debug(4, "--------------- Done destroying registry list\n");
      ao2_t_callback(peers, OBJ_NODATA, peer_markall_func, 0, "callback to mark all peers");
   }
   
   /* Reset certificate handling for TLS sessions */
   if (reason != CHANNEL_MODULE_LOAD) {
      ast_free(default_tls_cfg.certfile);
      ast_free(default_tls_cfg.cipher);
      ast_free(default_tls_cfg.cafile);
      ast_free(default_tls_cfg.capath);
   }
   default_tls_cfg.certfile = ast_strdup(AST_CERTFILE); /*XXX Not sure if this is useful */
   default_tls_cfg.cipher = ast_strdup("");
   default_tls_cfg.cafile = ast_strdup("");
   default_tls_cfg.capath = ast_strdup("");
   
   /* Initialize copy of current global_regcontext for later use in removing stale contexts */
   ast_copy_string(oldcontexts, global_regcontext, sizeof(oldcontexts));
   oldregcontext = oldcontexts;

   /* Clear all flags before setting default values */
   /* Preserve debugging settings for console */
   sipdebug &= sip_debug_console;
   ast_clear_flag(&global_flags[0], AST_FLAGS_ALL);
   ast_clear_flag(&global_flags[1], AST_FLAGS_ALL);

   /* Reset IP addresses  */
   memset(&bindaddr, 0, sizeof(bindaddr));
   memset(&stunaddr, 0, sizeof(stunaddr));
   memset(&internip, 0, sizeof(internip));

   /* Free memory for local network address mask */
   ast_free_ha(localaddr);
   memset(&localaddr, 0, sizeof(localaddr));
   memset(&externip, 0, sizeof(externip));
   memset(&default_prefs, 0 , sizeof(default_prefs));
   memset(&global_outboundproxy, 0, sizeof(struct sip_proxy));
   global_outboundproxy.ip.sin_port = htons(STANDARD_SIP_PORT);
   global_outboundproxy.ip.sin_family = AF_INET;   /*!< Type of address: IPv4 */
   global_outboundproxy.force = FALSE;    /*!< Don't force proxy usage, use route: headers */
   ourport_tcp = STANDARD_SIP_PORT;
   ourport_tls = STANDARD_TLS_PORT;
   bindaddr.sin_port = htons(STANDARD_SIP_PORT);
   global_srvlookup = DEFAULT_SRVLOOKUP;
   global_tos_sip = DEFAULT_TOS_SIP;
   global_tos_audio = DEFAULT_TOS_AUDIO;
   global_tos_video = DEFAULT_TOS_VIDEO;
   global_tos_text = DEFAULT_TOS_TEXT;
   global_cos_sip = DEFAULT_COS_SIP;
   global_cos_audio = DEFAULT_COS_AUDIO;
   global_cos_video = DEFAULT_COS_VIDEO;
   global_cos_text = DEFAULT_COS_TEXT;

   externhost[0] = '\0';         /* External host name (for behind NAT DynDNS support) */
   externexpire = 0;       /* Expiration for DNS re-issuing */
   externrefresh = 10;

   /* Reset channel settings to default before re-configuring */
   allow_external_domains = DEFAULT_ALLOW_EXT_DOM;          /* Allow external invites */
   global_regcontext[0] = '\0';
   global_regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
   global_notifyringing = DEFAULT_NOTIFYRINGING;
   global_notifyhold = FALSE;    /*!< Keep track of hold status for a peer */
   global_directrtpsetup = FALSE;      /* Experimental feature, disabled by default */
   global_alwaysauthreject = 0;
   global_allowsubscribe = FALSE;
   snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ast_get_version());
   snprintf(global_sdpsession, sizeof(global_sdpsession), "%s %s", DEFAULT_SDPSESSION, ast_get_version());
   snprintf(global_sdpowner, sizeof(global_sdpowner), "%s", DEFAULT_SDPOWNER);
   ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
   ast_copy_string(global_realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(global_realm));
   ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
   compactheaders = DEFAULT_COMPACTHEADERS;
   global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
   global_regattempts_max = 0;
   pedanticsipchecking = DEFAULT_PEDANTIC;
   autocreatepeer = DEFAULT_AUTOCREATEPEER;
   global_autoframing = 0;
   global_allowguest = DEFAULT_ALLOWGUEST;
   global_callcounter = DEFAULT_CALLCOUNTER;
   global_match_auth_username = FALSE;    /*!< Match auth username if available instead of From: Default off. */
   global_rtptimeout = 0;
   global_rtpholdtimeout = 0;
   global_rtpkeepalive = 0;
   global_allowtransfer = TRANSFER_OPENFORALL;  /* Merrily accept all transfers by default */
   global_rtautoclear = 120;
   ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE);   /* Default for all devices: TRUE */
   ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP);     /* Default for all devices: TRUE */
   sip_cfg.peer_rtupdate = TRUE;

   /* Session-Timers */
   global_st_mode = SESSION_TIMER_MODE_ACCEPT;    
   global_st_refresher = SESSION_TIMER_REFRESHER_UAS;
   global_min_se  = DEFAULT_MIN_SE;
   global_max_se  = DEFAULT_MAX_SE;

   /* Initialize some reasonable defaults at SIP reload (used both for channel and as default for devices */
   ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context));
   default_subscribecontext[0] = '\0';
   default_language[0] = '\0';
   default_fromdomain[0] = '\0';
   default_qualify = DEFAULT_QUALIFY;
   default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
   ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
   ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
   ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
   ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833);        /*!< Default DTMF setting: RFC2833 */
   ast_set_flag(&global_flags[0], SIP_NAT_RFC3581);         /*!< NAT support if requested by device with rport */
   ast_set_flag(&global_flags[0], SIP_CAN_REINVITE);        /*!< Allow re-invites */

   /* Debugging settings, always default to off */
   dumphistory = FALSE;
   recordhistory = FALSE;
   sipdebug &= ~sip_debug_config;

   /* Misc settings for the channel */
   global_relaxdtmf = FALSE;
   global_callevents = FALSE;
   global_authfailureevents = FALSE;
   global_t1 = SIP_TIMER_T1;
   global_timer_b = 64 * SIP_TIMER_T1;
   global_t1min = DEFAULT_T1MIN;
   global_qualifyfreq = DEFAULT_QUALIFYFREQ;

   global_matchexterniplocally = FALSE;

   /* Copy the default jb config over global_jbconf */
   memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));

   ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
   ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);


   /* Read the [general] config section of sip.conf (or from realtime config) */
   for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
      if (handle_common_options(&global_flags[0], &dummy[0], v))
         continue;
      /* handle jb conf */
      if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
         continue;

      if (!strcasecmp(v->name, "context")) {
         ast_copy_string(default_context, v->value, sizeof(default_context));
      } else if (!strcasecmp(v->name, "subscribecontext")) {
         ast_copy_string(default_subscribecontext, v->value, sizeof(default_subscribecontext));
      } else if (!strcasecmp(v->name, "callcounter")) {
         global_callcounter = ast_true(v->value) ? 1 : 0;
      } else if (!strcasecmp(v->name, "allowguest")) {
         global_allowguest = ast_true(v->value) ? 1 : 0;
      } else if (!strcasecmp(v->name, "realm")) {
         ast_copy_string(global_realm, v->value, sizeof(global_realm));
      } else if (!strcasecmp(v->name, "useragent")) {
         ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
         ast_debug(1, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
      } else if (!strcasecmp(v->name, "sdpsession")) {
         ast_copy_string(global_sdpsession, v->value, sizeof(global_sdpsession));
      } else if (!strcasecmp(v->name, "sdpowner")) {
         /* Field cannot contain spaces */
         if (!strstr(v->value, " "))
            ast_copy_string(global_sdpowner, v->value, sizeof(global_sdpowner));
         else
            ast_log(LOG_WARNING, "'%s' must not contain spaces at line %d.  Using default.\n", v->value, v->lineno);
      } else if (!strcasecmp(v->name, "allowtransfer")) {
         global_allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
      } else if (!strcasecmp(v->name, "rtcachefriends")) {
         ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS);   
      } else if (!strcasecmp(v->name, "rtsavesysname")) {
         sip_cfg.rtsave_sysname = ast_true(v->value);
      } else if (!strcasecmp(v->name, "rtupdate")) {
         sip_cfg.peer_rtupdate = ast_true(v->value);
      } else if (!strcasecmp(v->name, "ignoreregexpire")) {
         sip_cfg.ignore_regexpire = ast_true(v->value);
      } else if (!strcasecmp(v->name, "timert1")) {
         /* Defaults to 500ms, but RFC 3261 states that it is recommended
          * for the value to be set higher, though a lower value is only
          * allowed on private networks unconnected to the Internet. */
         global_t1 = atoi(v->value);
         /* Note that timer B is dependent on the value of T1 */
         global_timer_b = global_t1 * 64;
      } else if (!strcasecmp(v->name, "t1min")) {
         global_t1min = atoi(v->value);
      } else if (!strcasecmp(v->name, "tcpenable")) {
         sip_tcp_desc.local_address.sin_family = ast_false(v->value) ? 0 : AF_INET;
         ast_debug(2, "Enabling TCP socket for listening\n");
      } else if (!strcasecmp(v->name, "tcpbindaddr")) {
         int family = sip_tcp_desc.local_address.sin_family;
         if (ast_parse_arg(v->value, PARSE_INADDR, &sip_tcp_desc.local_address))
            ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n", v->name, v->value, v->lineno, config);
         sip_tcp_desc.local_address.sin_family = family;
         ast_debug(2, "Setting TCP socket address to %s\n", v->value);
      } else if (!strcasecmp(v->name, "tlsenable")) {
         default_tls_cfg.enabled = ast_true(v->value) ? TRUE : FALSE;
         sip_tls_desc.local_address.sin_family = AF_INET;
      } else if (!strcasecmp(v->name, "tlscertfile")) {
         ast_free(default_tls_cfg.certfile);
         default_tls_cfg.certfile = ast_strdup(v->value);
      } else if (!strcasecmp(v->name, "tlscipher")) {
         ast_free(default_tls_cfg.cipher);
         default_tls_cfg.cipher = ast_strdup(v->value);
      } else if (!strcasecmp(v->name, "tlscafile")) {
         ast_free(default_tls_cfg.cafile);
         default_tls_cfg.cafile = ast_strdup(v->value);
      } else if (!strcasecmp(v->name, "tlscapath")) {
         ast_free(default_tls_cfg.capath);
         default_tls_cfg.capath = ast_strdup(v->value);
      } else if (!strcasecmp(v->name, "tlsverifyclient")) {
         ast_set2_flag(&default_tls_cfg.flags, ast_true(v->value), AST_SSL_VERIFY_CLIENT);   
      } else if (!strcasecmp(v->name, "tlsdontverifyserver")) {
         ast_set2_flag(&default_tls_cfg.flags, ast_true(v->value), AST_SSL_DONT_VERIFY_SERVER); 
      } else if (!strcasecmp(v->name, "tlsbindaddr")) {
         if (ast_parse_arg(v->value, PARSE_INADDR, &sip_tls_desc.local_address))
            ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n", v->name, v->value, v->lineno, config);
      } else if (!strcasecmp(v->name, "dynamic_exclude_static") || !strcasecmp(v->name, "dynamic_excludes_static")) {
         global_dynamic_exclude_static = ast_true(v->value);
      } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) {
         int ha_error = 0;
         global_contact_ha = ast_append_ha(v->name + 7, v->value, global_contact_ha, &ha_error);
         if (ha_error) {
            ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
         }
      } else if (!strcasecmp(v->name, "rtautoclear")) {
         int i = atoi(v->value);
         if (i > 0)
            global_rtautoclear = i;
         else
            i = 0;
         ast_set2_flag(&global_flags[1], i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR);
      } else if (!strcasecmp(v->name, "usereqphone")) {
         ast_set2_flag(&global_flags[0], ast_true(v->value), SIP_USEREQPHONE);   
      } else if (!strcasecmp(v->name, "relaxdtmf")) {
         global_relaxdtmf = ast_true(v->value);
      } else if (!strcasecmp(v->name, "vmexten")) {
         ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
      } else if (!strcasecmp(v->name, "rtptimeout")) {
         if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
            global_rtptimeout = 0;
         }
      } else if (!strcasecmp(v->name, "rtpholdtimeout")) {
         if ((sscanf(v->value, "%d", &global_rtpholdtimeout) != 1) || (global_rtpholdtimeout < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
            global_rtpholdtimeout = 0;
         }
      } else if (!strcasecmp(v->name, "rtpkeepalive")) {
         if ((sscanf(v->value, "%d", &global_rtpkeepalive) != 1) || (global_rtpkeepalive < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d.  Using default.\n", v->value, v->lineno);
            global_rtpkeepalive = 0;
         }
      } else if (!strcasecmp(v->name, "compactheaders")) {
         compactheaders = ast_true(v->value);
      } else if (!strcasecmp(v->name, "notifymimetype")) {
         ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
      } else if (!strcasecmp(v->name, "directrtpsetup")) {
         global_directrtpsetup = ast_true(v->value);
      } else if (!strcasecmp(v->name, "notifyringing")) {
         global_notifyringing = ast_true(v->value);
      } else if (!strcasecmp(v->name, "notifyhold")) {
         global_notifyhold = ast_true(v->value);
      } else if (!strcasecmp(v->name, "alwaysauthreject")) {
         global_alwaysauthreject = ast_true(v->value);
      } else if (!strcasecmp(v->name, "mohinterpret")) {
         ast_copy_string(default_mohinterpret, v->value, sizeof(default_mohinterpret));
      } else if (!strcasecmp(v->name, "mohsuggest")) {
         ast_copy_string(default_mohsuggest, v->value, sizeof(default_mohsuggest));
      } else if (!strcasecmp(v->name, "language")) {
         ast_copy_string(default_language, v->value, sizeof(default_language));
      } else if (!strcasecmp(v->name, "regcontext")) {
         ast_copy_string(newcontexts, v->value, sizeof(newcontexts));
         stringp = newcontexts;
         /* Let's remove any contexts that are no longer defined in regcontext */
         cleanup_stale_contexts(stringp, oldregcontext);
         /* Create contexts if they don't exist already */
         while ((context = strsep(&stringp, "&"))) {
            ast_copy_string(used_context, context, sizeof(used_context));
            ast_context_find_or_create(NULL, NULL, context, "SIP");
         }
         ast_copy_string(global_regcontext, v->value, sizeof(global_regcontext));
      } else if (!strcasecmp(v->name, "regextenonqualify")) {
         global_regextenonqualify = ast_true(v->value);
      } else if (!strcasecmp(v->name, "callerid")) {
         ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
      } else if (!strcasecmp(v->name, "fromdomain")) {
         ast_copy_string(default_fromdomain, v->value, sizeof(default_fromdomain));
      } else if (!strcasecmp(v->name, "outboundproxy")) {
         int portnum;
         char *tok, *proxyname;

         if (ast_strlen_zero(v->value)) {
            ast_log(LOG_WARNING, "no value given for outbound proxy on line %d of sip.conf.", v->lineno);
            continue;
         }

         tok = ast_skip_blanks(strtok(ast_strdupa(v->value), ","));

         sip_parse_host(tok, v->lineno, &proxyname, &portnum, &global_outboundproxy.transport);

         global_outboundproxy.ip.sin_port = htons(portnum);
   
         if ((tok = strtok(NULL, ","))) {
            global_outboundproxy.force = !strncasecmp(ast_skip_blanks(tok), "force", 5);
         } else {
            global_outboundproxy.force = FALSE;
         }

         if (ast_strlen_zero(proxyname)) {
            ast_log(LOG_WARNING, "you must specify a name for the outboundproxy on line %d of sip.conf.", v->lineno);
            global_outboundproxy.name[0] = '\0';
            continue;
         }

         ast_copy_string(global_outboundproxy.name, proxyname, sizeof(global_outboundproxy.name));

         proxy_update(&global_outboundproxy);
      } else if (!strcasecmp(v->name, "autocreatepeer")) {
         autocreatepeer = ast_true(v->value);
      } else if (!strcasecmp(v->name, "match_auth_username")) {
         global_match_auth_username = ast_true(v->value);
      } else if (!strcasecmp(v->name, "srvlookup")) {
         global_srvlookup = ast_true(v->value);
      } else if (!strcasecmp(v->name, "pedantic")) {
         pedanticsipchecking = ast_true(v->value);
      } else if (!strcasecmp(v->name, "maxexpirey") || !strcasecmp(v->name, "maxexpiry")) {
         max_expiry = atoi(v->value);
         if (max_expiry < 1)
            max_expiry = DEFAULT_MAX_EXPIRY;
      } else if (!strcasecmp(v->name, "minexpirey") || !strcasecmp(v->name, "minexpiry")) {
         min_expiry = atoi(v->value);
         if (min_expiry < 1)
            min_expiry = DEFAULT_MIN_EXPIRY;
      } else if (!strcasecmp(v->name, "defaultexpiry") || !strcasecmp(v->name, "defaultexpirey")) {
         default_expiry = atoi(v->value);
         if (default_expiry < 1)
            default_expiry = DEFAULT_DEFAULT_EXPIRY;
      } else if (!strcasecmp(v->name, "sipdebug")) {
         if (ast_true(v->value))
            sipdebug |= sip_debug_config;
      } else if (!strcasecmp(v->name, "dumphistory")) {
         dumphistory = ast_true(v->value);
      } else if (!strcasecmp(v->name, "recordhistory")) {
         recordhistory = ast_true(v->value);
      } else if (!strcasecmp(v->name, "registertimeout")) {
         global_reg_timeout = atoi(v->value);
         if (global_reg_timeout < 1)
            global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
      } else if (!strcasecmp(v->name, "registerattempts")) {
         global_regattempts_max = atoi(v->value);
      } else if (!strcasecmp(v->name, "stunaddr")) {
         stunaddr.sin_port = htons(3478);
         if (ast_parse_arg(v->value, PARSE_INADDR, &stunaddr))
            ast_log(LOG_WARNING, "Invalid STUN server address: %s\n", v->value);
         externexpire = time(NULL);
      } else if (!strcasecmp(v->name, "bindaddr") || !strcasecmp(v->name, "udpbindaddr")) {
         if (ast_parse_arg(v->value, PARSE_INADDR, &bindaddr))
            ast_log(LOG_WARNING, "Invalid address: %s\n", v->value);
      } else if (!strcasecmp(v->name, "localnet")) {
         struct ast_ha *na;
         int ha_error = 0;

         if (!(na = ast_append_ha("d", v->value, localaddr, &ha_error)))
            ast_log(LOG_WARNING, "Invalid localnet value: %s\n", v->value);
         else
            localaddr = na;
         if (ha_error)
            ast_log(LOG_ERROR, "Bad localnet configuration value line %d : %s\n", v->lineno, v->value);
      } else if (!strcasecmp(v->name, "externip")) {
         if (ast_parse_arg(v->value, PARSE_INADDR, &externip))
            ast_log(LOG_WARNING, "Invalid address for externip keyword: %s\n", v->value);
         externexpire = 0;
         /* If no port was specified use the value of bindport */
         if (!externip.sin_port)
            externip.sin_port = bindaddr.sin_port;
      } else if (!strcasecmp(v->name, "externhost")) {
         ast_copy_string(externhost, v->value, sizeof(externhost));
         if (ast_parse_arg(externhost, PARSE_INADDR, &externip))
            ast_log(LOG_WARNING, "Invalid address for externhost keyword: %s\n", externhost);
         externexpire = time(NULL);
         /* If no port was specified use the value of bindport */
         if (!externip.sin_port)
            externip.sin_port = bindaddr.sin_port;
      } else if (!strcasecmp(v->name, "externrefresh")) {
         if (sscanf(v->value, "%d", &externrefresh) != 1) {
            ast_log(LOG_WARNING, "Invalid externrefresh value '%s', must be an integer >0 at line %d\n", v->value, v->lineno);
            externrefresh = 10;
         }
      } else if (!strcasecmp(v->name, "allow")) {
         int error =  ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, TRUE);
         if (error)
            ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
      } else if (!strcasecmp(v->name, "disallow")) {
         int error =  ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, FALSE);
         if (error)
            ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
      } else if (!strcasecmp(v->name, "autoframing")) {
         global_autoframing = ast_true(v->value);
      } else if (!strcasecmp(v->name, "allowexternaldomains")) {
         allow_external_domains = ast_true(v->value);
      } else if (!strcasecmp(v->name, "autodomain")) {
         auto_sip_domains = ast_true(v->value);
      } else if (!strcasecmp(v->name, "domain")) {
         char *domain = ast_strdupa(v->value);
         char *cntx = strchr(domain, ',');

         if (cntx)
            *cntx++ = '\0';

         if (ast_strlen_zero(cntx))
            ast_debug(1, "No context specified at line %d for domain '%s'\n", v->lineno, domain);
         if (ast_strlen_zero(domain))
            ast_log(LOG_WARNING, "Empty domain specified at line %d\n", v->lineno);
         else
            add_sip_domain(ast_strip(domain), SIP_DOMAIN_CONFIG, cntx ? ast_strip(cntx) : "");
      } else if (!strcasecmp(v->name, "register")) {
         if (sip_register(v->value, v->lineno) == 0)
            registry_count++;
      } else if (!strcasecmp(v->name, "tos_sip")) {
         if (ast_str2tos(v->value, &global_tos_sip))
            ast_log(LOG_WARNING, "Invalid tos_sip value at line %d, refer to QoS documentation\n", v->lineno);
      } else if (!strcasecmp(v->name, "tos_audio")) {
         if (ast_str2tos(v->value, &global_tos_audio))
            ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
      } else if (!strcasecmp(v->name, "tos_video")) {
         if (ast_str2tos(v->value, &global_tos_video))
            ast_log(LOG_WARNING, "Invalid tos_video value at line %d, refer to QoS documentation\n", v->lineno);
      } else if (!strcasecmp(v->name, "tos_text")) {
         if (ast_str2tos(v->value, &global_tos_text))
            ast_log(LOG_WARNING, "Invalid tos_text value at line %d, refer to QoS documentation\n", v->lineno);
      } else if (!strcasecmp(v->name, "cos_sip")) {
         if (ast_str2cos(v->value, &global_cos_sip))
            ast_log(LOG_WARNING, "Invalid cos_sip value at line %d, refer to QoS documentation\n", v->lineno);
      } else if (!strcasecmp(v->name, "cos_audio")) {
         if (ast_str2cos(v->value, &global_cos_audio))
            ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
      } else if (!strcasecmp(v->name, "cos_video")) {
         if (ast_str2cos(v->value, &global_cos_video))
            ast_log(LOG_WARNING, "Invalid cos_video value at line %d, refer to QoS documentation\n", v->lineno);
      } else if (!strcasecmp(v->name, "cos_text")) {
         if (ast_str2cos(v->value, &global_cos_text))
            ast_log(LOG_WARNING, "Invalid cos_text value at line %d, refer to QoS documentation\n", v->lineno);
      } else if (!strcasecmp(v->name, "bindport")) {
         int i;
         if (sscanf(v->value, "%d", &i) == 1) {
            bindaddr.sin_port = htons(i);
         } else {
            ast_log(LOG_WARNING, "Invalid port number '%s' at line %d of %s\n", v->value, v->lineno, config);
         }
      } else if (!strcasecmp(v->name, "hash_user")) {
         int i;
         if (sscanf(v->value, "%d", &i) == 1 && i > 2) {
            hash_user_size = i;
         } else {
            ast_log(LOG_WARNING, "Invalid hash_user size '%s' at line %d of %s -- should be much larger than 2\n", v->value, v->lineno, config);
         }
      } else if (!strcasecmp(v->name, "hash_peer")) {
         int i;
         if (sscanf(v->value, "%d", &i) == 1 && i > 2) {
            hash_peer_size = i;
         } else {
            ast_log(LOG_WARNING, "Invalid hash_peer size '%s' at line %d of %s -- should be much larger than 2\n", v->value, v->lineno, config);
         }
      } else if (!strcasecmp(v->name, "hash_dialog")) {
         int i;
         if (sscanf(v->value, "%d", &i) == 1 && i > 2) {
            hash_dialog_size = i;
         } else {
            ast_log(LOG_WARNING, "Invalid hash_dialog size '%s' at line %d of %s -- should be much larger than 2\n", v->value, v->lineno, config);
         }
      } else if (!strcasecmp(v->name, "qualify")) {
         if (!strcasecmp(v->value, "no")) {
            default_qualify = 0;
         } else if (!strcasecmp(v->value, "yes")) {
            default_qualify = DEFAULT_MAXMS;
         } else if (sscanf(v->value, "%d", &default_qualify) != 1) {
            ast_log(LOG_WARNING, "Qualification default should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", v->lineno);
            default_qualify = 0;
         }
      } else if (!strcasecmp(v->name, "qualifyfreq")) {
         int i;
         if (sscanf(v->value, "%d", &i) == 1)
            global_qualifyfreq = i * 1000;
         else {
            ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_qualifyfreq = DEFAULT_QUALIFYFREQ;
         }
      } else if (!strcasecmp(v->name, "callevents")) {
         global_callevents = ast_true(v->value);
      } else if (!strcasecmp(v->name, "authfailureevents")) {
         global_authfailureevents = ast_true(v->value);
      } else if (!strcasecmp(v->name, "maxcallbitrate")) {
         default_maxcallbitrate = atoi(v->value);
         if (default_maxcallbitrate < 0)
            default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
      } else if (!strcasecmp(v->name, "matchexterniplocally")) {
         global_matchexterniplocally = ast_true(v->value);
      } else if (!strcasecmp(v->name, "session-timers")) {
         int i = (int) str2stmode(v->value); 
         if (i < 0) {
            ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_st_mode = SESSION_TIMER_MODE_ACCEPT;
         } else {
            global_st_mode = i;
         }
      } else if (!strcasecmp(v->name, "session-expires")) {
         if (sscanf(v->value, "%d", &global_max_se) != 1) {
            ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_max_se = DEFAULT_MAX_SE;
         } 
      } else if (!strcasecmp(v->name, "session-minse")) {
         if (sscanf(v->value, "%d", &global_min_se) != 1) {
            ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_min_se = DEFAULT_MIN_SE;
         } 
         if (global_min_se < 90) {
            ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < 90 secs\n", v->value, v->lineno, config);
            global_min_se = DEFAULT_MIN_SE;
         } 
      } else if (!strcasecmp(v->name, "session-refresher")) {
         int i = (int) str2strefresher(v->value); 
         if (i < 0) {
            ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_st_refresher = SESSION_TIMER_REFRESHER_UAS;
         } else {
            global_st_refresher = i;
         }
      }
   }

   if (!allow_external_domains && AST_LIST_EMPTY(&domain_list)) {
      ast_log(LOG_WARNING, "To disallow external domains, you need to configure local SIP domains.\n");
      allow_external_domains = 1;
   }
   
   /* Build list of authentication to various SIP realms, i.e. service providers */
   for (v = ast_variable_browse(cfg, "authentication"); v ; v = v->next) {
      /* Format for authentication is auth = username:password@realm */
      if (!strcasecmp(v->name, "auth"))
         authl = add_realm_authentication(authl, v->value, v->lineno);
   }
   
   if (ucfg) {
      struct ast_variable *gen;
      int genhassip, genregistersip;
      const char *hassip, *registersip;
      
      genhassip = ast_true(ast_variable_retrieve(ucfg, "general", "hassip"));
      genregistersip = ast_true(ast_variable_retrieve(ucfg, "general", "registersip"));
      gen = ast_variable_browse(ucfg, "general");
      cat = ast_category_browse(ucfg, NULL);
      while (cat) {
         if (strcasecmp(cat, "general")) {
            hassip = ast_variable_retrieve(ucfg, cat, "hassip");
            registersip = ast_variable_retrieve(ucfg, cat, "registersip");
            if (ast_true(hassip) || (!hassip && genhassip)) {
               peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0);
               if (peer) {
                  ao2_t_link(peers, peer, "link peer into peer table");
                  if ((peer->type & SIP_TYPE_PEER) && peer->addr.sin_addr.s_addr) {
                     ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
                  }
                  
                  unref_peer(peer, "unref_peer: from reload_config");
                  peer_count++;
               }
            }
            if (ast_true(registersip) || (!registersip && genregistersip)) {
               char tmp[256];
               const char *host = ast_variable_retrieve(ucfg, cat, "host");
               const char *username = ast_variable_retrieve(ucfg, cat, "username");
               const char *secret = ast_variable_retrieve(ucfg, cat, "secret");
               const char *contact = ast_variable_retrieve(ucfg, cat, "contact");
               if (!host)
                  host = ast_variable_retrieve(ucfg, "general", "host");
               if (!username)
                  username = ast_variable_retrieve(ucfg, "general", "username");
               if (!secret)
                  secret = ast_variable_retrieve(ucfg, "general", "secret");
               if (!contact)
                  contact = "s";
               if (!ast_strlen_zero(username) && !ast_strlen_zero(host)) {
                  if (!ast_strlen_zero(secret))
                     snprintf(tmp, sizeof(tmp), "%s:%s@%s/%s", username, secret, host, contact);
                  else
                     snprintf(tmp, sizeof(tmp), "%s@%s/%s", username, host, contact);
                  if (sip_register(tmp, 0) == 0)
                     registry_count++;
               }
            }
         }
         cat = ast_category_browse(ucfg, cat);
      }
      ast_config_destroy(ucfg);
   }
   

   /* Load peers, users and friends */
   cat = NULL;
   while ( (cat = ast_category_browse(cfg, cat)) ) {
      const char *utype;
      if (!strcasecmp(cat, "general") || !strcasecmp(cat, "authentication"))
         continue;
      utype = ast_variable_retrieve(cfg, cat, "type");
      if (!utype) {
         ast_log(LOG_WARNING, "Section '%s' lacks type\n", cat);
         continue;
      } else {
         if (!strcasecmp(utype, "user")) {
            ;
         } else if (!strcasecmp(utype, "friend")) {
            ;
         } else if (!strcasecmp(utype, "peer")) {
            ;
         } else {
            ast_log(LOG_WARNING, "Unknown type '%s' for '%s' in %s\n", utype, cat, "sip.conf");
            continue;
         }
         peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0);
         if (peer) {
            ao2_t_link(peers, peer, "link peer into peers table");
            if ((peer->type & SIP_TYPE_PEER) && peer->addr.sin_addr.s_addr) {
               ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
            }
            unref_peer(peer, "unref the result of the build_peer call. Now, the links from the tables are the only ones left.");
            peer_count++;
         }
      }
   }
   
   /* Set UDP address and open socket */
   bindaddr.sin_family = AF_INET;
   internip = bindaddr;
   if (ast_find_ourip(&internip.sin_addr, bindaddr)) {
      ast_log(LOG_WARNING, "Unable to get own IP address, SIP disabled\n");
      ast_config_destroy(cfg);
      return 0;
   }
   ast_mutex_lock(&netlock);
   if ((sipsock > -1) && (memcmp(&old_bindaddr, &bindaddr, sizeof(struct sockaddr_in)))) {
      close(sipsock);
      sipsock = -1;
   }
   if (sipsock < 0) {
      sipsock = socket(AF_INET, SOCK_DGRAM, 0);
      if (sipsock < 0) {
         ast_log(LOG_WARNING, "Unable to create SIP socket: %s\n", strerror(errno));
         ast_config_destroy(cfg);
         return -1;
      } else {
         /* Allow SIP clients on the same host to access us: */
         const int reuseFlag = 1;

         setsockopt(sipsock, SOL_SOCKET, SO_REUSEADDR,
               (const char*)&reuseFlag,
               sizeof reuseFlag);

         ast_enable_packet_fragmentation(sipsock);

         if (bind(sipsock, (struct sockaddr *)&bindaddr, sizeof(bindaddr)) < 0) {
            ast_log(LOG_WARNING, "Failed to bind to %s:%d: %s\n",
            ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port),
            strerror(errno));
            close(sipsock);
            sipsock = -1;
         } else {
            ast_verb(2, "SIP Listening on %s:%d\n",
                  ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port));
            ast_netsock_set_qos(sipsock, global_tos_sip, global_cos_sip, "SIP");
         }
      }
   }
   if (stunaddr.sin_addr.s_addr != 0) {
      ast_debug(1, "stun to %s:%d\n",
         ast_inet_ntoa(stunaddr.sin_addr) , ntohs(stunaddr.sin_port));
      ast_stun_request(sipsock, &stunaddr,
         NULL, &externip);
      ast_debug(1, "STUN sees us at %s:%d\n", 
         ast_inet_ntoa(externip.sin_addr) , ntohs(externip.sin_port));
   }
   ast_mutex_unlock(&netlock);

   /* Start TCP server */
   ast_tcptls_server_start(&sip_tcp_desc);

   /* Start TLS server if needed */
   memcpy(sip_tls_desc.tls_cfg, &default_tls_cfg, sizeof(default_tls_cfg));

   if (ast_ssl_setup(sip_tls_desc.tls_cfg))
      ast_tcptls_server_start(&sip_tls_desc);
   else if (sip_tls_desc.tls_cfg->enabled) {
      sip_tls_desc.tls_cfg = NULL;
      ast_log(LOG_WARNING, "SIP TLS server did not load because of errors.\n");
   }


   /* Add default domains - host name, IP address and IP:port
    * Only do this if user added any sip domain with "localdomains" 
    * In order to *not* break backwards compatibility 
    *    Some phones address us at IP only, some with additional port number 
    */
   if (auto_sip_domains) {
      char temp[MAXHOSTNAMELEN];

      /* First our default IP address */
      if (bindaddr.sin_addr.s_addr)
         add_sip_domain(ast_inet_ntoa(bindaddr.sin_addr), SIP_DOMAIN_AUTO, NULL);
      else
         ast_log(LOG_NOTICE, "Can't add wildcard IP address to domain list, please add IP address to domain manually.\n");

      /* If TCP is running on a different IP than UDP, then add it too */
      if (sip_tcp_desc.local_address.sin_addr.s_addr && !inaddrcmp(&bindaddr, &sip_tcp_desc.local_address))
         add_sip_domain(ast_inet_ntoa(sip_tcp_desc.local_address.sin_addr), SIP_DOMAIN_AUTO, NULL);

      /* If TLS is running on a differen IP than UDP and TCP, then add that too */
      if (sip_tls_desc.local_address.sin_addr.s_addr && !inaddrcmp(&bindaddr, &sip_tls_desc.local_address) && inaddrcmp(&sip_tcp_desc.local_address, &sip_tls_desc.local_address))
         add_sip_domain(ast_inet_ntoa(sip_tls_desc.local_address.sin_addr), SIP_DOMAIN_AUTO, NULL);

      /* Our extern IP address, if configured */
      if (externip.sin_addr.s_addr)
         add_sip_domain(ast_inet_ntoa(externip.sin_addr), SIP_DOMAIN_AUTO, NULL);

      /* Extern host name (NAT traversal support) */
      if (!ast_strlen_zero(externhost))
         add_sip_domain(externhost, SIP_DOMAIN_AUTO, NULL);
      
      /* Our host name */
      if (!gethostname(temp, sizeof(temp)))
         add_sip_domain(temp, SIP_DOMAIN_AUTO, NULL);
   }

   /* Release configuration from memory */
   ast_config_destroy(cfg);

   /* Load the list of manual NOTIFY types to support */
   if (notify_types)
      ast_config_destroy(notify_types);
   notify_types = ast_config_load(notify_config, config_flags);

   /* Done, tell the manager */
   manager_event(EVENT_FLAG_SYSTEM, "ChannelReload", "ChannelType: SIP\r\nReloadReason: %s\r\nRegistry_Count: %d\r\nPeer_Count: %d\r\n", channelreloadreason2txt(reason), registry_count, peer_count);
   run_end = time(0);
   ast_debug(4, "SIP reload_config done...Runtime= %d sec\n", (int)(run_end-run_start));

   return 0;
}

static char * remove_uri_parameters

(

char *

uri

)

[static]

Definition at line 9112 of file chan_sip.c.

Referenced by extract_uri(), parse_moved_contact(), register_verify(), reqprep(), and transmit_state_notify().

{
   char *atsign;
   atsign = strchr(uri, '@'); /* First, locate the at sign */
   if (!atsign)
      atsign = uri;  /* Ok hostname only, let's stick with the rest */
   atsign = strchr(atsign, ';'); /* Locate semi colon */
   if (atsign)
      *atsign = '\0';   /* Kill at the semi colon */
   return uri;
}

static void replace_cid	(	struct sip_pvt *	p,
		const char *	rpid_num,
		const char *	calleridname
	)			[static]

helper function for check_{user|peer}_ok()

Definition at line 12192 of file chan_sip.c.

References ast_is_shrinkable_phonenumber(), ast_shrink_phone_number(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, cid_name, cid_num, sip_pvt::flags, and SIP_TRUSTRPID.

Referenced by check_peer_ok(), and check_user_full().

{
   /* replace callerid if rpid found, and not restricted */
   if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) {
      char *tmp = ast_strdupa(rpid_num); /* XXX the copy can be done later */
      if (!ast_strlen_zero(calleridname))
         ast_string_field_set(p, cid_name, calleridname);
      if (ast_is_shrinkable_phonenumber(tmp))
         ast_shrink_phone_number(tmp);
      ast_string_field_set(p, cid_num, tmp);
   }
}

static int reply_digest	(	struct sip_pvt *	p,
		struct sip_request *	req,
		char *	header,
		int	sipmethod,
		char *	digest,
		int	digest_len
	)			[static]

reply to authentication for outbound registrations

Returns:

Returns -1 if we have no auth

Note:

This is used for register= servers in sip.conf, SIP proxies we register with for receiving calls from.

Definition at line 15288 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_skip_blanks(), ast_string_field_ptr_set, ast_string_field_set, ast_strlen_zero(), build_reply_digest(), sip_pvt::domain, get_header(), LOG_WARNING, sip_registry::nonce, sip_pvt::nonce, sip_registry::noncecount, sip_pvt::noncecount, sip_registry::opaque, sip_pvt::opaque, sip_registry::qop, sip_pvt::qop, sip_registry::realm, sip_pvt::realm, sip_pvt::registry, and strsep().

Referenced by do_proxy_auth(), and do_register_auth().

{
   char tmp[512];
   char *c;
   char oldnonce[256];

   /* table of recognised keywords, and places where they should be copied */
   const struct x {
      const char *key;
      const ast_string_field *field;
   } *i, keys[] = {
      { "realm=", &p->realm },
      { "nonce=", &p->nonce },
      { "opaque=", &p->opaque },
      { "qop=", &p->qop },
      { "domain=", &p->domain },
      { NULL, 0 },
   };

   ast_copy_string(tmp, get_header(req, header), sizeof(tmp));
   if (ast_strlen_zero(tmp)) 
      return -1;
   if (strncasecmp(tmp, "Digest ", strlen("Digest "))) {
      ast_log(LOG_WARNING, "missing Digest.\n");
      return -1;
   }
   c = tmp + strlen("Digest ");
   ast_copy_string(oldnonce, p->nonce, sizeof(oldnonce));
   while (c && *(c = ast_skip_blanks(c))) {  /* lookup for keys */
      for (i = keys; i->key != NULL; i++) {
         char *src, *separator;
         if (strncasecmp(c, i->key, strlen(i->key)) != 0)
            continue;
         /* Found. Skip keyword, take text in quotes or up to the separator. */
         c += strlen(i->key);
         if (*c == '"') {
            src = ++c;
            separator = "\"";
         } else {
            src = c;
            separator = ",";
         }
         strsep(&c, separator); /* clear separator and move ptr */
         ast_string_field_ptr_set(p, i->field, src);
         break;
      }
      if (i->key == NULL) /* not found, try ',' */
         strsep(&c, ",");
   }
   /* Reset nonce count */
   if (strcmp(p->nonce, oldnonce)) 
      p->noncecount = 0;

   /* Save auth data for following registrations */
   if (p->registry) {
      struct sip_registry *r = p->registry;

      if (strcmp(r->nonce, p->nonce)) {
         ast_string_field_set(r, realm, p->realm);
         ast_string_field_set(r, nonce, p->nonce);
         ast_string_field_set(r, domain, p->domain);
         ast_string_field_set(r, opaque, p->opaque);
         ast_string_field_set(r, qop, p->qop);
         r->noncecount = 0;
      }
   }
   return build_reply_digest(p, sipmethod, digest, digest_len); 
}

static int reqprep	(	struct sip_request *	req,
		struct sip_pvt *	p,
		int	sipmethod,
		int	seqno,
		int	newbranch
	)			[static]

Initialize a SIP request message (not the initial one in a dialog).

< Strict routing flag

Definition at line 8029 of file chan_sip.c.

References add_header(), add_route(), ast_copy_string(), ast_debug, ast_random(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, sip_pvt::branch, build_via(), sip_pvt::callid, copy_header(), DEFAULT_MAX_FORWARDS, FALSE, sip_pvt::flags, get_header(), get_in_brackets(), sip_route::hop, init_req(), sip_pvt::initreq, sip_pvt::invite_branch, sip_pvt::lastmsg, sip_route::next, sip_pvt::ocseq, sip_pvt::okcontacturi, sip_pvt::our_contact, remove_uri_parameters(), REQ_OFFSET_TO_STR, sip_request::rlPart2, sip_pvt::route, sip_pvt::rpid, set_destination(), SIP_ACK, SIP_BYE, SIP_CANCEL, SIP_INVITE, SIP_MESSAGE, sip_methods, SIP_OUTGOING, sip_st_dlg::st_active, sip_st_dlg::st_active_peer_ua, st_get_se(), sip_st_dlg::st_interval, sip_st_dlg::st_ref, sip_pvt::stimer, strcasestr(), strefresher2str(), sip_pvt::tag, text, cfsip_methods::text, sip_pvt::theirtag, TRUE, sip_pvt::uri, url, sip_pvt::url, and sip_pvt::via.

{
   struct sip_request *orig = &p->initreq;
   char stripped[80];
   char tmp[80];
   char newto[256];
   const char *c;
   const char *ot, *of;
   int is_strict = FALSE;     /*!< Strict routing flag */
   int is_outbound = ast_test_flag(&p->flags[0], SIP_OUTGOING);   /* Session direction */

   memset(req, 0, sizeof(struct sip_request));
   
   snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text);
   
   if (!seqno) {
      p->ocseq++;
      seqno = p->ocseq;
   }
   
   if (sipmethod == SIP_CANCEL || sipmethod == SIP_INVITE) {
      p->branch = p->invite_branch;
      build_via(p);
   } else if (newbranch) {
      p->branch ^= ast_random();
      build_via(p);
   }

   /* Check for strict or loose router */
   if (p->route && !ast_strlen_zero(p->route->hop) && strstr(p->route->hop, ";lr") == NULL) {
      is_strict = TRUE;
      if (sipdebug)
         ast_debug(1, "Strict routing enforced for session %s\n", p->callid);
   }
   
   if (sipmethod == SIP_CANCEL)
      c = REQ_OFFSET_TO_STR(&p->initreq, rlPart2); /* Use original URI */
   else if (sipmethod == SIP_ACK) {
      /* Use URI from Contact: in 200 OK (if INVITE) 
      (we only have the contacturi on INVITEs) */
      if (!ast_strlen_zero(p->okcontacturi))
         c = is_strict ? p->route->hop : p->okcontacturi;
      else
         c = REQ_OFFSET_TO_STR(&p->initreq, rlPart2);
   } else if (!ast_strlen_zero(p->okcontacturi)) 
      c = is_strict ? p->route->hop : p->okcontacturi; /* Use for BYE or REINVITE */
   else if (!ast_strlen_zero(p->uri)) 
      c = p->uri;
   else {
      char *n;
      /* We have no URI, use To: or From:  header as URI (depending on direction) */
      ast_copy_string(stripped, get_header(orig, is_outbound ? "To" : "From"),
            sizeof(stripped));
      n = get_in_brackets(stripped);
      c = remove_uri_parameters(n);
   }  
   init_req(req, sipmethod, c);

   snprintf(tmp, sizeof(tmp), "%d %s", seqno, sip_methods[sipmethod].text);

   add_header(req, "Via", p->via);
   if (p->route) {
      set_destination(p, p->route->hop);
      add_route(req, is_strict ? p->route->next : p->route);
   }
   add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);

   ot = get_header(orig, "To");
   of = get_header(orig, "From");

   /* Add tag *unless* this is a CANCEL, in which case we need to send it exactly
      as our original request, including tag (or presumably lack thereof) */
   if (!strcasestr(ot, "tag=") && sipmethod != SIP_CANCEL) {
      /* Add the proper tag if we don't have it already.  If they have specified
         their tag, use it.  Otherwise, use our own tag */
      if (is_outbound && !ast_strlen_zero(p->theirtag))
         snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
      else if (!is_outbound)
         snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
      else
         snprintf(newto, sizeof(newto), "%s", ot);
      ot = newto;
   }

   if (is_outbound) {
      add_header(req, "From", of);
      add_header(req, "To", ot);
   } else {
      add_header(req, "From", ot);
      add_header(req, "To", of);
   }
   /* Do not add Contact for MESSAGE, BYE and Cancel requests */
   if (sipmethod != SIP_BYE && sipmethod != SIP_CANCEL && sipmethod != SIP_MESSAGE)
      add_header(req, "Contact", p->our_contact);

   copy_header(req, orig, "Call-ID");
   add_header(req, "CSeq", tmp);

   if (!ast_strlen_zero(global_useragent))
      add_header(req, "User-Agent", global_useragent);

   if (!ast_strlen_zero(p->rpid))
      add_header(req, "Remote-Party-ID", p->rpid);

   if (!ast_strlen_zero(p->url)) {
      add_header(req, "Access-URL", p->url);
      ast_string_field_set(p, url, NULL);
   }

   /* Add Session-Timers related headers if the feature is active for this session.
      An exception to this behavior is the ACK request. Since Asterisk never requires 
      session-timers support from a remote end-point (UAS) in an INVITE, it must 
      not send 'Require: timer' header in the ACK request. 
      This should only be added in the INVITE transactions, not MESSAGE or REFER or other
      in-dialog messages.
   */
   if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_active_peer_ua == TRUE 
       && sipmethod == SIP_INVITE) {
      char se_hdr[256];
      snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval, 
         strefresher2str(p->stimer->st_ref));
      add_header(req, "Require", "timer");
      add_header(req, "Session-Expires", se_hdr);
      snprintf(se_hdr, sizeof(se_hdr), "%d", st_get_se(p, FALSE));
      add_header(req, "Min-SE", se_hdr);
   }

   return 0;
}

static int resp_needs_contact	(	const char *	msg,
		enum sipmethod	method
	)			[inline, static]

Test if this response needs a contact header.

Definition at line 7904 of file chan_sip.c.

References SIP_ACK, SIP_BYE, SIP_CANCEL, SIP_INFO, SIP_INVITE, SIP_MESSAGE, SIP_NOTIFY, SIP_OPTIONS, SIP_PING, SIP_PRACK, SIP_PUBLISH, SIP_REFER, SIP_REGISTER, SIP_SUBSCRIBE, and SIP_UPDATE.

Referenced by respprep().

                                                                             {
   /* Requirements for Contact header inclusion in responses generated
    * from the header tables found in the following RFCs.  Where the
    * Contact header was marked mandatory (m) or optional (o) this
    * function returns 1.
    *
    * - RFC 3261 (ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER)
    * - RFC 2976 (INFO)
    * - RFC 3262 (PRACK)
    * - RFC 3265 (SUBSCRIBE, NOTIFY)
    * - RFC 3311 (UPDATE)
    * - RFC 3428 (MESSAGE)
    * - RFC 3515 (REFER)
    * - RFC 3903 (PUBLISH)
    */

   switch (method) {
      /* 1xx, 2xx, 3xx, 485 */
      case SIP_INVITE:
      case SIP_UPDATE:
      case SIP_SUBSCRIBE:
      case SIP_NOTIFY:
         if ((msg[0] >= '1' && msg[0] <= '3') || !strncmp(msg, "485", 3))
            return 1;
         break;

      /* 2xx, 3xx, 485 */
      case SIP_REGISTER:
      case SIP_OPTIONS:
         if (msg[0] == '2' || msg[0] == '3' || !strncmp(msg, "485", 3))
            return 1;
         break;

      /* 3xx, 485 */
      case SIP_BYE:
      case SIP_PRACK:
      case SIP_MESSAGE:
      case SIP_PUBLISH:
         if (msg[0] == '3' || !strncmp(msg, "485", 3))
            return 1;
         break;

      /* 2xx, 3xx, 4xx, 5xx, 6xx */
      case SIP_REFER:
         if (msg[0] >= '2' && msg[0] <= '6')
            return 1;
         break;

      /* contact will not be included for everything else */
      case SIP_ACK:
      case SIP_CANCEL:
      case SIP_INFO:
      case SIP_PING:
      default:
         return 0;
   }
   return 0;
}

static int respprep	(	struct sip_request *	resp,
		struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req
	)			[static]

Prepare SIP response packet.

Definition at line 7965 of file chan_sip.c.

References add_header(), ALLOWED_METHODS, ast_copy_string(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, copy_all_header(), copy_header(), copy_via_headers(), sip_pvt::expiry, sip_pvt::flags, get_header(), init_resp(), sip_pvt::method, sip_pvt::our_contact, resp_needs_contact(), SIP_INVITE, SIP_OUTGOING, SIP_REGISTER, SIP_SUBSCRIBE, SIPBUFSIZE, sip_st_dlg::st_active, sip_st_dlg::st_active_peer_ua, sip_st_dlg::st_interval, sip_st_dlg::st_ref, sip_pvt::stimer, strcasestr(), strefresher2str(), SUPPORTED_EXTENSIONS, sip_pvt::tag, sip_pvt::theirtag, TRUE, url, and sip_pvt::url.

{
   char newto[256];
   const char *ot;

   init_resp(resp, msg);
   copy_via_headers(p, resp, req, "Via");
   if (msg[0] == '1' || msg[0] == '2')
      copy_all_header(resp, req, "Record-Route");
   copy_header(resp, req, "From");
   ot = get_header(req, "To");
   if (!strcasestr(ot, "tag=") && strncmp(msg, "100", 3)) {
      /* Add the proper tag if we don't have it already.  If they have specified
         their tag, use it.  Otherwise, use our own tag */
      if (!ast_strlen_zero(p->theirtag) && ast_test_flag(&p->flags[0], SIP_OUTGOING))
         snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
      else if (p->tag && !ast_test_flag(&p->flags[0], SIP_OUTGOING))
         snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
      else
         ast_copy_string(newto, ot, sizeof(newto));
      ot = newto;
   }
   add_header(resp, "To", ot);
   copy_header(resp, req, "Call-ID");
   copy_header(resp, req, "CSeq");
   if (!ast_strlen_zero(global_useragent))
      add_header(resp, "Server", global_useragent);
   add_header(resp, "Allow", ALLOWED_METHODS);
   add_header(resp, "Supported", SUPPORTED_EXTENSIONS);

   /* If this is an invite, add Session-Timers related headers if the feature is active for this session */
   if (p->method == SIP_INVITE && p->stimer && p->stimer->st_active == TRUE && p->stimer->st_active_peer_ua == TRUE) {
      char se_hdr[256];
      snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval, 
         strefresher2str(p->stimer->st_ref));
      add_header(resp, "Require", "timer");
      add_header(resp, "Session-Expires", se_hdr);
   }

   if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER)) {
      /* For registration responses, we also need expiry and
         contact info */
      char tmp[256];

      snprintf(tmp, sizeof(tmp), "%d", p->expiry);
      add_header(resp, "Expires", tmp);
      if (p->expiry) {  /* Only add contact if we have an expiry time */
         char contact[SIPBUFSIZE];
         snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry);
         add_header(resp, "Contact", contact);  /* Not when we unregister */
      }
   } else if (!ast_strlen_zero(p->our_contact) && resp_needs_contact(msg, p->method)) {
      add_header(resp, "Contact", p->our_contact);
   }

   if (!ast_strlen_zero(p->url)) {
      add_header(resp, "Access-URL", p->url);
      ast_string_field_set(p, url, NULL);
   }

   return 0;
}

static int restart_monitor

(

void

)

[static]

Start the channel monitor thread.

Definition at line 20419 of file chan_sip.c.

References ast_log(), ast_mutex_lock(), ast_mutex_unlock(), ast_pthread_create_background, AST_PTHREADT_NULL, AST_PTHREADT_STOP, do_monitor(), LOG_ERROR, LOG_WARNING, and monlock.

{
   /* If we're supposed to be stopped -- stay stopped */
   if (monitor_thread == AST_PTHREADT_STOP)
      return 0;
   ast_mutex_lock(&monlock);
   if (monitor_thread == pthread_self()) {
      ast_mutex_unlock(&monlock);
      ast_log(LOG_WARNING, "Cannot kill myself\n");
      return -1;
   }
   if (monitor_thread != AST_PTHREADT_NULL) {
      /* Wake up the thread */
      pthread_kill(monitor_thread, SIGURG);
   } else {
      /* Start a new monitor */
      if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
         ast_mutex_unlock(&monlock);
         ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
         return -1;
      }
   }
   ast_mutex_unlock(&monlock);
   return 0;
}

static void restart_session_timer

(

struct sip_pvt *

p

)

[static]

Session-Timers: Restart session timer.

Definition at line 20447 of file chan_sip.c.

References ast_debug, ast_log(), AST_SCHED_DEL_UNREF, sip_pvt::callid, dialog_unref(), LOG_WARNING, sched, sip_st_dlg::st_active, sip_st_dlg::st_schedid, start_session_timer(), sip_pvt::stimer, and TRUE.

Referenced by handle_request_invite().

{
   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in restart_session_timer - %s\n", p->callid);
      return;
   }

   if (p->stimer->st_active == TRUE) {
      AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
            dialog_unref(p, "Removing session timer ref"));
      ast_debug(2, "Session timer stopped: %d - %s\n", p->stimer->st_schedid, p->callid);
      start_session_timer(p);
   }
}

static int retrans_pkt

(

const void *

data

)

[static]

Retransmit SIP message if no answer (Called from scheduler).

Definition at line 3022 of file chan_sip.c.

References __sip_xmit(), append_history, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_PROTOCOL_ERROR, ast_channel_trylock, ast_channel_unlock, ast_debug, ast_free, ast_inet_ntoa(), ast_log(), ast_queue_hangup_with_cause(), ast_verbose, sip_pvt::callid, sip_pkt::data, DEFAULT_RETRANS, dialog_unref(), ast_channel::hangupcause, sip_pkt::is_fatal, sip_pkt::is_resp, LOG_WARNING, MAX_RETRANS, sip_pkt::method, sip_pvt::needdestroy, sip_pkt::next, sip_pvt::owner, sip_pkt::owner, sip_pkt::packetlen, sip_pvt::packets, sip_pkt::retrans, sip_pkt::retransid, sip_pkt::seqno, sip_alreadygone(), SIP_BYE, sip_debug_test_pvt(), SIP_INVITE, sip_methods, sip_nat_mode(), SIP_OPTIONS, sip_pvt_lock, sip_pvt_unlock, sip_real_dst(), SIP_REGISTER, ast_str::str, sip_pkt::timer_a, sip_pkt::timer_t1, UNLINK, and XMIT_ERROR.

{
   struct sip_pkt *pkt = (struct sip_pkt *)data, *prev, *cur = NULL;
   int reschedule = DEFAULT_RETRANS;
   int xmitres = 0;
   
   /* Lock channel PVT */
   sip_pvt_lock(pkt->owner);

   if (pkt->retrans < MAX_RETRANS) {
      pkt->retrans++;
      if (!pkt->timer_t1) {   /* Re-schedule using timer_a and timer_t1 */
         if (sipdebug)
            ast_debug(4, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method);
      } else {
         int siptimer_a;

         if (sipdebug)
            ast_debug(4, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
         if (!pkt->timer_a)
            pkt->timer_a = 2 ;
         else
            pkt->timer_a = 2 * pkt->timer_a;
 
         /* For non-invites, a maximum of 4 secs */
         siptimer_a = pkt->timer_t1 * pkt->timer_a;   /* Double each time */
         if (pkt->method != SIP_INVITE && siptimer_a > 4000)
            siptimer_a = 4000;
      
         /* Reschedule re-transmit */
         reschedule = siptimer_a;
         ast_debug(4, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid);
      } 

      if (sip_debug_test_pvt(pkt->owner)) {
         const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
         ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
            pkt->retrans, sip_nat_mode(pkt->owner),
            ast_inet_ntoa(dst->sin_addr),
            ntohs(dst->sin_port), pkt->data->str);
      }

      append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data->str);
      xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
      sip_pvt_unlock(pkt->owner);
      if (xmitres == XMIT_ERROR)
         ast_log(LOG_WARNING, "Network error on retransmit in dialog %s\n", pkt->owner->callid);
      else 
         return  reschedule;
   } 
   /* Too many retries */
   if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) {
      if (pkt->is_fatal || sipdebug)   /* Tell us if it's critical or if we're debugging */
         ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s) -- See doc/sip-retransmit.txt.\n",
            pkt->owner->callid, pkt->seqno,
            pkt->is_fatal ? "Critical" : "Non-critical", pkt->is_resp ? "Response" : "Request");
   } else if (pkt->method == SIP_OPTIONS && sipdebug) {
         ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s)  -- See doc/sip-retransmit.txt.\n", pkt->owner->callid);

   } 
   if (xmitres == XMIT_ERROR) {
      ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission on Call ID %s\n", pkt->owner->callid);
      append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
   } else 
      append_history(pkt->owner, "MaxRetries", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
      
   pkt->retransid = -1;

   if (pkt->is_fatal) {
      while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
         sip_pvt_unlock(pkt->owner);   /* SIP_PVT, not channel */
         usleep(1);
         sip_pvt_lock(pkt->owner);
      }

      if (pkt->owner->owner && !pkt->owner->owner->hangupcause) 
         pkt->owner->owner->hangupcause = AST_CAUSE_NO_USER_RESPONSE;
      
      if (pkt->owner->owner) {
         sip_alreadygone(pkt->owner);
         ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see doc/sip-retransmit.txt).\n", pkt->owner->callid);
         ast_queue_hangup_with_cause(pkt->owner->owner, AST_CAUSE_PROTOCOL_ERROR);
         ast_channel_unlock(pkt->owner->owner);
      } else {
         /* If no channel owner, destroy now */

         /* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
         if (pkt->method != SIP_OPTIONS && pkt->method != SIP_REGISTER) {
            pkt->owner->needdestroy = 1;
            sip_alreadygone(pkt->owner);
            append_history(pkt->owner, "DialogKill", "Killing this failed dialog immediately");
         }
      }
   }

   if (pkt->method == SIP_BYE) {
      /* We're not getting answers on SIP BYE's.  Tear down the call anyway. */
      if (pkt->owner->owner) 
         ast_channel_unlock(pkt->owner->owner);
      append_history(pkt->owner, "ByeFailure", "Remote peer doesn't respond to bye. Destroying call anyway.");
      pkt->owner->needdestroy = 1;
   }

   /* Remove the packet */
   for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
      if (cur == pkt) {
         UNLINK(cur, pkt->owner->packets, prev);
         sip_pvt_unlock(pkt->owner);
         if (pkt->owner)
            pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
         if (pkt->data)
            ast_free(pkt->data);
         pkt->data = NULL;
         ast_free(pkt);
         return 0;
      }
   }
   /* error case */
   ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
   sip_pvt_unlock(pkt->owner);
   return 0;
}

static int scheduler_process_request_queue

(

const void *

data

)

[static]

Definition at line 19789 of file chan_sip.c.

References ast_channel_trylock, sip_pvt::owner, sip_pvt_lock, and sip_pvt_unlock.

Referenced by queue_request().

{
   struct sip_pvt *p = (struct sip_pvt *) data;
   int recount = 0;
   int nounlock = 0;
   int lockretry;

   for (lockretry = 10; lockretry > 0; lockretry--) {
      sip_pvt_lock(p);

      /* lock the owner if it has one -- we may need it */
      /* because this is deadlock-prone, we need to try and unlock if failed */
      if (!p->owner || !ast_channel_trylock(p->owner)) {
         break;   /* locking succeeded */
      }

      if (lockretry != 1) {
         sip_pvt_unlock(p);
         /* Sleep for a very short amount of time */
         usleep(1);
      }
   }

   if (!lockretry) {
      int retry = !AST_LIST_EMPTY(&p->request_queue);

      /* we couldn't get the owner lock, which is needed to process
         the queued requests, so return a non-zero value, which will
         cause the scheduler to run this request again later if there
         still requests to be processed
      */
      sip_pvt_unlock(p);
      if (!retry) {
         dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");
      }
      return retry;
   };

   process_request_queue(p, &recount, &nounlock);
   p->request_queue_sched_id = -1;

   if (p->owner && !nounlock) {
      ast_channel_unlock(p->owner);
   }
   sip_pvt_unlock(p);

   if (recount) {
      ast_update_use_count();
   }

   dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");

   return 0;
}

static int send_request	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum xmittype	reliable,
		int	seqno
	)			[static]

Send SIP Request to the other part of the dialogue.

Definition at line 3461 of file chan_sip.c.

References __sip_reliable_xmit(), __sip_xmit(), add_blank(), append_history, ast_free, ast_inet_ntoa(), ast_test_flag, ast_verbose, sip_request::data, sip_pvt::do_history, sip_pvt::flags, get_header(), sip_proxy::ip, sip_request::len, sip_request::method, sip_pvt::outboundproxy, parse_copy(), sip_pvt::recv, sip_request::rlPart1, sip_pvt::sa, sip_debug_test_pvt(), sip_methods, SIP_NAT_ROUTE, ast_str::str, and XMIT_CRITICAL.

{
   int res;

   /* If we have an outbound proxy, reset peer address 
      Only do this once.
   */
   if (p->outboundproxy) {
      p->sa = p->outboundproxy->ip;
   }

   add_blank(req);
   if (sip_debug_test_pvt(p)) {
      if (ast_test_flag(&p->flags[0], SIP_NAT_ROUTE))
         ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port), req->data->str);
      else
         ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data->str);
   }
   if (p->do_history) {
      struct sip_request tmp = { .rlPart1 = 0, };
      parse_copy(&tmp, req);
      append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data->str, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
      ast_free(tmp.data);
   }
   res = (reliable) ?
      __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
      __sip_xmit(p, req->data, req->len);
   if (req->data) {
      ast_free(req->data);
      req->data = NULL;
   }
   return res;
}

static int send_response	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum xmittype	reliable,
		int	seqno
	)			[static]

Transmit response on SIP request.

Definition at line 3430 of file chan_sip.c.

References __sip_reliable_xmit(), __sip_xmit(), add_blank(), append_history, ast_free, ast_inet_ntoa(), ast_verbose, sip_request::data, sip_pvt::do_history, get_header(), sip_request::len, sip_request::method, parse_copy(), REQ_OFFSET_TO_STR, sip_request::rlPart1, sip_request::rlPart2, sip_debug_test_pvt(), sip_methods, sip_nat_mode(), sip_real_dst(), SIP_RESPONSE, SIP_UNKNOWN, ast_str::str, cfsip_methods::text, and XMIT_CRITICAL.

{
   int res;

   add_blank(req);
   if (sip_debug_test_pvt(p)) {
      const struct sockaddr_in *dst = sip_real_dst(p);

      ast_verbose("\n<--- %sTransmitting (%s) to %s:%d --->\n%s\n<------------>\n",
         reliable ? "Reliably " : "", sip_nat_mode(p),
         ast_inet_ntoa(dst->sin_addr),
         ntohs(dst->sin_port), req->data->str);
   }
   if (p->do_history) {
      struct sip_request tmp = { .rlPart1 = 0, };
      parse_copy(&tmp, req);
      append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data->str, get_header(&tmp, "CSeq"), 
         (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? REQ_OFFSET_TO_STR(&tmp, rlPart2) : sip_methods[tmp.method].text);
      ast_free(tmp.data);
   }
   res = (reliable) ?
       __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
      __sip_xmit(p, req->data, req->len);
   ast_free(req->data);
   req->data = NULL;
   if (res > 0)
      return 0;
   return res;
}

static int set_address_from_contact

(

struct sip_pvt *

pvt

)

[static]

Change the other partys IP address based on given contact.

Definition at line 10568 of file chan_sip.c.

References __set_address_from_contact(), ast_test_flag, sip_pvt::flags, sip_pvt::fullcontact, sip_pvt::recv, sip_pvt::sa, SIP_NAT_ROUTE, SIP_TRANSPORT_TLS, sip_pvt::socket, and sip_socket::type.

Referenced by handle_response_invite().

{
   if (ast_test_flag(&pvt->flags[0], SIP_NAT_ROUTE)) {
      /* NAT: Don't trust the contact field.  Just use what they came to us
         with. */
      pvt->sa = pvt->recv;
      return 0;
   }

   return __set_address_from_contact(pvt->fullcontact, &pvt->sa, pvt->socket.type == SIP_TRANSPORT_TLS ? 1 : 0);
}

static void set_destination	(	struct sip_pvt *	p,
		char *	uri
	)			[static]

Set destination from SIP URI.

Definition at line 7810 of file chan_sip.c.

References ahp, ast_copy_string(), ast_gethostbyname(), ast_inet_ntoa(), ast_log(), ast_verbose, sip_request::debug, hostname, hp, LOG_WARNING, sip_pvt::sa, sip_debug_test_pvt(), and STANDARD_SIP_PORT.

Referenced by reqprep().

{
   char *h, *maddr, hostname[256];
   int port, hn;
   struct hostent *hp;
   struct ast_hostent ahp;
   int debug=sip_debug_test_pvt(p);

   /* Parse uri to h (host) and port - uri is already just the part inside the <> */
   /* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...] */

   if (debug)
      ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);

   /* Find and parse hostname */
   h = strchr(uri, '@');
   if (h)
      ++h;
   else {
      h = uri;
      if (!strncasecmp(h, "sip:", 4))
         h += 4;
      else if (!strncasecmp(h, "sips:", 5))
         h += 5;
   }
   hn = strcspn(h, ":;>") + 1;
   if (hn > sizeof(hostname)) 
      hn = sizeof(hostname);
   ast_copy_string(hostname, h, hn);
   /* XXX bug here if string has been trimmed to sizeof(hostname) */
   h += hn - 1;

   /* Is "port" present? if not default to STANDARD_SIP_PORT */
   if (*h == ':') {
      /* Parse port */
      ++h;
      port = strtol(h, &h, 10);
   }
   else
      port = STANDARD_SIP_PORT;

   /* Got the hostname:port - but maybe there's a "maddr=" to override address? */
   maddr = strstr(h, "maddr=");
   if (maddr) {
      maddr += 6;
      hn = strspn(maddr, "0123456789.") + 1;
      if (hn > sizeof(hostname))
         hn = sizeof(hostname);
      ast_copy_string(hostname, maddr, hn);
   }
   
   hp = ast_gethostbyname(hostname, &ahp);
   if (hp == NULL)  {
      ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
      return;
   }
   p->sa.sin_family = AF_INET;
   memcpy(&p->sa.sin_addr, hp->h_addr, sizeof(p->sa.sin_addr));
   p->sa.sin_port = htons(port);
   if (debug)
      ast_verbose("set_destination: set destination to %s, port %d\n", ast_inet_ntoa(p->sa.sin_addr), port);
}

static void set_insecure_flags	(	struct ast_flags *	flags,
		const char *	value,
		int	lineno
	)			[static]

Parse insecure= setting in sip.conf and set flags according to setting.

Definition at line 21152 of file chan_sip.c.

References ast_copy_string(), ast_false(), ast_log(), ast_set_flag, ast_strlen_zero(), buf, ast_channel::flags, LOG_WARNING, ast_channel::next, SIP_INSECURE_INVITE, SIP_INSECURE_PORT, and strsep().

Referenced by get_insecure_variable_from_config(), and handle_common_options().

{
   if (ast_strlen_zero(value))
      return;

   if (!ast_false(value)) {
      char buf[64];
      char *word, *next;

      ast_copy_string(buf, value, sizeof(buf));
      next = buf;
      while ((word = strsep(&next, ","))) {
         if (!strcasecmp(word, "port"))
            ast_set_flag(&flags[0], SIP_INSECURE_PORT);
         else if (!strcasecmp(word, "invite"))
            ast_set_flag(&flags[0], SIP_INSECURE_INVITE);
         else
            ast_log(LOG_WARNING, "Unknown insecure mode '%s' on line %d\n", value, lineno);
      }
   }
}

static void set_peer_defaults

(

struct sip_peer *

peer

)

[static]

Set peer defaults before configuring specific configurations.

Definition at line 21472 of file chan_sip.c.

References sip_peer::addr, sip_peer::allowtransfer, ast_copy_flags, sip_peer::autoframing, sip_peer::call_limit, sip_peer::callgroup, sip_peer::capability, sip_peer::cid_name, sip_peer::cid_num, clear_peer_mailboxes(), sip_peer::context, sip_peer::defaddr, default_prefs, sip_peer::expire, sip_socket::fd, sip_peer::flags, sip_peer::fromdomain, sip_peer::fromuser, global_flags, sip_peer::language, sip_peer::maxcallbitrate, sip_peer::maxms, sip_peer::md5secret, sip_peer::mohinterpret, sip_peer::mohsuggest, sip_peer::pickupgroup, sip_peer::pokeexpire, sip_peer::prefs, sip_peer::qualifyfreq, sip_peer::regexten, sip_peer::rtpholdtimeout, sip_peer::rtpkeepalive, sip_peer::rtptimeout, sip_peer::secret, SIP_FLAGS_TO_COPY, SIP_PAGE2_FLAGS_TO_COPY, SIP_TRANSPORT_UDP, SIP_TYPE_PEER, sip_peer::socket, sip_st_cfg::st_max_se, sip_st_cfg::st_min_se, sip_st_cfg::st_mode_oper, sip_st_cfg::st_ref, STANDARD_SIP_PORT, sip_peer::stimer, sip_peer::subscribecontext, sip_peer::timer_b, sip_peer::timer_t1, sip_peer::type, sip_socket::type, and sip_peer::vmexten.

Referenced by build_peer(), and temp_peer().

{
   if (peer->expire == 0) {
      /* Don't reset expire or port time during reload 
         if we have an active registration 
      */
      peer->expire = -1;
      peer->pokeexpire = -1;
      peer->addr.sin_port = htons(STANDARD_SIP_PORT);
      peer->socket.type = SIP_TRANSPORT_UDP;
      peer->socket.fd = -1;
   }
   peer->type = SIP_TYPE_PEER;
   ast_copy_flags(&peer->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   strcpy(peer->context, default_context);
   strcpy(peer->subscribecontext, default_subscribecontext);
   strcpy(peer->language, default_language);
   strcpy(peer->mohinterpret, default_mohinterpret);
   strcpy(peer->mohsuggest, default_mohsuggest);
   peer->addr.sin_family = AF_INET;
   peer->defaddr.sin_family = AF_INET;
   peer->capability = global_capability;
   peer->maxcallbitrate = default_maxcallbitrate;
   peer->rtptimeout = global_rtptimeout;
   peer->rtpholdtimeout = global_rtpholdtimeout;
   peer->rtpkeepalive = global_rtpkeepalive;
   peer->allowtransfer = global_allowtransfer;
   peer->autoframing = global_autoframing;
   peer->qualifyfreq = global_qualifyfreq;
   if (global_callcounter)
      peer->call_limit=999;
   strcpy(peer->vmexten, default_vmexten);
   peer->secret[0] = '\0';
   peer->md5secret[0] = '\0';
   peer->cid_num[0] = '\0';
   peer->cid_name[0] = '\0';
   peer->fromdomain[0] = '\0';
   peer->fromuser[0] = '\0';
   peer->regexten[0] = '\0';
   peer->callgroup = 0;
   peer->pickupgroup = 0;
   peer->maxms = default_qualify;
   peer->prefs = default_prefs;
   peer->stimer.st_mode_oper = global_st_mode;  /* Session-Timers */
   peer->stimer.st_ref = global_st_refresher;
   peer->stimer.st_min_se = global_min_se;
   peer->stimer.st_max_se = global_max_se;
   peer->timer_t1 = global_t1;
   peer->timer_b = global_timer_b;
   clear_peer_mailboxes(peer);
}

static void set_t38_capabilities

(

struct sip_pvt *

p

)

[static]

Set the global T38 capabilities on a SIP dialog structure.

Definition at line 4273 of file chan_sip.c.

References ast_udptl_get_error_correction_scheme(), t38properties::capability, sip_pvt::t38, T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF, T38FAX_UDP_EC_FEC, T38FAX_UDP_EC_NONE, T38FAX_UDP_EC_REDUNDANCY, sip_pvt::udptl, UDPTL_ERROR_CORRECTION_FEC, UDPTL_ERROR_CORRECTION_NONE, and UDPTL_ERROR_CORRECTION_REDUNDANCY.

Referenced by create_addr_from_peer(), and sip_alloc().

{
   p->t38.capability = global_t38_capability;
   if (p->udptl) {
      if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_FEC )
         p->t38.capability |= T38FAX_UDP_EC_FEC;
      else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
         p->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
      else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_NONE )
         p->t38.capability |= T38FAX_UDP_EC_NONE;
      p->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
   }
}

static int show_channels_cb	(	void *	__cur,
		void *	__arg,
		int	flags
	)			[static]

callback for show channel|subscription

Definition at line 14411 of file chan_sip.c.

References ast_cli(), ast_extension_state2str(), ast_getformatname_multiple(), ast_inet_ntoa(), ast_str_alloca, ast_test_flag, sip_pvt::callid, sip_pvt::cid_num, cli_yesno(), sip_pvt::expiry, __show_chan_arg::fd, sip_pvt::flags, FORMAT, FORMAT4, sip_pvt::lastmsg, sip_pvt::laststate, ast_channel::nativeformats, sip_pvt::needdestroy, NONE, __show_chan_arg::numchans, sip_pvt::owner, peer_mailboxes_to_str(), sip_pvt::refer, referstatus2str(), sip_pvt::relatedpeer, S_OR, SIP_PAGE2_CALL_ONHOLD, sip_real_dst(), SIPBUFSIZE, sip_refer::status, ast_str::str, sip_pvt::subscribed, sip_pvt::subscribeuri, subscription_type2str(), __show_chan_arg::subscriptions, and sip_pvt::username.

Referenced by sip_show_channels().

{
   struct sip_pvt *cur = __cur;
   struct __show_chan_arg *arg = __arg;
   const struct sockaddr_in *dst = sip_real_dst(cur);
   
   /* XXX indentation preserved to reduce diff. Will be fixed later */
   if (cur->subscribed == NONE && !arg->subscriptions) {
      /* set if SIP transfer in progress */
      const char *referstatus = cur->refer ? referstatus2str(cur->refer->status) : "";
      char formatbuf[SIPBUFSIZE/2];
      
      ast_cli(arg->fd, FORMAT, ast_inet_ntoa(dst->sin_addr), 
            S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
            cur->callid, 
            ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0),
            cli_yesno(ast_test_flag(&cur->flags[1], SIP_PAGE2_CALL_ONHOLD)),
            cur->needdestroy ? "(d)" : "",
            cur->lastmsg ,
            referstatus
         );
      arg->numchans++;
   }
   if (cur->subscribed != NONE && arg->subscriptions) {
      struct ast_str *mailbox_str = ast_str_alloca(512);
      if (cur->subscribed == MWI_NOTIFICATION && cur->relatedpeer)
         peer_mailboxes_to_str(&mailbox_str, cur->relatedpeer);
      ast_cli(arg->fd, FORMAT4, ast_inet_ntoa(dst->sin_addr),
            S_OR(cur->username, S_OR(cur->cid_num, "(None)")), 
               cur->callid,
            /* the 'complete' exten/context is hidden in the refer_to field for subscriptions */
            cur->subscribed == MWI_NOTIFICATION ? "--" : cur->subscribeuri,
            cur->subscribed == MWI_NOTIFICATION ? "<none>" : ast_extension_state2str(cur->laststate), 
            subscription_type2str(cur->subscribed),
            cur->subscribed == MWI_NOTIFICATION ? S_OR(mailbox_str->str, "<none>") : "<none>",
            cur->expiry
         );
      arg->numchans++;
   }
   return 0;   /* don't care, we scan all channels */
}

static int show_chanstats_cb	(	void *	__cur,
		void *	__arg,
		int	flags
	)			[static]

Callback for show_chanstats.

Definition at line 14092 of file chan_sip.c.

References ast_cli(), ast_inet_ntoa(), ast_rtp_get_qosvalue(), AST_RTP_RXCOUNT, AST_RTP_RXJITTER, AST_RTP_RXPLOSS, AST_RTP_TXCOUNT, AST_RTP_TXJITTER, AST_RTP_TXPLOSS, ast_tvdiff_ms(), ast_tvnow(), ast_tvzero(), sip_pvt::callid, ast_channel::cdr, __show_chan_arg::fd, FORMAT, sip_pvt::invitestate, invitestate2string, NONE, __show_chan_arg::numchans, sip_pvt::owner, sip_pvt::rtp, sip_pvt::sa, ast_cdr::start, and sip_pvt::subscribed.

Referenced by sip_show_channelstats().

{
#define FORMAT2 "%-15.15s  %-11.11s  %-8.8s %-10.10s  %-10.10s (%-2.2s) %-6.6s %-10.10s  %-10.10s ( %%) %-6.6s\n"
#define FORMAT  "%-15.15s  %-11.11s  %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"
   struct sip_pvt *cur = __cur;
   unsigned int rxcount;
   unsigned int txcount;
   char durbuf[10];
        int duration;
        int durh, durm, durs;
   struct ast_channel *c = cur->owner;
   struct __show_chan_arg *arg = __arg;
   int fd = arg->fd;


   if (cur->subscribed != NONE) /* Subscriptions */
      return 0;   /* don't care, we scan all channels */

   if (!cur->rtp) {
      if (sipdebug)
         ast_cli(fd, "%-15.15s  %-11.11s (inv state: %s) -- %s\n", ast_inet_ntoa(cur->sa.sin_addr), cur->callid, invitestate2string[cur->invitestate].desc, "-- No RTP active");
      return 0;   /* don't care, we scan all channels */
   }
   rxcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXCOUNT);
   txcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXCOUNT);

   /* Find the duration of this channel */
   if (c && c->cdr && !ast_tvzero(c->cdr->start)) {
      duration = (int)(ast_tvdiff_ms(ast_tvnow(), c->cdr->start) / 1000);
      durh = duration / 3600;
      durm = (duration % 3600) / 60;
      durs = duration % 60;
      snprintf(durbuf, sizeof(durbuf), "%02d:%02d:%02d", durh, durm, durs);
   } else {
      durbuf[0] = '\0';
   }
   /* Print stats for every call with RTP */
   ast_cli(fd, FORMAT, 
      ast_inet_ntoa(cur->sa.sin_addr), 
      cur->callid, 
      durbuf,
      rxcount > (unsigned int) 100000 ? (unsigned int) (rxcount)/(unsigned int) 1000 : rxcount,
      rxcount > (unsigned int) 100000 ? "K":" ",
      ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS),
      rxcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) / rxcount * 100) : 0,
      ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXJITTER),
      txcount > (unsigned int) 100000 ? (unsigned int) (txcount)/(unsigned int) 1000 : txcount,
      txcount > (unsigned int) 100000 ? "K":" ",
      ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS),
      txcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS)/ txcount * 100) : 0,
      ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXJITTER)
   );
   arg->numchans++;

   return 0;   /* don't care, we scan all channels */
}

static int sip_addheader	(	struct ast_channel *	chan,
		void *	data
	)			[static]

Add a SIP header to an outbound INVITE.

Definition at line 23138 of file chan_sip.c.

References ast_channel_lock, ast_channel_unlock, ast_debug, ast_get_encoded_str(), ast_log(), ast_strlen_zero(), chan, FALSE, inbuf(), len(), LOG_WARNING, pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), and TRUE.

Referenced by load_module().

{
   int no = 0;
   int ok = FALSE;
   char varbuf[30];
   char *inbuf = data, *subbuf;
   
   if (ast_strlen_zero(inbuf)) {
      ast_log(LOG_WARNING, "This application requires the argument: Header\n");
      return 0;
   }
   ast_channel_lock(chan);

   /* Check for headers */
   while (!ok && no <= 50) {
      no++;
      snprintf(varbuf, sizeof(varbuf), "__SIPADDHEADER%.2d", no);

      /* Compare without the leading underscores */
      if ((pbx_builtin_getvar_helper(chan, (const char *) varbuf + 2) == (const char *) NULL)) {
         ok = TRUE;
      }
   }
   if (ok) {
      size_t len = strlen(inbuf);
      subbuf = alloca(len + 1);
      ast_get_encoded_str(inbuf, subbuf, len + 1);
      pbx_builtin_setvar_helper(chan, varbuf, subbuf);
      if (sipdebug) {
         ast_debug(1, "SIP Header added \"%s\" as %s\n", inbuf, varbuf);
      }
   } else {
      ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n");
   }
   ast_channel_unlock(chan);
   return 0;
}

static struct sip_pvt * sip_alloc	(	ast_string_field	callid,
		struct sockaddr_in *	sin,
		int	useglobal_nat,
		const int	intended_method
	)			[static]

Allocate sip_pvt structure, set defaults and link in the container. Returns a reference to the object so whoever uses it later must remember to release the reference.

Definition at line 6213 of file chan_sip.c.

References ao2_t_alloc, ao2_t_link, ao2_t_ref, ast_copy_flags, ast_debug, AST_LIST_HEAD_INIT_NOLOCK, ast_log(), ast_random(), ast_rtp_codec_setpref(), AST_RTP_DTMF, ast_rtp_new_with_bindaddr(), ast_rtp_set_rtpholdtimeout(), ast_rtp_set_rtpkeepalive(), ast_rtp_set_rtptimeout(), ast_rtp_setdtmf(), ast_rtp_setdtmfcompensate(), ast_rtp_setqos(), ast_sip_ouraddrfor(), ast_string_field_init, ast_string_field_set, ast_test_flag, ast_udptl_new_with_bindaddr(), ast_udptl_setqos(), ast_variables_destroy(), build_callid_pvt(), build_via(), context, default_prefs, dialogs, do_setnat(), errno, sip_pvt::fromdomain, global_flags, INITIAL_CSEQ, io, LOG_WARNING, make_our_tag(), mohinterpret, mohsuggest, NONE, parkinglot, sched, set_t38_capabilities(), sip_destroy_fn(), SIP_DTMF, SIP_DTMF_AUTO, SIP_DTMF_RFC2833, SIP_FLAGS_TO_COPY, sip_methods, SIP_NAT, SIP_NAT_ROUTE, SIP_OPTIONS, SIP_PAGE2_FLAGS_TO_COPY, SIP_PAGE2_RFC2833_COMPENSATE, SIP_PAGE2_T38SUPPORT, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_VIDEOSUPPORT, SIP_REGISTER, SIP_TRANSPORT_UDP, cfsip_methods::text, and TRUE.

Referenced by find_call(), manager_sipnotify(), sip_cli_notify(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), and transmit_register().

{
   struct sip_pvt *p;

   if (!(p = ao2_t_alloc(sizeof(*p), sip_destroy_fn, "allocate a dialog(pvt) struct")))
      return NULL;

   if (ast_string_field_init(p, 512)) {
      ao2_t_ref(p, -1, "failed to string_field_init, drop p");
      return NULL;
   }

   p->socket.fd = -1;
   p->socket.type = SIP_TRANSPORT_UDP;
   p->method = intended_method;
   p->initid = -1;
   p->waitid = -1;
   p->autokillid = -1;
   p->request_queue_sched_id = -1;
   p->t38id = -1;
   p->subscribed = NONE;
   p->stateid = -1;
   p->sessionversion_remote = -1;
   p->session_modify = TRUE;
   p->stimer = NULL;
   p->prefs = default_prefs;     /* Set default codecs for this call */

   if (intended_method != SIP_OPTIONS) {  /* Peerpoke has it's own system */
      p->timer_t1 = global_t1;   /* Default SIP retransmission timer T1 (RFC 3261) */
      p->timer_b = global_timer_b;  /* Default SIP transaction timer B (RFC 3261) */
   }

   if (!sin)
      p->ourip = internip;
   else {
      p->sa = *sin;
      ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
   }

   /* Copy global flags to this PVT at setup. */
   ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);

   p->do_history = recordhistory;

   p->branch = ast_random();  
   make_our_tag(p->tag, sizeof(p->tag));
   p->ocseq = INITIAL_CSEQ;

   if (sip_methods[intended_method].need_rtp) {
      p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
      /* If the global videosupport flag is on, we always create a RTP interface for video */
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
         p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT))
         p->trtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
         p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
      if (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp) 
            || (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && !p->trtp)) {
         ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n",
            ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "",
            ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "and text " : "", strerror(errno));
         if (p->chanvars) {
            ast_variables_destroy(p->chanvars);
            p->chanvars = NULL;
         }
         ao2_t_ref(p, -1, "failed to create RTP audio session, drop p");
         return NULL;
      }
      ast_rtp_setqos(p->rtp, global_tos_audio, global_cos_audio, "SIP RTP");
      ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
      ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
      ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
      ast_rtp_set_rtpholdtimeout(p->rtp, global_rtpholdtimeout);
      ast_rtp_set_rtpkeepalive(p->rtp, global_rtpkeepalive);
      if (p->vrtp) {
         ast_rtp_setqos(p->vrtp, global_tos_video, global_cos_video, "SIP VRTP");
         ast_rtp_setdtmf(p->vrtp, 0);
         ast_rtp_setdtmfcompensate(p->vrtp, 0);
         ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
         ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout);
         ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
      }
      if (p->trtp) {
         ast_rtp_setqos(p->trtp, global_tos_text, global_cos_text, "SIP TRTP");
         ast_rtp_setdtmf(p->trtp, 0);
         ast_rtp_setdtmfcompensate(p->trtp, 0);
      }
      if (p->udptl)
         ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
      p->maxcallbitrate = default_maxcallbitrate;
      p->autoframing = global_autoframing;
      ast_rtp_codec_setpref(p->rtp, &p->prefs);
   }

   if (useglobal_nat && sin) {
      /* Setup NAT structure according to global settings if we have an address */
      ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT);
      p->recv = *sin;
      do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
   }

   if (p->method != SIP_REGISTER)
      ast_string_field_set(p, fromdomain, default_fromdomain);
   build_via(p);
   if (!callid)
      build_callid_pvt(p);
   else
      ast_string_field_set(p, callid, callid);
   /* Assign default music on hold class */
   ast_string_field_set(p, mohinterpret, default_mohinterpret);
   ast_string_field_set(p, mohsuggest, default_mohsuggest);
   p->capability = global_capability;
   p->allowtransfer = global_allowtransfer;
   if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
       (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
      p->noncodeccapability |= AST_RTP_DTMF;
   if (p->udptl) {
      ast_copy_flags(&p->t38.t38support, &p->flags[1], SIP_PAGE2_T38SUPPORT);
      set_t38_capabilities(p);
      p->t38.jointcapability = p->t38.capability;
   }
   ast_string_field_set(p, context, default_context);
   ast_string_field_set(p, parkinglot, default_parkinglot);

   AST_LIST_HEAD_INIT_NOLOCK(&p->request_queue);

   /* Add to active dialog list */

   ao2_t_link(dialogs, p, "link pvt into dialogs table");
   
   ast_debug(1, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : p->callid, sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP");
   return p;
}

static void sip_alreadygone

(

struct sip_pvt *

dialog

)

[static]

Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging.

Definition at line 2643 of file chan_sip.c.

References sip_pvt::alreadygone, ast_debug, and sip_pvt::callid.

Referenced by handle_request_bye(), handle_request_cancel(), handle_request_invite(), handle_request_refer(), handle_response(), handle_response_invite(), retrans_pkt(), sip_indicate(), and sip_sipredirect().

{
   ast_debug(3, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
   dialog->alreadygone = 1;
}

static int sip_answer

(

struct ast_channel *

ast

)

[static]

sip_answer: Answer SIP call , send 200 OK on Invite Part of PBX interface

Definition at line 5333 of file chan_sip.c.

References ast_channel::_state, ast_debug, ast_rtp_new_source(), ast_set_flag, ast_setstate(), AST_STATE_UP, change_t38_state(), FALSE, ast_channel::name, SIP_PAGE2_DIALOG_ESTABLISHED, sip_pvt_lock, sip_pvt_unlock, T38_ENABLED, T38_PEER_DIRECT, ast_channel::tech_pvt, transmit_response_with_sdp(), try_suggested_sip_codec(), and XMIT_CRITICAL.

{
   int res = 0;
   struct sip_pvt *p = ast->tech_pvt;

   sip_pvt_lock(p);
   if (ast->_state != AST_STATE_UP) {
      try_suggested_sip_codec(p);   

      ast_setstate(ast, AST_STATE_UP);
      ast_debug(1, "SIP answering channel: %s\n", ast->name);
      if (p->t38.state == T38_PEER_DIRECT) {
         change_t38_state(p, T38_ENABLED);
      }
      ast_rtp_new_source(p->rtp);
      res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
      ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
   }
   sip_pvt_unlock(p);
   return res;
}

static int sip_call	(	struct ast_channel *	ast,
		char *	dest,
		int	timeout
	)			[static]

Initiate SIP call from PBX used from the dial() application.

Definition at line 4566 of file chan_sip.c.

References ast_channel::_state, sip_invite_param::addsipheaders, AST_CAUSE_USER_BUSY, ast_debug, AST_FORMAT_AUDIO_MASK, AST_LIST_TRAVERSE, ast_log(), AST_SCHED_REPLACE_UNREF, ast_set_flag, AST_STATE_DOWN, AST_STATE_RESERVED, ast_string_field_set, ast_translate_available_formats(), ast_var_name(), ast_var_value(), auto_congest(), buf, sip_pvt::callingpres, t38properties::capability, sip_pvt::capability, ast_channel::cid, cid_name, sip_pvt::cid_name, ast_callerid::cid_pres, dialog_ref(), dialog_unref(), ast_var_t::entries, sip_pvt::flags, ast_channel::hangupcause, INC_CALL_RINGING, sip_pvt::initid, INV_CALLING, sip_pvt::invitestate, t38properties::jointcapability, sip_pvt::jointcapability, sip_pvt::jointnoncodeccapability, LOG_WARNING, ast_channel::name, sip_pvt::noncodeccapability, sip_pvt::options, sip_pvt::prefcodec, sip_invite_param::replaces, sched, SIP_INVITE, SIP_OUTGOING, SIPBUFSIZE, sip_pvt::t38, ast_channel::tech_pvt, sip_pvt::timer_b, sip_invite_param::transfer, transmit_invite(), update_call_counter(), sip_invite_param::uri_options, sip_pvt::username, ast_channel::varshead, sip_invite_param::vxml_url, and XMIT_ERROR.

{
   int res;
   struct sip_pvt *p = ast->tech_pvt;  /* chan is locked, so the reference cannot go away */
   struct varshead *headp;
   struct ast_var_t *current;
   const char *referer = NULL;   /* SIP referrer */

   if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
      ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
      return -1;
   }

   /* Check whether there is vxml_url, distinctive ring variables */
   headp=&ast->varshead;
   AST_LIST_TRAVERSE(headp, current, entries) {
      /* Check whether there is a VXML_URL variable */
      if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
         p->options->vxml_url = ast_var_value(current);
      } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
         p->options->uri_options = ast_var_value(current);
      } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
         /* Check whether there is a variable with a name starting with SIPADDHEADER */
         p->options->addsipheaders = 1;
      } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
         /* This is a transfered call */
         p->options->transfer = 1;
      } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
         /* This is the referrer */
         referer = ast_var_value(current);
      } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
         /* We're replacing a call. */
         p->options->replaces = ast_var_value(current);
      }
   }

   res = 0;
   ast_set_flag(&p->flags[0], SIP_OUTGOING);

   if (p->options->transfer) {
      char buf[SIPBUFSIZE/2];

      if (referer) {
         if (sipdebug)
            ast_debug(3, "Call for %s transfered by %s\n", p->username, referer);
         snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
      } else 
         snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
      ast_string_field_set(p, cid_name, buf);
   } 
   ast_debug(1, "Outgoing Call for %s\n", p->username);

   res = update_call_counter(p, INC_CALL_RINGING);

   if (res == -1) {
      ast->hangupcause = AST_CAUSE_USER_BUSY;
      return res;
   }
   p->callingpres = ast->cid.cid_pres;
   p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
   p->jointnoncodeccapability = p->noncodeccapability;

   /* If there are no audio formats left to offer, punt */
   if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
      ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
      res = -1;
   } else {
      int xmitres;

      p->t38.jointcapability = p->t38.capability;
      ast_debug(2, "Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);

      xmitres = transmit_invite(p, SIP_INVITE, 1, 2);
      if (xmitres == XMIT_ERROR)
         return -1;
      p->invitestate = INV_CALLING;
   
      /* Initialize auto-congest time */
      AST_SCHED_REPLACE_UNREF(p->initid, sched, p->timer_b, auto_congest, p, 
                        dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"), 
                        dialog_unref(p, "dialog ptr dec when SCHED_REPLACE add failed"),
                        dialog_ref(p, "dialog ptr inc when SCHED_REPLACE add succeeded") );
   }
   return res;
}

static int sip_cancel_destroy

(

struct sip_pvt *

p

)

[static]

Cancel destruction of SIP dialog. Be careful as this also absorbs the reference - if you call it from within the scheduler, this might be the last reference.

Definition at line 3293 of file chan_sip.c.

References append_history, ast_sched_del(), sip_pvt::autokillid, dialog_unref(), and sched.

Referenced by cb_extensionstate(), handle_request_invite(), handle_request_subscribe(), handle_response(), handle_response_invite(), register_verify(), sip_hangup(), and sip_scheddestroy().

{
   int res = 0;
   if (p->autokillid > -1) {
      int res3;
      
      if (!(res3 = ast_sched_del(sched, p->autokillid))) {
         append_history(p, "CancelDestroy", "");
         p->autokillid = -1;
         dialog_unref(p, "dialog unrefd because autokillid is de-sched'd");
      }
   }
   return res;
}

static char * sip_cli_notify	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Cli command to send SIP notify to peer.

Definition at line 15110 of file chan_sip.c.

References ao2_t_link, ao2_t_unlink, ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_log(), ast_set_flag, ast_sip_ouraddrfor(), ast_variable_browse(), build_callid_pvt(), build_via(), CLI_FAILURE, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, ast_cli_entry::command, complete_sipnotify(), create_addr(), dialog_ref(), dialog_unlink_all(), dialog_unref(), dialogs, ast_cli_args::fd, sip_pvt::flags, ast_cli_args::line, LOG_WARNING, ast_cli_args::n, notify_types, sip_pvt::ourip, ast_cli_args::pos, sip_pvt::sa, sip_alloc(), SIP_NOTIFY, SIP_OUTGOING, sip_scheddestroy(), SIP_TRANS_TIMEOUT, transmit_notify_custom(), TRUE, ast_cli_entry::usage, and ast_cli_args::word.

{
   struct ast_variable *varlist;
   int i;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip notify";
      e->usage =
         "Usage: sip notify <type> <peer> [<peer>...]\n"
         "       Send a NOTIFY message to a SIP peer or peers\n"
         "       Message types are defined in sip_notify.conf\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sipnotify(a->line, a->word, a->pos, a->n);
   }

   if (a->argc < 4)
      return CLI_SHOWUSAGE;

   if (!notify_types) {
      ast_cli(a->fd, "No %s file found, or no types listed there\n", notify_config);
      return CLI_FAILURE;
   }

   varlist = ast_variable_browse(notify_types, a->argv[2]);

   if (!varlist) {
      ast_cli(a->fd, "Unable to find notify type '%s'\n", a->argv[2]);
      return CLI_FAILURE;
   }

   for (i = 3; i < a->argc; i++) {
      struct sip_pvt *p;

      if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY))) {
         ast_log(LOG_WARNING, "Unable to build sip pvt data for notify (memory/socket error)\n");
         return CLI_FAILURE;
      }

      if (create_addr(p, a->argv[i], NULL, 0)) {
         /* Maybe they're not registered, etc. */
         dialog_unlink_all(p, TRUE, TRUE);
         dialog_unref(p, "unref dialog inside for loop" );
         /* sip_destroy(p); */
         ast_cli(a->fd, "Could not create address for '%s'\n", a->argv[i]);
         continue;
      }

      /* Notify is outgoing call */
      ast_set_flag(&p->flags[0], SIP_OUTGOING);

      /* Recalculate our side, and recalculate Call ID */
      ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
      build_via(p);
      ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
      build_callid_pvt(p);
      ao2_t_link(dialogs, p, "Linking in new name");
      ast_cli(a->fd, "Sending NOTIFY of type '%s' to '%s'\n", a->argv[2], a->argv[i]);
      dialog_ref(p, "bump the count of p, which transmit_sip_request will decrement.");
      sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
      transmit_notify_custom(p, varlist);
   }

   return CLI_SUCCESS;
}

static int sip_debug_test_addr

(

const struct sockaddr_in *

addr

)

[inline, static]

See if we pass debug IP filter.

Definition at line 2782 of file chan_sip.c.

Referenced by check_peer_ok(), handle_request_do(), and sip_debug_test_pvt().

{
   if (!sipdebug)
      return 0;
   if (debugaddr.sin_addr.s_addr) {
      if (((ntohs(debugaddr.sin_port) != 0)
         && (debugaddr.sin_port != addr->sin_port))
         || (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
         return 0;
   }
   return 1;
}

static int sip_debug_test_pvt

(

struct sip_pvt *

p

)

[inline, static]

Test PVT for debugging output.

Definition at line 2811 of file chan_sip.c.

References sip_debug_test_addr(), and sip_real_dst().

Referenced by __sip_destroy(), add_sdp(), build_route(), check_via(), do_register_auth(), get_also_info(), get_destination(), get_rdnis(), get_refer_info(), handle_incoming(), process_sdp(), receive_message(), retrans_pkt(), send_request(), send_response(), set_destination(), sip_scheddestroy(), sip_sendhtml(), sip_sendtext(), and transmit_register().

{
   if (!sipdebug)
      return 0;
   return sip_debug_test_addr(sip_real_dst(p));
}

static struct sip_pvt * sip_destroy

(

struct sip_pvt *

p

)

[static]

Destroy SIP call structure. Make it return NULL so the caller can do things like foo = sip_destroy(foo); and reduce the chance of bugs due to dangling pointers.

Definition at line 4956 of file chan_sip.c.

References __sip_destroy(), ast_debug, sip_pvt::callid, and TRUE.

Referenced by sip_destroy_fn().

{
   ast_debug(3, "Destroying SIP dialog %s\n", p->callid);
   __sip_destroy(p, TRUE, TRUE);
   return NULL;
}

static void sip_destroy_fn

(

void *

p

)

[static]

Definition at line 4946 of file chan_sip.c.

References sip_destroy().

Referenced by sip_alloc().

{
   sip_destroy(p);
}

static void sip_destroy_peer

(

struct sip_peer *

peer

)

[static]

Destroy peer object from memory.

Definition at line 3857 of file chan_sip.c.

References ao2_ref, ast_atomic_fetchadd_int(), ast_debug, ast_dnsmgr_release(), ast_free_ha(), ast_variables_destroy(), sip_peer::auth, sip_peer::call, sip_peer::chanvars, clear_peer_mailboxes(), clear_realm_authentication(), dialog_unlink_all(), dialog_unref(), sip_peer::dnsmgr, FALSE, sip_peer::ha, sip_peer::is_realtime, sip_peer::mwipvt, sip_peer::name, sip_peer::outboundproxy, register_peer_exten(), sip_peer::selfdestruct, sip_peer::socket, sip_socket::tcptls_session, and TRUE.

Referenced by sip_destroy_peer_fn().

{
   ast_debug(3, "Destroying SIP peer %s\n", peer->name);
   if (peer->outboundproxy)
      ao2_ref(peer->outboundproxy, -1);
   peer->outboundproxy = NULL;

   /* Delete it, it needs to disappear */
   if (peer->call) {
      dialog_unlink_all(peer->call, TRUE, TRUE);
      peer->call = dialog_unref(peer->call, "peer->call is being unset");
   }
   

   if (peer->mwipvt) {  /* We have an active subscription, delete it */
      dialog_unlink_all(peer->mwipvt, TRUE, TRUE);
      peer->mwipvt = dialog_unref(peer->mwipvt, "unreffing peer->mwipvt");
   }
   
   if (peer->chanvars) {
      ast_variables_destroy(peer->chanvars);
      peer->chanvars = NULL;
   }
   
   register_peer_exten(peer, FALSE);
   ast_free_ha(peer->ha);
   if (peer->selfdestruct)
      ast_atomic_fetchadd_int(&apeerobjs, -1);
   else if (peer->is_realtime) {
      ast_atomic_fetchadd_int(&rpeerobjs, -1);
      ast_debug(3, "-REALTIME- peer Destroyed. Name: %s. Realtime Peer objects: %d\n", peer->name, rpeerobjs);
   } else
      ast_atomic_fetchadd_int(&speerobjs, -1);
   clear_realm_authentication(peer->auth);
   peer->auth = NULL;
   if (peer->dnsmgr)
      ast_dnsmgr_release(peer->dnsmgr);
   clear_peer_mailboxes(peer);

   if (peer->socket.tcptls_session) {
      ao2_ref(peer->socket.tcptls_session, -1);
      peer->socket.tcptls_session = NULL;
   }
}

static void sip_destroy_peer_fn

(

void *

peer

)

[static]

Definition at line 3851 of file chan_sip.c.

References sip_destroy_peer().

Referenced by build_peer(), and temp_peer().

{
   sip_destroy_peer(peer);
}

static int sip_devicestate

(

void *

data

)

[static]

Part of PBX channel interface.

Note:

Return values:---

If we have qualify on and the device is not reachable, regardless of registration state we return AST_DEVICE_UNAVAILABLE

For peers with call limit:

• not registered AST_DEVICE_UNAVAILABLE
• registered, no call AST_DEVICE_NOT_INUSE
• registered, active calls AST_DEVICE_INUSE
• registered, call limit reached AST_DEVICE_BUSY
• registered, onhold AST_DEVICE_ONHOLD
• registered, ringing AST_DEVICE_RINGING

For peers without call limit:

• not registered AST_DEVICE_UNAVAILABLE
• registered AST_DEVICE_NOT_INUSE
• fixed IP (!dynamic) AST_DEVICE_NOT_INUSE

Peers that does not have a known call and can't be reached by OPTIONS

• unreachable AST_DEVICE_UNAVAILABLE

If we return AST_DEVICE_UNKNOWN, the device state engine will try to find out a state by walking the channel list.

The queue system (app_queue.c) treats a member as "active" if devicestate is != AST_DEVICE_UNAVAILBALE && != AST_DEVICE_INVALID

When placing a call to the queue member, queue system sets a member to busy if != AST_DEVICE_NOT_INUSE and != AST_DEVICE_UNKNOWN

Definition at line 20909 of file chan_sip.c.

References sip_peer::addr, ast_debug, AST_DEVICE_BUSY, AST_DEVICE_INUSE, AST_DEVICE_INVALID, AST_DEVICE_NOT_INUSE, AST_DEVICE_ONHOLD, AST_DEVICE_RINGING, AST_DEVICE_RINGINUSE, AST_DEVICE_UNAVAILABLE, AST_DEVICE_UNKNOWN, ast_strdupa, sip_peer::busy_level, sip_peer::call_limit, sip_peer::defaddr, FALSE, find_peer(), FINDALLDEVICES, sip_peer::inRinging, sip_peer::inUse, sip_peer::lastms, sip_peer::maxms, sip_peer::onHold, TRUE, and unref_peer().

{
   char *host;
   char *tmp;
   struct sip_peer *p;

   int res = AST_DEVICE_INVALID;

   /* make sure data is not null. Maybe unnecessary, but better be safe */
   host = ast_strdupa(data ? data : "");
   if ((tmp = strchr(host, '@')))
      host = tmp + 1;

   ast_debug(3, "Checking device state for peer %s\n", host);

   /* If find_peer asks for a realtime peer, then this breaks rtautoclear.  This
    * is because when a peer tries to autoexpire, the last thing it does is to
    * queue up an event telling the system that the devicestate has changed
    * (presumably to unavailable).  If we ask for a realtime peer here, this would
    * load it BACK into memory, thus defeating the point of trying to clear dead
    * hosts out of memory.
    */
   if ((p = find_peer(host, NULL, FALSE, FINDALLDEVICES, TRUE))) {
      if (p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) {
         /* we have an address for the peer */
      
         /* Check status in this order
            - Hold
            - Ringing
            - Busy (enforced only by call limit)
            - Inuse (we have a call)
            - Unreachable (qualify)
            If we don't find any of these state, report AST_DEVICE_NOT_INUSE
            for registered devices */

         if (p->onHold)
            /* First check for hold or ring states */
            res = AST_DEVICE_ONHOLD;
         else if (p->inRinging) {
            if (p->inRinging == p->inUse)
               res = AST_DEVICE_RINGING;
            else
               res = AST_DEVICE_RINGINUSE;
         } else if (p->call_limit && (p->inUse == p->call_limit))
            /* check call limit */
            res = AST_DEVICE_BUSY;
         else if (p->call_limit && p->busy_level && p->inUse >= p->busy_level)
            /* We're forcing busy before we've reached the call limit */
            res = AST_DEVICE_BUSY;
         else if (p->call_limit && p->inUse)
            /* Not busy, but we do have a call */
            res = AST_DEVICE_INUSE;
         else if (p->maxms && ((p->lastms > p->maxms) || (p->lastms < 0))) 
            /* We don't have a call. Are we reachable at all? Requires qualify= */
            res = AST_DEVICE_UNAVAILABLE;
         else  /* Default reply if we're registered and have no other data */
            res = AST_DEVICE_NOT_INUSE;
      } else {
         /* there is no address, it's unavailable */
         res = AST_DEVICE_UNAVAILABLE;
      }
      unref_peer(p, "unref_peer, from sip_devicestate, release ref from find_peer");
   } else {
      res = AST_DEVICE_UNKNOWN;
   }

   return res;
}

static char * sip_do_debug	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Turn on SIP debugging (CLI command).

Note:

this can be a special debug command - "sip debug text" or something

Definition at line 15067 of file chan_sip.c.

References ast_cli_args::argc, ast_cli_entry::args, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_peer(), ast_cli_args::fd, ast_cli_args::line, ast_cli_args::n, ast_cli_args::pos, sip_debug_console, sip_do_debug_ip(), sip_do_debug_peer(), strcasestr(), ast_cli_entry::usage, and ast_cli_args::word.

{
   int oldsipdebug = sipdebug & sip_debug_console;
   char *what;

   if (cmd == CLI_INIT) {
      e->command = "sip set debug {on|off|ip|peer}";
      e->usage =
         "Usage: sip set debug {off|on|ip addr[:port]|peer peername}\n"
         "       Globally disables dumping of SIP packets,\n"
         "       or enables it either globally or for a (single)\n"
         "       IP address or registered peer.\n";
      return NULL;
   } else if (cmd == CLI_GENERATE) {
      if (a->pos == 4 && strcasestr(a->line, " peer")) /* XXX should check on argv too */
         return complete_sip_peer(a->word, a->n, 0);
      return NULL;
        }

   what = a->argv[e->args-1];      /* guaranteed to exist */
   if (a->argc == e->args) {       /* on/off */
      if (!strcasecmp(what, "on")) {
         sipdebug |= sip_debug_console;
         sipdebug_text = 1;   /*! \note this can be a special debug command - "sip debug text" or something */
         memset(&debugaddr, 0, sizeof(debugaddr));
         ast_cli(a->fd, "SIP Debugging %senabled\n", oldsipdebug ? "re-" : "");
         return CLI_SUCCESS;
      } else if (!strcasecmp(what, "off")) {
         sipdebug &= ~sip_debug_console;
         sipdebug_text = 0;
         ast_cli(a->fd, "SIP Debugging Disabled\n");
         return CLI_SUCCESS;
      }
   } else if (a->argc == e->args +1) {/* ip/peer */
      if (!strcasecmp(what, "ip"))
         return sip_do_debug_ip(a->fd, a->argv[e->args]);
      else if (!strcasecmp(what, "peer"))
         return sip_do_debug_peer(a->fd, a->argv[e->args]);
   }
   return CLI_SHOWUSAGE;   /* default, failure */
}

static char * sip_do_debug_ip	(	int	fd,
		char *	arg
	)			[static]

Enable SIP Debugging for a single IP.

Definition at line 15017 of file chan_sip.c.

References ahp, ast_cli(), ast_gethostbyname(), ast_inet_ntoa(), CLI_SHOWUSAGE, CLI_SUCCESS, hp, sip_debug_console, and strsep().

Referenced by sip_do_debug().

{
   struct hostent *hp;
   struct ast_hostent ahp;
   int port = 0;
   char *p;

   p = arg;
   strsep(&p, ":");
   if (p)
      port = atoi(p);
   hp = ast_gethostbyname(arg, &ahp);
   if (hp == NULL)
      return CLI_SHOWUSAGE;

   debugaddr.sin_family = AF_INET;
   memcpy(&debugaddr.sin_addr, hp->h_addr, sizeof(debugaddr.sin_addr));
   debugaddr.sin_port = htons(port);
   if (port == 0)
      ast_cli(fd, "SIP Debugging Enabled for IP: %s\n", ast_inet_ntoa(debugaddr.sin_addr));
   else
      ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(debugaddr.sin_addr), port);

   sipdebug |= sip_debug_console;

   return CLI_SUCCESS;
}

static char * sip_do_debug_peer	(	int	fd,
		char *	arg
	)			[static]

Turn on SIP debugging for a given peer.

Definition at line 15046 of file chan_sip.c.

References sip_peer::addr, ast_cli(), ast_inet_ntoa(), CLI_SUCCESS, FALSE, find_peer(), FINDPEERS, sip_debug_console, TRUE, and unref_peer().

Referenced by sip_do_debug().

{
   struct sip_peer *peer = find_peer(arg, NULL, TRUE, FINDPEERS, FALSE);
   if (!peer)
      ast_cli(fd, "No such peer '%s'\n", arg);
   else if (peer->addr.sin_addr.s_addr == 0)
      ast_cli(fd, "Unable to get IP address of peer '%s'\n", arg);
   else {
      debugaddr.sin_family = AF_INET;
      debugaddr.sin_addr = peer->addr.sin_addr;
      debugaddr.sin_port = peer->addr.sin_port;
      ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n",
         ast_inet_ntoa(debugaddr.sin_addr), ntohs(debugaddr.sin_port));
      sipdebug |= sip_debug_console;
   }
   if (peer)
      unref_peer(peer, "sip_do_debug_peer: unref_peer, from find_peer call");
   return CLI_SUCCESS;
}

static char * sip_do_history_deprecated	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Enable/Disable SIP History logging (CLI) - deprecated. use sip_set_history instead.

Definition at line 15178 of file chan_sip.c.

References ast_cli_args::argc, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, FALSE, ast_cli_args::fd, TRUE, and ast_cli_entry::usage.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip history [off]";
      e->usage =
         "Usage: sip history [off]\n"
         "       Enables/Disables recording of SIP dialog history for debugging purposes.\n"
         "       Use 'sip show history' to view the history of a call number.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc < 2 || a->argc > 3) {
      return CLI_SHOWUSAGE;
   }
   if (a->argc == 2) {
      recordhistory = TRUE;
      ast_cli(a->fd, "SIP History Recording Enabled (use 'sip show history')\n");
   } else {
      if (strncasecmp(a->argv[2], "off", 3))
         return CLI_SHOWUSAGE;
      recordhistory = FALSE;
      ast_cli(a->fd, "SIP History Recording Disabled\n");
   }
   return CLI_SUCCESS;
}

static int sip_do_reload

(

enum channelreloadreason

reason

)

[static]

Reload module.

Definition at line 23294 of file chan_sip.c.

References ao2_t_callback, ast_debug, ast_sched_dump(), OBJ_MULTIPLE, OBJ_NODATA, OBJ_UNLINK, peer_is_marked(), reload_config(), sched, sip_poke_all_peers(), and sip_send_all_registers().

Referenced by do_monitor().

{
   time_t start_poke, end_poke;
   
   reload_config(reason);
   ast_sched_dump(sched);

   start_poke = time(0);
   /* Prune peers who still are supposed to be deleted */
   ao2_t_callback(peers, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE, peer_is_marked, 0, 
         "callback to remove marked peers");
   
   ast_debug(4, "--------------- Done destroying pruned peers\n");

   /* Send qualify (OPTIONS) to all peers */
   sip_poke_all_peers();

   /* Register with all services */
   sip_send_all_registers();
   end_poke = time(0);
   
   ast_debug(4, "do_reload finished. peer poke/prune reg contact time = %d sec.\n", (int)(end_poke-start_poke));

   ast_debug(4, "--------------- SIP reload done\n");

   return 0;
}

static int sip_dtmfmode	(	struct ast_channel *	chan,
		void *	data
	)			[static]

Set the DTMFmode for an outbound SIP call (application).

Definition at line 23080 of file chan_sip.c.

References ast_channel_lock, ast_channel_unlock, ast_clear_flag, ast_dsp_free(), ast_dsp_new(), ast_dsp_set_features(), ast_log(), AST_RTP_DTMF, ast_rtp_setdtmf(), ast_set_flag, ast_test_flag, chan, DSP_FEATURE_DIGIT_DETECT, sip_pvt::flags, IS_SIP_TECH, sip_pvt::jointnoncodeccapability, LOG_WARNING, sip_pvt::rtp, SIP_DTMF, SIP_DTMF_INBAND, SIP_DTMF_INFO, SIP_DTMF_RFC2833, SIP_DTMF_SHORTINFO, sip_pvt_lock, sip_pvt_unlock, ast_channel::tech, ast_channel::tech_pvt, and sip_pvt::vad.

Referenced by load_module().

{
   struct sip_pvt *p;
   char *mode = data;

   if (!data) {
      ast_log(LOG_WARNING, "This application requires the argument: info, inband, rfc2833\n");
      return 0;
   }
   ast_channel_lock(chan);
   if (!IS_SIP_TECH(chan->tech)) {
      ast_log(LOG_WARNING, "Call this application only on SIP incoming calls\n");
      ast_channel_unlock(chan);
      return 0;
   }
   p = chan->tech_pvt;
   if (!p) {
      ast_channel_unlock(chan);
      return 0;
   }
   sip_pvt_lock(p);
   if (!strcasecmp(mode, "info")) {
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      ast_set_flag(&p->flags[0], SIP_DTMF_INFO);
      p->jointnoncodeccapability &= ~AST_RTP_DTMF;
   } else if (!strcasecmp(mode, "shortinfo")) {
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      ast_set_flag(&p->flags[0], SIP_DTMF_SHORTINFO);
      p->jointnoncodeccapability &= ~AST_RTP_DTMF;
   } else if (!strcasecmp(mode, "rfc2833")) {
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
      p->jointnoncodeccapability |= AST_RTP_DTMF;
   } else if (!strcasecmp(mode, "inband")) { 
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
      p->jointnoncodeccapability &= ~AST_RTP_DTMF;
   } else
      ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n", mode);
   if (p->rtp)
      ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
   if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
      if (!p->vad) {
         p->vad = ast_dsp_new();
         ast_dsp_set_features(p->vad, DSP_FEATURE_DIGIT_DETECT);
      }
   } else {
      if (p->vad) {
         ast_dsp_free(p->vad);
         p->vad = NULL;
      }
   }
   sip_pvt_unlock(p);
   ast_channel_unlock(chan);
   return 0;
}

static void sip_dump_history

(

struct sip_pvt *

dialog

)

[static]

Dump SIP history to debug log file at end of lifespan for SIP dialog.

Definition at line 14802 of file chan_sip.c.

References ast_debug, AST_LIST_TRAVERSE, ast_log(), sip_pvt::callid, sip_history::event, sip_pvt::history, sip_history::list, LOG_NOTICE, option_debug, and sip_pvt::subscribed.

Referenced by __sip_destroy().

{
   int x = 0;
   struct sip_history *hist;
   static int errmsg = 0;

   if (!dialog)
      return;

   if (!option_debug && !sipdebug) {
      if (!errmsg) {
         ast_log(LOG_NOTICE, "You must have debugging enabled (SIP or Asterisk) in order to dump SIP history.\n");
         errmsg = 1;
      }
      return;
   }

   ast_debug(1, "\n---------- SIP HISTORY for '%s' \n", dialog->callid);
   if (dialog->subscribed)
      ast_debug(1, "  * Subscription\n");
   else
      ast_debug(1, "  * SIP Call\n");
   if (dialog->history)
      AST_LIST_TRAVERSE(dialog->history, hist, list)
         ast_debug(1, "  %-3.3d. %s\n", ++x, hist->event);
   if (!x)
      ast_debug(1, "Call '%s' has no history\n", dialog->callid);
   ast_debug(1, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid);
}

static int sip_fixup	(	struct ast_channel *	oldchan,
		struct ast_channel *	newchan
	)			[static]

sip_fixup: Fix up a channel: If a channel is consumed, this is called. Basically update any ->owner links

Definition at line 5470 of file chan_sip.c.

References append_history, ast_debug, AST_FLAG_ZOMBIE, ast_log(), ast_test_flag, sip_pvt::callid, LOG_WARNING, ast_channel::name, sip_pvt::owner, sip_pvt_lock, sip_pvt_unlock, sip_set_rtp_peer(), and ast_channel::tech_pvt.

{
   int ret = -1;
   struct sip_pvt *p;

   if (newchan && ast_test_flag(newchan, AST_FLAG_ZOMBIE))
      ast_debug(1, "New channel is zombie\n");
   if (oldchan && ast_test_flag(oldchan, AST_FLAG_ZOMBIE))
      ast_debug(1, "Old channel is zombie\n");

   if (!newchan || !newchan->tech_pvt) {
      if (!newchan)
         ast_log(LOG_WARNING, "No new channel! Fixup of %s failed.\n", oldchan->name);
      else
         ast_log(LOG_WARNING, "No SIP tech_pvt! Fixup of %s failed.\n", oldchan->name);
      return -1;
   }
   p = newchan->tech_pvt;

   sip_pvt_lock(p);
   append_history(p, "Masq", "Old channel: %s\n", oldchan->name);
   append_history(p, "Masq (cont)", "...new owner: %s\n", newchan->name);
   if (p->owner != oldchan)
      ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
   else {
      p->owner = newchan;
      /* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native
         RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be
         able to do this if the masquerade happens before the bridge breaks (e.g., AMI
         redirect of both channels). Note that a channel can not be masqueraded *into*
         a native bridge. So there is no danger that this breaks a native bridge that
         should stay up. */
      sip_set_rtp_peer(newchan, NULL, NULL, 0, 0, 0);
      ret = 0;
   }
   ast_debug(3, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name);

   sip_pvt_unlock(p);
   return ret;
}

static const char * sip_get_callid

(

struct ast_channel *

chan

)

[static]

Deliver SIP call ID for the call.

Definition at line 3718 of file chan_sip.c.

References chan, and ast_channel::tech_pvt.

{
   return chan->tech_pvt ? ((struct sip_pvt *) chan->tech_pvt)->callid : "";
}

static int sip_get_codec

(

struct ast_channel *

chan

)

[static]

Return SIP UA's codec (part of the RTP interface).

Definition at line 23237 of file chan_sip.c.

References sip_pvt::capability, chan, sip_pvt::jointcapability, and ast_channel::tech_pvt.

{
   struct sip_pvt *p = chan->tech_pvt;
   return p->jointcapability ? p->jointcapability : p->capability;   
}

static enum ast_rtp_get_result sip_get_rtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp **	rtp
	)			[static]

Returns null if we can't reinvite audio (part of RTP interface).

Definition at line 22914 of file chan_sip.c.

References AST_JB_FORCED, AST_RTP_GET_FAILED, ast_rtp_getnat(), AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_test_flag, chan, sip_pvt::flags, global_jbconf, sip_pvt::rtp, SIP_CAN_REINVITE, SIP_CAN_REINVITE_NAT, sip_pvt_lock, sip_pvt_unlock, and ast_channel::tech_pvt.

{
   struct sip_pvt *p = NULL;
   enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;

   if (!(p = chan->tech_pvt))
      return AST_RTP_GET_FAILED;

   sip_pvt_lock(p);
   if (!(p->rtp)) {
      sip_pvt_unlock(p);
      return AST_RTP_GET_FAILED;
   }

   *rtp = p->rtp;

   if (ast_rtp_getnat(*rtp) && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT))
      res = AST_RTP_TRY_PARTIAL;
   else if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
      res = AST_RTP_TRY_NATIVE;
   else if (ast_test_flag(&global_jbconf, AST_JB_FORCED))
      res = AST_RTP_GET_FAILED;

   sip_pvt_unlock(p);

   return res;
}

static enum ast_rtp_get_result sip_get_trtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp **	rtp
	)			[static]

Returns null if we can't reinvite text (part of RTP interface).

Definition at line 22968 of file chan_sip.c.

References AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_test_flag, chan, sip_pvt::flags, sip_pvt::rtp, SIP_CAN_REINVITE, sip_pvt_lock, sip_pvt_unlock, ast_channel::tech_pvt, and sip_pvt::trtp.

{
   struct sip_pvt *p = NULL;
   enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
   
   if (!(p = chan->tech_pvt))
      return AST_RTP_GET_FAILED;

   sip_pvt_lock(p);
   if (!(p->trtp)) {
      sip_pvt_unlock(p);
      return AST_RTP_GET_FAILED;
   }

   *rtp = p->trtp;

   if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
      res = AST_RTP_TRY_NATIVE;

   sip_pvt_unlock(p);

   return res;
}

static struct ast_udptl * sip_get_udptl_peer

(

struct ast_channel *

chan

)

[static]

Definition at line 22870 of file chan_sip.c.

References ast_test_flag, chan, sip_pvt::flags, SIP_CAN_REINVITE, sip_pvt_lock, sip_pvt_unlock, ast_channel::tech_pvt, and sip_pvt::udptl.

{
   struct sip_pvt *p;
   struct ast_udptl *udptl = NULL;
   
   p = chan->tech_pvt;
   if (!p)
      return NULL;
   
   sip_pvt_lock(p);
   if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
      udptl = p->udptl;
   sip_pvt_unlock(p);
   return udptl;
}

static enum ast_rtp_get_result sip_get_vrtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp **	rtp
	)			[static]

Returns null if we can't reinvite video (part of RTP interface).

Definition at line 22943 of file chan_sip.c.

References AST_RTP_GET_FAILED, AST_RTP_TRY_NATIVE, AST_RTP_TRY_PARTIAL, ast_test_flag, chan, sip_pvt::flags, sip_pvt::rtp, SIP_CAN_REINVITE, sip_pvt_lock, sip_pvt_unlock, ast_channel::tech_pvt, and sip_pvt::vrtp.

{
   struct sip_pvt *p = NULL;
   enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
   
   if (!(p = chan->tech_pvt))
      return AST_RTP_GET_FAILED;

   sip_pvt_lock(p);
   if (!(p->vrtp)) {
      sip_pvt_unlock(p);
      return AST_RTP_GET_FAILED;
   }

   *rtp = p->vrtp;

   if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
      res = AST_RTP_TRY_NATIVE;

   sip_pvt_unlock(p);

   return res;
}

static int sip_hangup

(

struct ast_channel *

ast

)

[static]

sip_hangup: Hangup SIP call Part of PBX interface, called from ast_hangup

Definition at line 5126 of file chan_sip.c.

References __sip_pretend_ack(), ast_channel::_state, sip_pvt::alreadygone, sip_pvt::answered_elsewhere, append_history, ast_bridged_channel(), ast_cause2str(), ast_clear_flag, ast_debug, ast_dsp_free(), AST_FLAG_ANSWERED_ELSEWHERE, AST_FLAG_ZOMBIE, ast_log(), ast_module_unref(), ast_rtp_get_quality(), ast_rtp_set_vars(), AST_SCHED_DEL_UNREF, ast_set_flag, ast_state2str(), AST_STATE_UP, ast_strlen_zero(), ast_test_flag, sip_pvt::autokillid, sip_pvt::callid, DEC_CALL_LIMIT, DEFAULT_TRANS_TIMEOUT, dialog_unref(), FALSE, sip_pvt::flags, hangup_cause2sip(), ast_channel::hangupcause, sip_pvt::hangupcause, INV_CALLING, INV_CANCELLED, INV_COMPLETED, INV_TERMINATED, sip_pvt::invitestate, IS_SIP_TECH, LOG_WARNING, ast_channel::name, sip_pvt::needdestroy, sip_pvt::owner, pbx_builtin_setvar_helper(), sip_pvt::refer, sip_pvt::rtp, RTPQOS_SUMMARY, sched, SIP_BYE, SIP_CANCEL, sip_cancel_destroy(), SIP_DEFER_BYE_ON_TRANSFER, SIP_INC_COUNT, SIP_NEEDREINVITE, SIP_OUTGOING, SIP_PAGE2_CALL_ONHOLD, SIP_PENDINGBYE, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), stop_media_flows(), stop_session_timer(), ast_channel::tech, ast_channel::tech_pvt, transmit_request(), transmit_request_with_auth(), transmit_response_reliable(), TRUE, update_call_counter(), sip_pvt::username, sip_pvt::vad, and XMIT_RELIABLE.

{
   struct sip_pvt *p = ast->tech_pvt;
   int needcancel = FALSE;
   int needdestroy = 0;
   struct ast_channel *oldowner = ast;

   if (!p) {
      ast_debug(1, "Asked to hangup channel that was not connected\n");
      return 0;
   }
   if (ast_test_flag(ast, AST_FLAG_ANSWERED_ELSEWHERE)) {
      ast_debug(1, "This call was answered elsewhere");
      append_history(p, "Cancel", "Call answered elsewhere");
      p->answered_elsewhere = TRUE;
   }

   /* Store hangupcause locally in PVT so we still have it before disconnect */
   if (p->owner)
      p->hangupcause = p->owner->hangupcause;

   if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
      if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
         if (sipdebug)
            ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
         update_call_counter(p, DEC_CALL_LIMIT);
      }
      ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
      if (p->autokillid > -1 && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */
      p->needdestroy = 0;
      p->owner->tech_pvt = dialog_unref(p->owner->tech_pvt, "unref p->owner->tech_pvt");
      sip_pvt_lock(p);
      p->owner = NULL;  /* Owner will be gone after we return, so take it away */
      sip_pvt_unlock(p);
      return 0;
   }

   if (ast_test_flag(ast, AST_FLAG_ZOMBIE)) {
      if (p->refer)
         ast_debug(1, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
      else
         ast_debug(1, "Hanging up zombie call. Be scared.\n");
   } else
      ast_debug(1, "Hangup call %s, SIP callid %s\n", ast->name, p->callid);

   sip_pvt_lock(p);
   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
      if (sipdebug)
         ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
      update_call_counter(p, DEC_CALL_LIMIT);
   }

   /* Determine how to disconnect */
   if (p->owner != ast) {
      ast_log(LOG_WARNING, "Huh?  We aren't the owner? Can't hangup call.\n");
      sip_pvt_unlock(p);
      return 0;
   }
   /* If the call is not UP, we need to send CANCEL instead of BYE */
   /* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
   if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) {
      needcancel = TRUE;
      ast_debug(4, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
   }

   stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */

   append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->hangupcause) : "Unknown");

   /* Disconnect */
   if (p->vad)
      ast_dsp_free(p->vad);

   p->owner = NULL;
   ast->tech_pvt = dialog_unref(ast->tech_pvt, "unref ast->tech_pvt");

   ast_module_unref(ast_module_info->self);
   /* Do not destroy this pvt until we have timeout or
      get an answer to the BYE or INVITE/CANCEL 
      If we get no answer during retransmit period, drop the call anyway.
      (Sorry, mother-in-law, you can't deny a hangup by sending
      603 declined to BYE...)
   */
   if (p->alreadygone)
      needdestroy = 1;  /* Set destroy flag at end of this function */
   else if (p->invitestate != INV_CALLING)
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

   /* Start the process if it's not already started */
   if (!p->alreadygone && p->initreq.data && !ast_strlen_zero(p->initreq.data->str)) {
      if (needcancel) { /* Outgoing call, not up */
         if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
            /* stop retransmitting an INVITE that has not received a response */
            __sip_pretend_ack(p);

            /* if we can't send right now, mark it pending */
            if (p->invitestate == INV_CALLING) {
               /* We can't send anything in CALLING state */
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
               /* Do we need a timer here if we don't hear from them at all? */
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
               append_history(p, "DELAY", "Not sending cancel, waiting for timeout");
            } else {
               p->invitestate = INV_CANCELLED;
               /* Send a new request: CANCEL */
               transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
               /* Actually don't destroy us yet, wait for the 487 on our original 
                  INVITE, but do set an autodestruct just in case we never get it. */
               needdestroy = 0;
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
         } else { /* Incoming call, not up */
            const char *res;
            if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
               transmit_response_reliable(p, res, &p->initreq);
            else 
               transmit_response_reliable(p, "603 Declined", &p->initreq);
            p->invitestate = INV_TERMINATED;
         }
      } else { /* Call is in UP state, send BYE */
         if (p->stimer->st_active == TRUE) {
            stop_session_timer(p);
         }

         if (!p->pendinginvite) {
            struct ast_channel *bridge = ast_bridged_channel(oldowner);
            char *audioqos = "";
            char *videoqos = "";
            char *textqos = "";

            if (p->rtp)
               ast_rtp_set_vars(oldowner, p->rtp);

            if (bridge) {
               struct sip_pvt *q = bridge->tech_pvt;

               if (IS_SIP_TECH(bridge->tech) && q && q->rtp)
                  ast_rtp_set_vars(bridge, q->rtp);
            }

            if (p->vrtp)
               videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY);
            if (p->trtp)
               textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY);
            /* Send a hangup */
            transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);

            /* Get RTCP quality before end of call */
            if (p->do_history) {
               if (p->rtp)
                  append_history(p, "RTCPaudio", "Quality:%s", audioqos);
               if (p->vrtp)
                  append_history(p, "RTCPvideo", "Quality:%s", videoqos);
               if (p->trtp)
                  append_history(p, "RTCPtext", "Quality:%s", textqos);
            }
            if (p->rtp && oldowner)
               pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos);
            if (p->vrtp && oldowner)
               pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos);
            if (p->trtp && oldowner)
               pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", textqos);
         } else {
            /* Note we will need a BYE when this all settles out
               but we can't send one while we have "INVITE" outstanding. */
            ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
            ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE); 
            AST_SCHED_DEL_UNREF(sched, p->waitid, dialog_unref(p, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));
            if (sip_cancel_destroy(p))
               ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
         }
      }
   }
   if (needdestroy)
      p->needdestroy = 1;
   sip_pvt_unlock(p);
   return 0;
}

static int sip_indicate	(	struct ast_channel *	ast,
		int	condition,
		const void *	data,
		size_t	datalen
	)			[static]

Play indication to user With SIP a lot of indications is sent as messages, letting the device play the indication - busy signal, congestion etc.

Returns:

-1 to force ast_indicate to send indication in audio, 0 if SIP can handle the indication by sending a message

Definition at line 5581 of file chan_sip.c.

References ast_channel::_state, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_HOLD, AST_CONTROL_PROCEEDING, AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, AST_CONTROL_SRCUPDATE, AST_CONTROL_T38, AST_CONTROL_UNHOLD, AST_CONTROL_VIDUPDATE, ast_log(), ast_moh_start(), ast_moh_stop(), ast_rtp_new_source(), AST_SCHED_DEL_UNREF, ast_set_flag, AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), AST_STATE_RING, AST_STATE_UP, AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_test_flag, change_t38_state(), dialog_unref(), FALSE, sip_pvt::flags, sip_pvt::initreq, INV_COMPLETED, INV_EARLY_MEDIA, INV_PROCEEDING, sip_pvt::invitestate, LOG_ERROR, LOG_WARNING, sip_pvt::mohinterpret, sip_pvt::novideo, sip_pvt::rtp, sched, sip_alreadygone(), SIP_NEEDREINVITE, SIP_OUTGOING, SIP_PAGE2_T38SUPPORT, SIP_PENDINGBYE, SIP_PROG_INBAND, SIP_PROG_INBAND_NEVER, SIP_PROG_INBAND_YES, SIP_PROGRESS_SENT, sip_pvt_lock, sip_pvt_unlock, SIP_RINGING, t38properties::state, sip_pvt::t38, T38_DISABLED, T38_ENABLED, T38_LOCAL_REINVITE, T38_PEER_REINVITE, sip_pvt::t38id, t38properties::t38support, ast_channel::tech_pvt, transmit_info_with_vidupdate(), transmit_reinvite_with_sdp(), transmit_response(), transmit_response_reliable(), transmit_response_with_sdp(), transmit_response_with_t38_sdp(), TRUE, sip_pvt::vrtp, XMIT_CRITICAL, and XMIT_UNRELIABLE.

{
   struct sip_pvt *p = ast->tech_pvt;
   int res = 0;

   sip_pvt_lock(p);
   switch(condition) {
   case AST_CONTROL_RINGING:
      if (ast->_state == AST_STATE_RING) {
         p->invitestate = INV_EARLY_MEDIA;
         if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
             (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {            
            /* Send 180 ringing if out-of-band seems reasonable */
            transmit_response(p, "180 Ringing", &p->initreq);
            ast_set_flag(&p->flags[0], SIP_RINGING);
            if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES)
               break;
         } else {
            /* Well, if it's not reasonable, just send in-band */
         }
      }
      res = -1;
      break;
   case AST_CONTROL_BUSY:
      if (ast->_state != AST_STATE_UP) {
         transmit_response_reliable(p, "486 Busy Here", &p->initreq);
         p->invitestate = INV_COMPLETED;
         sip_alreadygone(p);
         ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
         break;
      }
      res = -1;
      break;
   case AST_CONTROL_CONGESTION:
      if (ast->_state != AST_STATE_UP) {
         transmit_response_reliable(p, "503 Service Unavailable", &p->initreq);
         p->invitestate = INV_COMPLETED;
         sip_alreadygone(p);
         ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
         break;
      }
      res = -1;
      break;
   case AST_CONTROL_PROCEEDING:
      if ((ast->_state != AST_STATE_UP) &&
          !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
          !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
         transmit_response(p, "100 Trying", &p->initreq);
         p->invitestate = INV_PROCEEDING;  
         break;
      }
      res = -1;
      break;
   case AST_CONTROL_PROGRESS:
      if ((ast->_state != AST_STATE_UP) &&
          !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
          !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
         p->invitestate = INV_EARLY_MEDIA;
         transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
         ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);  
         break;
      }
      res = -1;
      break;
   case AST_CONTROL_HOLD:
      ast_rtp_new_source(p->rtp);
      ast_moh_start(ast, data, p->mohinterpret);
      break;
   case AST_CONTROL_UNHOLD:
      ast_rtp_new_source(p->rtp);
      ast_moh_stop(ast);
      break;
   case AST_CONTROL_VIDUPDATE:   /* Request a video frame update */
      if (p->vrtp && !p->novideo) {
         transmit_info_with_vidupdate(p);
         /* ast_rtcp_send_h261fur(p->vrtp); */
      } else
         res = -1;
      break;
   case AST_CONTROL_T38:   /* T38 control frame */
      if (datalen != sizeof(enum ast_control_t38)) {
         ast_log(LOG_ERROR, "Invalid datalen for AST_CONTROL_T38. Expected %d, got %d\n", (int)sizeof(enum ast_control_t38), (int)datalen);
      } else {
         switch (*((enum ast_control_t38 *) data)) {
         case AST_T38_NEGOTIATED:
         case AST_T38_REQUEST_NEGOTIATE:     /* Request T38 */
            if (p->t38.state == T38_PEER_REINVITE) {
               AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
               change_t38_state(p, T38_ENABLED);
               transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
            } else if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT) && p->t38.state != T38_ENABLED) {
               change_t38_state(p, T38_LOCAL_REINVITE);
               if (!p->pendinginvite) {
                  transmit_reinvite_with_sdp(p, TRUE, FALSE);
               } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
                  ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
               }
            }
            break;
         case AST_T38_TERMINATED:
         case AST_T38_REFUSED:
         case AST_T38_REQUEST_TERMINATE:     /* Shutdown T38 */
            if (p->t38.state == T38_PEER_REINVITE) {
               AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
               change_t38_state(p, T38_DISABLED);
               transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
            } else if (p->t38.state == T38_ENABLED)
               transmit_reinvite_with_sdp(p, FALSE, FALSE);
            break;
         default:
            break;
         }
      }
      break;
   case AST_CONTROL_SRCUPDATE:
      ast_rtp_new_source(p->rtp);
      break;
   case -1:
      res = -1;
      break;
   default:
      ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
      res = -1;
      break;
   }
   sip_pvt_unlock(p);
   return res;
}

static const char * sip_nat_mode

(

const struct sip_pvt *

p

)

[static]

Display SIP nat mode.

Definition at line 2805 of file chan_sip.c.

References ast_test_flag, sip_pvt::flags, SIP_NAT, and SIP_NAT_ROUTE.

Referenced by check_via(), retrans_pkt(), and send_response().

{
   return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? "NAT" : "no NAT";
}

static struct ast_channel* sip_new	(	struct sip_pvt *	i,
		int	state,
		const char *	title
	)			[static]

Initiate a call in the SIP channel called from sip_request_call (calls from the pbx ) for outbound channels and from handle_request_invite for inbound channels.

Definition at line 5716 of file chan_sip.c.

References accountcode, sip_pvt::accountcode, ast_channel::adsicpe, ast_channel::amaflags, sip_pvt::amaflags, append_history, AST_ADSI_UNAVAILABLE, ast_best_codec(), AST_CAUSE_SWITCH_CONGESTION, ast_channel_alloc(), ast_channel_set_fd(), ast_codec_choose(), ast_copy_string(), ast_debug, ast_dsp_new(), ast_dsp_set_digitmode(), ast_dsp_set_features(), AST_FORMAT_TEXT_MASK, AST_FORMAT_VIDEO_MASK, ast_getformatname_multiple(), ast_hangup(), ast_jb_configure(), ast_log(), ast_module_ref(), ast_pbx_start(), ast_rtcp_fd(), ast_rtp_fd(), AST_STATE_DOWN, AST_STATE_RING, ast_strdup, ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_udptl_fd(), ast_uri_decode(), buf, sip_pvt::callgroup, ast_channel::callgroup, sip_pvt::callid, sip_pvt::callingpres, sip_pvt::capability, sip_pvt::chanvars, ast_channel::cid, ast_callerid::cid_ani, ast_callerid::cid_dnid, sip_pvt::cid_name, sip_pvt::cid_num, ast_callerid::cid_pres, ast_callerid::cid_rdnis, ast_channel::context, sip_pvt::context, dialog_ref(), sip_pvt::do_history, sip_pvt::domain, DSP_DIGITMODE_DTMF, DSP_DIGITMODE_RELAXDTMF, DSP_FEATURE_DIGIT_DETECT, EVENT_FLAG_SYSTEM, ast_channel::exten, sip_pvt::exten, sip_pvt::flags, sip_pvt::fromdomain, sip_pvt::fullcontact, global_jbconf, ast_channel::hangupcause, sip_pvt::jointcapability, language, sip_pvt::language, LOG_WARNING, manager_event, ast_channel::name, ast_variable::name, ast_channel::nativeformats, ast_variable::next, sip_pvt::owner, pbx_builtin_setvar_helper(), sip_pvt::pickupgroup, ast_channel::pickupgroup, sip_pvt::prefcodec, sip_pvt::prefs, ast_channel::priority, ast_channel::rawreadformat, ast_channel::rawwriteformat, sip_pvt::rdnis, ast_channel::readformat, ast_channel::rings, sip_pvt::rtp, SIP_DTMF, SIP_DTMF_AUTO, SIP_DTMF_INBAND, SIP_DTMF_INFO, SIP_DTMF_SHORTINFO, SIP_PAGE2_VIDEOSUPPORT, sip_pvt_lock, sip_pvt_unlock, sip_tech, sip_tech_info, SIPBUFSIZE, ast_channel::tech, ast_channel::tech_pvt, text, sip_pvt::trtp, sip_pvt::udptl, ast_channel::uniqueid, sip_pvt::uri, sip_pvt::vad, ast_variable::value, sip_pvt::vrtp, and ast_channel::writeformat.

Referenced by handle_request_invite(), and sip_request_call().

{
   struct ast_channel *tmp;
   struct ast_variable *v = NULL;
   int fmt;
   int what;
   int video;
   int text;
   int needvideo = 0;
   int needtext = 0;
   char buf[SIPBUFSIZE];
   char *decoded_exten;

   {
      const char *my_name; /* pick a good name */
   
      if (title)
         my_name = title;
      else if ( (my_name = strchr(i->fromdomain, ':')) )
         my_name++;  /* skip ':' */
      else
         my_name = i->fromdomain;

      sip_pvt_unlock(i);
      /* Don't hold a sip pvt lock while we allocate a channel */
      tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, i->amaflags, "SIP/%s-%08x", my_name, (int)(long) i);

   }
   if (!tmp) {
      ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
      sip_pvt_lock(i);
      return NULL;
   }
   sip_pvt_lock(i);

   tmp->tech = ( ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INFO || ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO) ?  &sip_tech_info : &sip_tech;

   /* Select our native format based on codec preference until we receive
      something from another device to the contrary. */
   if (i->jointcapability) {  /* The joint capabilities of us and peer */
      what = i->jointcapability;
      video = i->jointcapability & AST_FORMAT_VIDEO_MASK;
      text = i->jointcapability & AST_FORMAT_TEXT_MASK;
   } else if (i->capability) {      /* Our configured capability for this peer */
      what = i->capability;
      video = i->capability & AST_FORMAT_VIDEO_MASK;
      text = i->capability & AST_FORMAT_TEXT_MASK;
   } else {
      what = global_capability;  /* Global codec support */
      video = global_capability & AST_FORMAT_VIDEO_MASK;
      text = global_capability & AST_FORMAT_TEXT_MASK;
   }

   /* Set the native formats for audio  and merge in video */
   tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video | text;
   ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats));
   ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability));
   ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability));
   ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1)));
   if (i->prefcodec)
      ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcodec));

   /* XXX Why are we choosing a codec from the native formats?? */
   fmt = ast_best_codec(tmp->nativeformats);

   /* If we have a prefcodec setting, we have an inbound channel that set a 
      preferred format for this call. Otherwise, we check the jointcapability
      We also check for vrtp. If it's not there, we are not allowed do any video anyway.
    */
   if (i->vrtp) {
      if (ast_test_flag(&i->flags[1], SIP_PAGE2_VIDEOSUPPORT))
         needvideo = AST_FORMAT_VIDEO_MASK;
      else if (i->prefcodec)
         needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK;  /* Outbound call */
      else
         needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK;  /* Inbound call */
   }

   if (i->trtp) {
      if (i->prefcodec)
         needtext = i->prefcodec & AST_FORMAT_TEXT_MASK; /* Outbound call */
      else
         needtext = i->jointcapability & AST_FORMAT_TEXT_MASK; /* Inbound call */
   }

   if (needvideo) 
      ast_debug(3, "This channel can handle video! HOLLYWOOD next!\n");
   else
      ast_debug(3, "This channel will not be able to handle video.\n");

   if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
      i->vad = ast_dsp_new();
      ast_dsp_set_features(i->vad, DSP_FEATURE_DIGIT_DETECT);
      if (global_relaxdtmf)
         ast_dsp_set_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
   }

   /* Set file descriptors for audio, video, realtime text and UDPTL as needed */
   if (i->rtp) {
      ast_channel_set_fd(tmp, 0, ast_rtp_fd(i->rtp));
      ast_channel_set_fd(tmp, 1, ast_rtcp_fd(i->rtp));
   }
   if (needvideo && i->vrtp) {
      ast_channel_set_fd(tmp, 2, ast_rtp_fd(i->vrtp));
      ast_channel_set_fd(tmp, 3, ast_rtcp_fd(i->vrtp));
   }
   if (needtext && i->trtp) 
      ast_channel_set_fd(tmp, 4, ast_rtp_fd(i->trtp));
   if (i->udptl)
      ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));

   if (state == AST_STATE_RING)
      tmp->rings = 1;
   tmp->adsicpe = AST_ADSI_UNAVAILABLE;
   tmp->writeformat = fmt;
   tmp->rawwriteformat = fmt;
   tmp->readformat = fmt;
   tmp->rawreadformat = fmt;
   tmp->tech_pvt = dialog_ref(i, "sip_new: set chan->tech_pvt to i");

   tmp->callgroup = i->callgroup;
   tmp->pickupgroup = i->pickupgroup;
   tmp->cid.cid_pres = i->callingpres;
   if (!ast_strlen_zero(i->accountcode))
      ast_string_field_set(tmp, accountcode, i->accountcode);
   if (i->amaflags)
      tmp->amaflags = i->amaflags;
   if (!ast_strlen_zero(i->language))
      ast_string_field_set(tmp, language, i->language);
   i->owner = tmp;
   ast_module_ref(ast_module_info->self);
   ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
   /*Since it is valid to have extensions in the dialplan that have unescaped characters in them
    * we should decode the uri before storing it in the channel, but leave it encoded in the sip_pvt
    * structure so that there aren't issues when forming URI's
    */
   decoded_exten = ast_strdupa(i->exten);
   ast_uri_decode(decoded_exten);
   ast_copy_string(tmp->exten, decoded_exten, sizeof(tmp->exten));

   /* Don't use ast_set_callerid() here because it will
    * generate an unnecessary NewCallerID event  */
   tmp->cid.cid_ani = ast_strdup(i->cid_num);
   if (!ast_strlen_zero(i->rdnis))
      tmp->cid.cid_rdnis = ast_strdup(i->rdnis);
   
   if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
      tmp->cid.cid_dnid = ast_strdup(i->exten);

   tmp->priority = 1;
   if (!ast_strlen_zero(i->uri))
      pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri);
   if (!ast_strlen_zero(i->domain))
      pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
   if (!ast_strlen_zero(i->callid))
      pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
   if (i->rtp)
      ast_jb_configure(tmp, &global_jbconf);

   /* Set channel variables for this call from configuration */
   for (v = i->chanvars ; v ; v = v->next)
      pbx_builtin_setvar_helper(tmp, v->name, v->value);

   if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
      ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
      tmp->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
      ast_hangup(tmp);
      tmp = NULL;
   }

   if (i->do_history)
      append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);

   /* Inform manager user about new channel and their SIP call ID */
   if (global_callevents)
      manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
         "Channel: %s\r\nUniqueid: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\n",
         tmp->name, tmp->uniqueid, "SIP", i->callid, i->fullcontact);

   return tmp;
}

static int sip_park	(	struct ast_channel *	chan1,
		struct ast_channel *	chan2,
		struct sip_request *	req,
		int	seqno
	)			[static]

Park a call using the subsystem in res_features.c This is executed in a separate thread.

Definition at line 16978 of file chan_sip.c.

References ast_channel::accountcode, ast_channel::amaflags, ast_calloc, AST_CAUSE_SWITCH_CONGESTION, ast_channel_alloc(), ast_channel_lock, ast_channel_masquerade(), ast_channel_trylock, ast_channel_unlock, ast_copy_string(), ast_debug, ast_do_masquerade(), ast_free, ast_hangup(), ast_log(), ast_pthread_create_detached_background, AST_STATE_DOWN, ast_channel::context, copy_request(), ast_channel::exten, ast_channel::hangupcause, LOG_WARNING, ast_channel::name, ast_channel::priority, ast_channel::readformat, sip_park_thread(), sip_pvt_lock, sip_pvt_unlock, ast_channel::tech_pvt, and ast_channel::writeformat.

Referenced by handle_request_refer().

{
   struct sip_dual *d;
   struct ast_channel *transferee, *transferer;
      /* Chan2m: The transferer, chan1m: The transferee */
   pthread_t th;

   transferee = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, chan1->accountcode, chan1->exten, chan1->context, chan1->amaflags, "Parking/%s", chan1->name);
   transferer = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, chan2->accountcode, chan2->exten, chan2->context, chan2->amaflags, "SIPPeer/%s", chan2->name);
   if ((!transferer) || (!transferee)) {
      if (transferee) {
         transferee->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
         ast_hangup(transferee);
      }
      if (transferer) {
         transferer->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
         ast_hangup(transferer);
      }
      return -1;
   }

   /* Make formats okay */
   transferee->readformat = chan1->readformat;
   transferee->writeformat = chan1->writeformat;

   /* Prepare for taking over the channel */
   ast_channel_masquerade(transferee, chan1);

   /* Setup the extensions and such */
   ast_copy_string(transferee->context, chan1->context, sizeof(transferee->context));
   ast_copy_string(transferee->exten, chan1->exten, sizeof(transferee->exten));
   transferee->priority = chan1->priority;
      
   /* We make a clone of the peer channel too, so we can play
      back the announcement */

   /* Make formats okay */
   transferer->readformat = chan2->readformat;
   transferer->writeformat = chan2->writeformat;

   /* Prepare for taking over the channel.  Go ahead and grab this channel
    * lock here to avoid a deadlock with callbacks into the channel driver
    * that hold the channel lock and want the pvt lock.  */
   while (ast_channel_trylock(chan2)) {
      struct sip_pvt *pvt = chan2->tech_pvt;
      sip_pvt_unlock(pvt);
      usleep(1);
      sip_pvt_lock(pvt);
   }
   ast_channel_masquerade(transferer, chan2);
   ast_channel_unlock(chan2);

   /* Setup the extensions and such */
   ast_copy_string(transferer->context, chan2->context, sizeof(transferer->context));
   ast_copy_string(transferer->exten, chan2->exten, sizeof(transferer->exten));
   transferer->priority = chan2->priority;

   ast_channel_lock(transferer);
   if (ast_do_masquerade(transferer)) {
      ast_log(LOG_WARNING, "Masquerade failed :(\n");
      ast_channel_unlock(transferer);
      transferer->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
      ast_hangup(transferer);
      return -1;
   }
   ast_channel_unlock(transferer);
   if (!transferer || !transferee) {
      if (!transferer) { 
         ast_debug(1, "No transferer channel, giving up parking\n");
      }
      if (!transferee) {
         ast_debug(1, "No transferee channel, giving up parking\n");
      }
      return -1;
   }
   if ((d = ast_calloc(1, sizeof(*d)))) {

      /* Save original request for followup */
      copy_request(&d->req, req);
      d->chan1 = transferee;  /* Transferee */
      d->chan2 = transferer;  /* Transferer */
      d->seqno = seqno;
      if (ast_pthread_create_detached_background(&th, NULL, sip_park_thread, d) < 0) {
         /* Could not start thread */
         if (d->req.data)
            ast_free(d->req.data);
         ast_free(d);   /* We don't need it anymore. If thread is created, d will be free'd
                  by sip_park_thread() */
         return 0;
      }
   } 
   return -1;
}

static void * sip_park_thread

(

void *

stuff

)

[static]

Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup.

Definition at line 16906 of file chan_sip.c.

References append_history, AST_CAUSE_NORMAL_CLEARING, ast_channel_lock, ast_channel_unlock, ast_debug, ast_do_masquerade(), ast_free, ast_hangup(), ast_log(), ast_park_call(), buf, sip_dual::chan1, sip_dual::chan2, copy_request(), sip_request::data, ext, free, ast_channel::hangupcause, LOG_ERROR, LOG_WARNING, ast_channel::name, sip_dual::req, sip_dual::seqno, ast_channel::tech_pvt, transmit_message_with_text(), transmit_notify_with_sipfrag(), transmit_response(), and TRUE.

Referenced by sip_park().

{
   struct ast_channel *transferee, *transferer; /* Chan1: The transferee, Chan2: The transferer */
   struct sip_dual *d;
   struct sip_request req = {0,};
   int ext;
   int res;

   d = stuff;
   transferee = d->chan1;
   transferer = d->chan2;
   copy_request(&req, &d->req);

   if (!transferee || !transferer) {
      ast_log(LOG_ERROR, "Missing channels for parking! Transferer %s Transferee %s\n", transferer ? "<available>" : "<missing>", transferee ? "<available>" : "<missing>" );
      if (d->req.data)
         ast_free(d->req.data);
      free(d);
      return NULL;
   }
   ast_debug(4, "SIP Park: Transferer channel %s, Transferee %s\n", transferer->name, transferee->name);

   ast_channel_lock(transferee);
   if (ast_do_masquerade(transferee)) {
      ast_log(LOG_WARNING, "Masquerade failed.\n");
      transmit_response(transferer->tech_pvt, "503 Internal error", &req);
      ast_channel_unlock(transferee);
      if (d->req.data)
         ast_free(d->req.data);
      free(d);
      return NULL;
   } 
   ast_channel_unlock(transferee);

   res = ast_park_call(transferee, transferer, 0, &ext);
   

#ifdef WHEN_WE_KNOW_THAT_THE_CLIENT_SUPPORTS_MESSAGE
   if (!res) {
      transmit_message_with_text(transferer->tech_pvt, "Unable to park call.\n");
   } else {
      /* Then tell the transferer what happened */
      sprintf(buf, "Call parked on extension '%d'", ext);
      transmit_message_with_text(transferer->tech_pvt, buf);
   }
#endif

   /* Any way back to the current call??? */
   /* Transmit response to the REFER request */
   transmit_response(transferer->tech_pvt, "202 Accepted", &req);
   if (!res)   {
      /* Transfer succeeded */
      append_history(transferer->tech_pvt, "SIPpark", "Parked call on %d", ext);
      transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "200 OK", TRUE);
      transferer->hangupcause = AST_CAUSE_NORMAL_CLEARING;
      ast_hangup(transferer); /* This will cause a BYE */
      ast_debug(1, "SIP Call parked on extension '%d'\n", ext);
   } else {
      transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "503 Service Unavailable", TRUE);
      append_history(transferer->tech_pvt, "SIPpark", "Parking failed\n");
      ast_debug(1, "SIP Call parked failed \n");
      /* Do not hangup call */
   }
   if (d->req.data)
      ast_free(d->req.data);
   free(d);
   return NULL;
}

static int sip_parse_host	(	char *	line,
		int	lineno,
		char **	hostname,
		int *	portnum,
		enum sip_transport *	transport
	)			[static]

Small function to parse a config line for a host with a transport i.e. tls://www.google.com:8056.

Definition at line 20140 of file chan_sip.c.

References ast_log(), LOG_NOTICE, SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, SIP_TRANSPORT_UDP, STANDARD_SIP_PORT, and STANDARD_TLS_PORT.

Referenced by sip_register().

{
   char *port;

   if ((*hostname = strstr(line, "://"))) {
      *hostname += 3;

      if (!strncasecmp(line, "tcp", 3))
         *transport = SIP_TRANSPORT_TCP;
      else if (!strncasecmp(line, "tls", 3))
         *transport = SIP_TRANSPORT_TLS;
      else if (!strncasecmp(line, "udp", 3))
         *transport = SIP_TRANSPORT_UDP;
      else
         ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type on line %d of sip.conf. defaulting to udp.\n", line, lineno);
   } else {
      *hostname = line;
      *transport = SIP_TRANSPORT_UDP;
   }

   if ((line = strrchr(*hostname, '@')))
      line++;
   else
      line = *hostname;

   if ((port = strrchr(line, ':'))) {
      *port++ = '\0';

      if (!sscanf(port, "%u", portnum)) {
         ast_log(LOG_NOTICE, "'%s' is not a valid port number on line %d of sip.conf. using default.\n", port, lineno);
         port = NULL;
      }
   }

   if (!port) {
      if (*transport & SIP_TRANSPORT_TLS) {
         *portnum = STANDARD_TLS_PORT;
      } else {
         *portnum = STANDARD_SIP_PORT;
      }
   }

   return 0;
}

static void sip_peer_hold	(	struct sip_pvt *	p,
		int	hold
	)			[static]

Change onhold state of a peer using a pvt structure.

Definition at line 11065 of file chan_sip.c.

References ast_atomic_fetchadd_int(), AST_DEVICE_UNKNOWN, ast_devstate_changed(), FALSE, find_peer(), FINDALLDEVICES, sip_peer::name, sip_peer::onHold, sip_pvt::peername, and unref_peer().

Referenced by process_sdp(), and update_call_counter().

{
   struct sip_peer *peer = find_peer(p->peername, NULL, 1, FINDALLDEVICES, FALSE);

   if (!peer)
      return;

   /* If they put someone on hold, increment the value... otherwise decrement it */
   ast_atomic_fetchadd_int(&peer->onHold, (hold ? +1 : -1));

   /* Request device state update */
   ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
   unref_peer(peer, "sip_peer_hold: from find_peer operation");
   
   return;
}

static void sip_poke_all_peers

(

void

)

[static]

Send a poke to all known peers Space them out 100 ms apart XXX We might have a cool algorithm for this or use random - any suggestions?

Definition at line 23247 of file chan_sip.c.

References ao2_iterator_init(), ao2_lock(), ao2_t_iterator_next, ao2_unlock(), AST_SCHED_REPLACE_UNREF, sip_peer::pokeexpire, ref_peer(), sched, sip_poke_peer_s(), and unref_peer().

Referenced by load_module(), and sip_do_reload().

{
   int ms = 0;
   struct ao2_iterator i;
   struct sip_peer *peer;

   i = ao2_iterator_init(peers, 0);
   
   if (!speerobjs)   /* No peers, just give up */
      return;

   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
      ao2_lock(peer);
      ms += 100;
      AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, ms, sip_poke_peer_s, peer,
            unref_peer(_data, "removing poke peer ref"),
            unref_peer(peer, "removing poke peer ref"),
            ref_peer(peer, "adding poke peer ref"));
      ao2_unlock(peer);
      unref_peer(peer, "toss iterator peer ptr");
   }
}

static int sip_poke_noanswer

(

const void *

data

)

[static]

React to lack of answer to Qualify poke.

Definition at line 20754 of file chan_sip.c.

References ast_check_realtime(), AST_DEVICE_UNKNOWN, ast_devstate_changed(), ast_log(), AST_SCHED_REPLACE_UNREF, ast_update_realtime(), sip_peer::call, DEFAULT_FREQ_NOTOK, dialog_unlink_all(), dialog_unref(), EVENT_FLAG_SYSTEM, FALSE, sip_peer::lastms, LOG_NOTICE, manager_event, sip_peer::name, sip_settings::peer_rtupdate, sip_peer::pokeexpire, ref_peer(), register_peer_exten(), sched, SENTINEL, sip_cfg, sip_poke_peer_s(), TRUE, and unref_peer().

Referenced by sip_poke_peer(), and sip_show_sched().

{
   struct sip_peer *peer = (struct sip_peer *)data;
   
   peer->pokeexpire = -1;

   if (peer->lastms > -1) {
      ast_log(LOG_NOTICE, "Peer '%s' is now UNREACHABLE!  Last qualify: %d\n", peer->name, peer->lastms);
      if (sip_cfg.peer_rtupdate) {
         ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", "-1", SENTINEL);
      }
      manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unreachable\r\nTime: %d\r\n", peer->name, -1);
      if (global_regextenonqualify) {
         register_peer_exten(peer, FALSE);
      }
   }

   if (peer->call) {
      dialog_unlink_all(peer->call, TRUE, TRUE);
      peer->call = dialog_unref(peer->call, "unref dialog peer->call");
      /* peer->call = sip_destroy(peer->call);*/
   }
   
   peer->lastms = -1;
   ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);

   /* Try again quickly */
   AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, 
         DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer,
         unref_peer(_data, "removing poke peer ref"),
         unref_peer(peer, "removing poke peer ref"),
         ref_peer(peer, "adding poke peer ref"));

   /* Release the ref held by the running scheduler entry */
   unref_peer(peer, "release peer poke noanswer ref");

   return 0;
}

static int sip_poke_peer	(	struct sip_peer *	peer,
		int	force
	)			[static]

Check availability of peer, also keep NAT open.

Note:

This is done with 60 seconds between each ping, unless forced by cli or manager. If peer is unreachable, we check every 10th second by default.

Definition at line 20798 of file chan_sip.c.

References sip_peer::addr, ao2_t_link, ao2_t_unlink, ast_copy_flags, ast_copy_string(), ast_inet_ntoa(), ast_log(), AST_SCHED_DEL_UNREF, AST_SCHED_REPLACE_UNREF, ast_set_flag, ast_sip_ouraddrfor(), ast_string_field_set, ast_strlen_zero(), ast_tvnow(), build_callid_pvt(), build_via(), sip_peer::call, copy_socket_data(), dialog_ref(), dialog_unlink_all(), dialog_unref(), dialogs, sip_peer::flags, sip_pvt::flags, sip_pvt::fullcontact, sip_peer::fullcontact, sip_peer::lastms, LOG_NOTICE, sip_peer::maxms, sip_pvt::ourip, sip_peer::pokeexpire, sip_peer::ps, sip_pvt::recv, ref_peer(), sip_pvt::relatedpeer, sip_pvt::sa, sched, sip_alloc(), SIP_FLAGS_TO_COPY, SIP_INVITE, SIP_OPTIONS, SIP_OUTGOING, SIP_PAGE2_FLAGS_TO_COPY, sip_poke_noanswer(), sip_peer::socket, sip_pvt::socket, sip_pvt::tohost, sip_peer::tohost, transmit_invite(), TRUE, unref_peer(), sip_pvt::username, and XMIT_ERROR.

Referenced by _sip_qualify_peer(), build_peer(), parse_register_contact(), reg_source_db(), and sip_poke_peer_s().

{
   struct sip_pvt *p;
   int xmitres = 0;
   
   if ((!peer->maxms && !force) || !peer->addr.sin_addr.s_addr) {
      /* IF we have no IP, or this isn't to be monitored, return
        immediately after clearing things out */
      AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
            unref_peer(peer, "removing poke peer ref"));
      
      peer->lastms = 0;
      if (peer->call) {
         peer->call = dialog_unref(peer->call, "unref dialog peer->call");
      }
      return 0;
   }
   if (peer->call) {
      if (sipdebug) {
         ast_log(LOG_NOTICE, "Still have a QUALIFY dialog active, deleting\n");
      }
      dialog_unlink_all(peer->call, TRUE, TRUE);
      peer->call = dialog_unref(peer->call, "unref dialog peer->call");
      /* peer->call = sip_destroy(peer->call); */
   }
   if (!(p = sip_alloc(NULL, NULL, 0, SIP_OPTIONS))) {
      return -1;
   }
   peer->call = dialog_ref(p, "copy sip alloc from p to peer->call");
   
   p->sa = peer->addr;
   p->recv = peer->addr;
   copy_socket_data(&p->socket, &peer->socket);
   ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);

   /* Send OPTIONs to peer's fullcontact */
   if (!ast_strlen_zero(peer->fullcontact))
      ast_string_field_set(p, fullcontact, peer->fullcontact);

   if (!ast_strlen_zero(peer->tohost))
      ast_string_field_set(p, tohost, peer->tohost);
   else
      ast_string_field_set(p, tohost, ast_inet_ntoa(peer->addr.sin_addr));

   /* Recalculate our side, and recalculate Call ID */
   ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
   build_via(p);
   ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
   build_callid_pvt(p);
   ao2_t_link(dialogs, p, "Linking in under new name");

   AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
         unref_peer(peer, "removing poke peer ref"));
   
   if (p->relatedpeer)
      p->relatedpeer = unref_peer(p->relatedpeer,"unsetting the relatedpeer field in the dialog, before it is set to something else.");
   p->relatedpeer = ref_peer(peer, "setting the relatedpeer field in the dialog");
   ast_set_flag(&p->flags[0], SIP_OUTGOING);
#ifdef VOCAL_DATA_HACK
   ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username));
   xmitres = transmit_invite(p, SIP_INVITE, 0, 2); /* sinks the p refcount */
#else
   xmitres = transmit_invite(p, SIP_OPTIONS, 0, 2); /* sinks the p refcount */
#endif
   peer->ps = ast_tvnow();
   if (xmitres == XMIT_ERROR) {
      sip_poke_noanswer(peer);   /* Immediately unreachable, network problems */
   } else if (!force) {
      AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, peer->maxms * 2, sip_poke_noanswer, peer,
            unref_peer(_data, "removing poke peer ref"),
            unref_peer(peer, "removing poke peer ref"),
            ref_peer(peer, "adding poke peer ref"));
   }
   dialog_unref(p, "unref dialog at end of sip_poke_peer, obtained from sip_alloc, just before it goes out of scope");
   return 0;
}

static int sip_poke_peer_s

(

const void *

data

)

[static]

Poke peer (send qualify to check if peer is alive and well).

Definition at line 10428 of file chan_sip.c.

References sip_peer::pokeexpire, sip_poke_peer(), and unref_peer().

Referenced by handle_response_peerpoke(), reg_source_db(), sip_poke_all_peers(), sip_poke_noanswer(), and sip_show_sched().

{
   struct sip_peer *peer = (struct sip_peer *)data;

   peer->pokeexpire = -1;

   sip_poke_peer(peer, 0);

   unref_peer(peer, "removing poke peer ref");

   return 0;
}

static int sip_prepare_socket

(

struct sip_pvt *

p

)

[static]

Todo:

document this function.

Definition at line 20062 of file chan_sip.c.

References ast_tcptls_session_args::accept_fd, ao2_ref, ast_calloc, ast_copy_string(), ast_debug, ast_free, ast_pthread_create_background, ast_strlen_zero(), ast_tcptls_client_start(), default_tls_cfg, ast_tcptls_session_instance::fd, ast_tcptls_session_args::hostname, ast_tcptls_session_args::master, ast_tcptls_session_args::name, name, sip_pvt::outboundproxy, ast_tcptls_session_args::remote_address, s, sip_real_dst(), sip_tcp_locate(), sip_tcp_worker_fn(), SIP_TRANSPORT_TLS, SIP_TRANSPORT_UDP, sipsock, sip_pvt::socket, ast_tcptls_session_args::tls_cfg, sip_pvt::tohost, and sip_proxy::transport.

Referenced by __sip_xmit().

{
   struct sip_socket *s = &p->socket;
   static const char name[] = "SIP socket";
   struct ast_tcptls_session_instance *tcptls_session;
   struct ast_tcptls_session_args ca = {
      .name = name,
      .accept_fd = -1,
   };

   if (s->fd != -1)
      return s->fd;

   if (p->outboundproxy && p->outboundproxy->transport) {
      s->type = p->outboundproxy->transport;
   }

   if (s->type & SIP_TRANSPORT_UDP) {
      s->fd = sipsock;
      return s->fd;
   }

   ca.remote_address = *(sip_real_dst(p));

   if ((tcptls_session = sip_tcp_locate(&ca.remote_address))) {   /* Check if we have a thread handling a socket connected to this IP/port */
      s->fd = tcptls_session->fd;
      if (s->tcptls_session) {
         ao2_ref(s->tcptls_session, -1);
         s->tcptls_session = NULL;
      }
      s->tcptls_session = tcptls_session;
      return s->fd;
   }

   if (s->tcptls_session && s->tcptls_session->parent->tls_cfg) {
      ca.tls_cfg = s->tcptls_session->parent->tls_cfg;
   } else {
      if (s->type & SIP_TRANSPORT_TLS) {
         ca.tls_cfg = ast_calloc(1, sizeof(*ca.tls_cfg));
         if (!ca.tls_cfg)
            return -1;
         memcpy(ca.tls_cfg, &default_tls_cfg, sizeof(*ca.tls_cfg));
         if (!ast_strlen_zero(p->tohost))
            ast_copy_string(ca.hostname, p->tohost, sizeof(ca.hostname));
      }
   }
   
   if (s->tcptls_session) {
      /* the pvt socket already has a server instance ... */
   } else {
      s->tcptls_session = ast_tcptls_client_start(&ca); /* Start a client connection to this address */
   }

   if (!s->tcptls_session) {
      if (ca.tls_cfg)
         ast_free(ca.tls_cfg);
      return -1;
   }

   s->fd = ca.accept_fd;

   /* Give the new thread a reference */
   ao2_ref(s->tcptls_session, +1);

   if (ast_pthread_create_background(&ca.master, NULL, sip_tcp_worker_fn, s->tcptls_session)) {
      ast_debug(1, "Unable to launch '%s'.", ca.name);
      ao2_ref(s->tcptls_session, -1);
      close(ca.accept_fd);
      s->fd = ca.accept_fd = -1;
   }

   return s->fd;
}

static char * sip_prune_realtime	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Remove temporary realtime objects from memory (CLI).

Todo:

XXXX Propably needs an overhaul after removal of the devices

Definition at line 13312 of file chan_sip.c.

References sip_peer::addr, ao2_iterator_init(), ao2_lock(), ao2_t_callback, ao2_t_find, ao2_t_iterator_next, ao2_t_link, ao2_t_unlink, ao2_unlock(), ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_copy_string(), ast_test_flag, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_peer(), FALSE, ast_cli_args::fd, sip_peer::flags, ast_cli_args::line, ast_cli_args::n, sip_peer::name, name, OBJ_MULTIPLE, OBJ_NODATA, OBJ_POINTER, OBJ_UNLINK, peer_is_marked(), peers_by_ip, ast_cli_args::pos, SIP_PAGE2_RTCACHEFRIENDS, strcasestr(), sip_peer::the_mark, TRUE, unref_peer(), ast_cli_entry::usage, and ast_cli_args::word.

{
   struct sip_peer *peer, *pi;
   int prunepeer = FALSE;
   int multi = FALSE;
   char *name = NULL;
   regex_t regexbuf;
   struct ao2_iterator i;
   
   if (cmd == CLI_INIT) {
      e->command = "sip prune realtime [peer|all] [all|like]";
      e->usage =
         "Usage: sip prune realtime [peer [<name>|all|like <pattern>]|all]\n"
         "       Prunes object(s) from the cache.\n"
         "       Optional regular expression pattern is used to filter the objects.\n";
      return NULL;
   } else if (cmd == CLI_GENERATE) {
      if (a->pos == 4) {
         if (strcasestr(a->line, "realtime peer"))
            return complete_sip_peer(a->word, a->n, SIP_PAGE2_RTCACHEFRIENDS);
      }
      return NULL;
   }
   switch (a->argc) {
   case 4:
      name = a->argv[3];
      /* we accept a name in position 3, but keywords are not good. */
      if (!strcasecmp(name, "peer") || !strcasecmp(name, "like"))
         return CLI_SHOWUSAGE;
      prunepeer = TRUE;
      if (!strcasecmp(name, "all")) {
         multi = TRUE;
         name = NULL;
      }
      /* else a single name, already set */
      break;
   case 5:
      /* sip prune realtime {peer|like} name */
      name = a->argv[4];
      if (!strcasecmp(a->argv[3], "peer"))
         prunepeer = TRUE;
      else if (!strcasecmp(a->argv[3], "like")) {
         prunepeer = TRUE;
         multi = TRUE;
      } else
         return CLI_SHOWUSAGE;
      if (!strcasecmp(a->argv[4], "like"))
         return CLI_SHOWUSAGE;
      if (!multi && !strcasecmp(a->argv[4], "all")) {
         multi = TRUE;
         name = NULL;
      }
      break;
   case 6:
      name = a->argv[5];
      multi = TRUE;
      /* sip prune realtime {peer} like name */
      if (strcasecmp(a->argv[4], "like"))
         return CLI_SHOWUSAGE;
      if (!strcasecmp(a->argv[3], "peer")) {
         prunepeer = TRUE;
      } else
         return CLI_SHOWUSAGE;
      break;
   default:
      return CLI_SHOWUSAGE;
   }

   if (multi && name) {
      if (regcomp(&regexbuf, name, REG_EXTENDED | REG_NOSUB))
         return CLI_SHOWUSAGE;
   }

   if (multi) {
      if (prunepeer) {
         int pruned = 0;
         
         i = ao2_iterator_init(peers, 0);
         while ((pi = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
            ao2_lock(pi);
            if (name && regexec(&regexbuf, pi->name, 0, NULL, 0)) {
               unref_peer(pi, "toss iterator peer ptr before continue");
               ao2_unlock(pi);
               continue;
            };
            if (ast_test_flag(&pi->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
               pi->the_mark = 1;
               pruned++;
            }
            ao2_unlock(pi);
            unref_peer(pi, "toss iterator peer ptr");
         }
         if (pruned) {
            ao2_t_callback(peers, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE, peer_is_marked, 0, 
                  "initiating callback to remove marked peers");
            ast_cli(a->fd, "%d peers pruned.\n", pruned);
         } else
            ast_cli(a->fd, "No peers found to prune.\n");
      }
   } else {
      if (prunepeer) {
         struct sip_peer tmp;
         ast_copy_string(tmp.name, name, sizeof(tmp.name));
         if ((peer = ao2_t_find(peers, &tmp, OBJ_POINTER | OBJ_UNLINK, "finding to unlink from peers"))) {
            if (peer->addr.sin_addr.s_addr) {
               ao2_t_unlink(peers_by_ip, peer, "unlinking peer from peers_by_ip also");
            }
            if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
               ast_cli(a->fd, "Peer '%s' is not a Realtime peer, cannot be pruned.\n", name);
               /* put it back! */
               ao2_t_link(peers, peer, "link peer into peer table");
               if (peer->addr.sin_addr.s_addr) {
                  ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
               }
               
            } else
               ast_cli(a->fd, "Peer '%s' pruned.\n", name);
            unref_peer(peer, "sip_prune_realtime: unref_peer: tossing temp peer ptr");
         } else
            ast_cli(a->fd, "Peer '%s' not found.\n", name);
      }
   }

   return CLI_SUCCESS;
}

static char * sip_qualify_peer	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Send an OPTIONS packet to a SIP peer.

Definition at line 13590 of file chan_sip.c.

References _sip_qualify_peer(), ast_cli_args::argc, ast_cli_args::argv, CLI_GENERATE, CLI_INIT, ast_cli_entry::command, complete_sip_show_peer(), ast_cli_args::fd, ast_cli_args::line, ast_cli_args::n, ast_cli_args::pos, ast_cli_entry::usage, and ast_cli_args::word.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip qualify peer";
      e->usage =
         "Usage: sip qualify peer <name> [load]\n"
         "       Requests a response from one SIP peer and the current status.\n"
         "       Option \"load\" forces lookup of peer in realtime storage.\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_show_peer(a->line, a->word, a->pos, a->n);
   }
   return _sip_qualify_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
}

static int sip_queryoption	(	struct ast_channel *	chan,
		int	option,
		void *	data,
		int *	datalen
	)			[static]

Query an option on a SIP dialog.

Definition at line 3496 of file chan_sip.c.

References ast_log(), AST_OPTION_T38_STATE, ast_test_flag, chan, LOG_ERROR, SIP_PAGE2_T38SUPPORT, sip_pvt_lock, sip_pvt_unlock, t38properties::state, sip_pvt::t38, T38_ENABLED, T38_LOCAL_REINVITE, T38_PEER_DIRECT, T38_PEER_REINVITE, T38_STATE_NEGOTIATED, T38_STATE_NEGOTIATING, T38_STATE_UNAVAILABLE, T38_STATE_UNKNOWN, t38properties::t38support, and ast_channel::tech_pvt.

{
   int res = -1;
   enum ast_t38_state state = T38_STATE_UNAVAILABLE;
   struct sip_pvt *p = (struct sip_pvt *) chan->tech_pvt;

   switch (option) {
   case AST_OPTION_T38_STATE:
      /* Make sure we got an ast_t38_state enum passed in */
      if (*datalen != sizeof(enum ast_t38_state)) {
         ast_log(LOG_ERROR, "Invalid datalen for AST_OPTION_T38_STATE option. Expected %d, got %d\n", (int)sizeof(enum ast_t38_state), *datalen);
         return -1;
      }

      sip_pvt_lock(p);

      /* Now if T38 support is enabled we need to look and see what the current state is to get what we want to report back */
      if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT)) {
         switch (p->t38.state) {
         case T38_LOCAL_REINVITE:
         case T38_PEER_DIRECT:
         case T38_PEER_REINVITE:
            state = T38_STATE_NEGOTIATING;
            break;
         case T38_ENABLED:
            state = T38_STATE_NEGOTIATED;
            break;
         default:
            state = T38_STATE_UNKNOWN;
         }
      }

      sip_pvt_unlock(p);

      *((enum ast_t38_state *) data) = state;
      res = 0;

      break;
   default:
      break;
   }

   return res;
}

static struct ast_frame * sip_read

(

struct ast_channel *

ast

)

[static]

Read SIP RTP from channel.

Definition at line 6113 of file chan_sip.c.

References ast_channel::_state, ast_bridged_channel(), ast_debug, AST_FRAME_VOICE, ast_null_frame, ast_set_flag, AST_STATE_UP, ast_test_flag, change_t38_state(), FALSE, sip_pvt::flags, ast_frame::frametype, INV_EARLY_MEDIA, sip_pvt::invitestate, sip_pvt::lastrtprx, ast_channel::name, sip_pvt::pendinginvite, SIP_GOTREFER, SIP_NEEDREINVITE, SIP_PAGE2_T38SUPPORT_UDPTL, SIP_PENDINGBYE, sip_pvt_lock, sip_pvt_unlock, sip_rtp_read(), t38properties::state, sip_pvt::t38, T38_DISABLED, T38_LOCAL_REINVITE, t38properties::t38support, ast_channel::tech_pvt, transmit_reinvite_with_sdp(), and TRUE.

{
   struct ast_frame *fr;
   struct sip_pvt *p = ast->tech_pvt;
   int faxdetected = FALSE;

   sip_pvt_lock(p);
   fr = sip_rtp_read(ast, p, &faxdetected);
   p->lastrtprx = time(NULL);

   /* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
   /* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
   if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
      if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
         if (!p->pendinginvite) {
            ast_debug(3, "Sending reinvite on SIP (%s) for T.38 negotiation.\n", ast->name);
            change_t38_state(p, T38_LOCAL_REINVITE);
            transmit_reinvite_with_sdp(p, TRUE, FALSE);
         }
      } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
         ast_debug(3, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
         ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
      }
   }

   /* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
   if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
      fr = &ast_null_frame;
   }

   sip_pvt_unlock(p);

   return fr;
}

static struct sockaddr_in * sip_real_dst

(

const struct sip_pvt *

p

)

[static]

The real destination address for a write.

Definition at line 2796 of file chan_sip.c.

References ast_test_flag, sip_pvt::flags, sip_proxy::ip, sip_pvt::outboundproxy, sip_pvt::recv, sip_pvt::sa, SIP_NAT, and SIP_NAT_ROUTE.

Referenced by __sip_xmit(), check_via(), retrans_pkt(), send_response(), show_channels_cb(), sip_debug_test_pvt(), and sip_prepare_socket().

{
   if (p->outboundproxy)
      return &p->outboundproxy->ip;

   return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv : &p->sa;
}

static int sip_refer_allocate

(

struct sip_pvt *

p

)

[static]

Allocate SIP refer structure.

Definition at line 10208 of file chan_sip.c.

References ast_calloc, and sip_pvt::refer.

Referenced by get_also_info(), handle_request_invite(), handle_request_refer(), and transmit_refer().

{
   p->refer = ast_calloc(1, sizeof(struct sip_refer)); 
   return p->refer ? 1 : 0;
}

static int sip_reg_timeout

(

const void *

data

)

[static]

Registration timeout, register again Registered as a timeout handler during transmit_register(), to retransmit the packet if a reply does not come back. This is called by the scheduler so the event is not pending anymore when we are called.

Definition at line 9906 of file chan_sip.c.

References __sip_pretend_ack(), ast_dnsmgr_refresh(), ast_log(), sip_registry::call, dialog_unref(), sip_registry::dnsmgr, EVENT_FLAG_SYSTEM, sip_registry::hostname, LOG_NOTICE, manager_event, sip_pvt::needdestroy, REG_STATE_FAILED, REG_STATE_UNREGISTERED, sip_registry::regattempts, sip_pvt::registry, registry_unref(), sip_registry::regstate, regstate2str(), sip_pvt_lock, sip_pvt_unlock, SIP_REGISTER, sip_registry::timeout, transmit_register(), and sip_registry::username.

Referenced by sip_show_sched(), and transmit_register().

{

   /* if we are here, our registration timed out, so we'll just do it over */
   struct sip_registry *r = (struct sip_registry *)data; /* the ref count should have been bumped when the sched item was added */
   struct sip_pvt *p;
   int res;

   /* if we couldn't get a reference to the registry object, punt */
   if (!r)
      return 0;

   if (r->dnsmgr) {
      /* If the registration has timed out, maybe the IP changed.  Force a refresh. */
      ast_dnsmgr_refresh(r->dnsmgr);
   }

   ast_log(LOG_NOTICE, "   -- Registration for '%s@%s' timed out, trying again (Attempt #%d)\n", r->username, r->hostname, r->regattempts); 
   /* If the initial tranmission failed, we may not have an existing dialog,
    * so it is possible that r->call == NULL.
    * Otherwise destroy it, as we have a timeout so we don't want it.
    */
   if (r->call) {
      /* Unlink us, destroy old call.  Locking is not relevant here because all this happens
         in the single SIP manager thread. */
      p = r->call;
      sip_pvt_lock(p);
      p->needdestroy = 1;
      /* Pretend to ACK anything just in case */
      __sip_pretend_ack(p);
      sip_pvt_unlock(p);

      /* decouple the two objects */
      /* p->registry == r, so r has 2 refs, and the unref won't take the object away */
      if (p->registry)
         p->registry = registry_unref(p->registry, "p->registry unreffed");
      r->call = dialog_unref(r->call, "unrefing r->call");
   }
   /* If we have a limit, stop registration and give up */
   if (global_regattempts_max && r->regattempts > global_regattempts_max) {
      /* Ok, enough is enough. Don't try any more */
      /* We could add an external notification here... 
         steal it from app_voicemail :-) */
      ast_log(LOG_NOTICE, "   -- Giving up forever trying to register '%s@%s'\n", r->username, r->hostname);
      r->regstate = REG_STATE_FAILED;
   } else {
      r->regstate = REG_STATE_UNREGISTERED;
      r->timeout = -1;
      res=transmit_register(r, SIP_REGISTER, NULL, NULL);
   }
   manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
   registry_unref(r, "unreffing registry_unref r");
   return 0;
}

static int sip_register	(	const char *	value,
		int	lineno
	)			[static]

Parse register=> line in sip.conf and add to registry.

Definition at line 6528 of file chan_sip.c.

References ast_atomic_fetchadd_int(), ast_calloc, ast_copy_string(), ast_free, ast_log(), ast_string_field_init, ast_string_field_set, ast_strlen_zero(), ASTOBJ_CONTAINER_LINK, ASTOBJ_INIT, sip_registry::authuser, buf, sip_registry::callback, FALSE, hostname, INITIAL_CSEQ, LOG_ERROR, LOG_WARNING, registry_unref(), regl, secret, sip_parse_host(), SIP_TRANSPORT_UDP, sip_registry::transport, and sip_registry::username.

Referenced by build_peer().

{
   struct sip_registry *reg;
   int portnum = 0;
   enum sip_transport transport = SIP_TRANSPORT_UDP;
   char buf[256] = "";
   char *username = NULL;
   char *hostname=NULL, *secret=NULL, *authuser=NULL, *expire=NULL;
   char *callback=NULL;

   if (!value)
      return -1;

   ast_copy_string(buf, value, sizeof(buf));

   /* split [/contact][~expiry] */
   expire = strchr(buf, '~');
   if (expire)
      *expire++ = '\0';
   callback = strrchr(buf, '/');
   if (callback)
      *callback++ = '\0';
   if (ast_strlen_zero(callback))
      callback = "s";

   sip_parse_host(buf, lineno, &username, &portnum, &transport);

   /* First split around the last '@' then parse the two components. */
   hostname = strrchr(username, '@'); /* allow @ in the first part */
   if (hostname)
      *hostname++ = '\0';
   if (ast_strlen_zero(username) || ast_strlen_zero(hostname)) {
      ast_log(LOG_WARNING, "Format for registration is [transport://]user[:secret[:authuser]]@domain[:port][/extension][~expiry] at line %d\n", lineno);
      return -1;
   }
   /* split user[:secret[:authuser]] */
   secret = strchr(username, ':');
   if (secret) {
      *secret++ = '\0';
      authuser = strchr(secret, ':');
      if (authuser)
         *authuser++ = '\0';
   }

   if (!(reg = ast_calloc(1, sizeof(*reg)))) {
      ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n");
      return -1;
   }

   if (ast_string_field_init(reg, 256)) {
      ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry strings\n");
      ast_free(reg);
      return -1;
   }

   ast_atomic_fetchadd_int(&regobjs, 1);
   ASTOBJ_INIT(reg);
   ast_string_field_set(reg, callback, callback);
   if (!ast_strlen_zero(username))
      ast_string_field_set(reg, username, username);
   if (hostname)
      ast_string_field_set(reg, hostname, hostname);
   if (authuser)
      ast_string_field_set(reg, authuser, authuser);
   if (secret)
      ast_string_field_set(reg, secret, secret);
   reg->transport = transport;
   reg->expire = -1;
   reg->expiry = (expire ? atoi(expire) : default_expiry);
   reg->timeout =  -1;
   reg->refresh = reg->expiry;
   reg->portno = portnum;
   reg->callid_valid = FALSE;
   reg->ocseq = INITIAL_CSEQ;
   ASTOBJ_CONTAINER_LINK(&regl, reg); /* Add the new registry entry to the list */
   registry_unref(reg, "unref the reg pointer");   /* release the reference given by ASTOBJ_INIT. The container has another reference */
   return 0;
}

static void sip_registry_destroy

(

struct sip_registry *

reg

)

[static]

Destroy registry object Objects created with the register= statement in static configuration.

Definition at line 4654 of file chan_sip.c.

References ast_atomic_fetchadd_int(), ast_debug, ast_dnsmgr_release(), ast_free, AST_SCHED_DEL, ast_string_field_free_memory, sip_registry::call, dialog_unlink_all(), dialog_unref(), sip_registry::dnsmgr, sip_registry::expire, sip_registry::hostname, sip_pvt::registry, registry_unref(), sched, sip_registry::timeout, TRUE, and sip_registry::username.

Referenced by registry_unref(), reload_config(), and unload_module().

{
   /* Really delete */
   ast_debug(3, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);

   if (reg->call) {
      /* Clear registry before destroying to ensure
         we don't get reentered trying to grab the registry lock */
      reg->call->registry = registry_unref(reg->call->registry, "destroy reg->call->registry");
      ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
      dialog_unlink_all(reg->call, TRUE, TRUE);
      reg->call = dialog_unref(reg->call, "unref reg->call");
      /* reg->call = sip_destroy(reg->call); */
   }
   AST_SCHED_DEL(sched, reg->expire);  
   AST_SCHED_DEL(sched, reg->timeout);
   
   ast_string_field_free_memory(reg);
   ast_atomic_fetchadd_int(&regobjs, -1);
   ast_dnsmgr_release(reg->dnsmgr);
   ast_free(reg);
}

static int sip_reinvite_retry

(

const void *

data

)

[static]

Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite to avoid race conditions between asterisk servers. Called from the scheduler.

Definition at line 15845 of file chan_sip.c.

References ast_set_flag, check_pendings(), dialog_unref(), SIP_NEEDREINVITE, sip_pvt_lock, and sip_pvt_unlock.

Referenced by handle_response_invite(), and sip_show_sched().

{
   struct sip_pvt *p = (struct sip_pvt *) data;

   sip_pvt_lock(p); /* called from schedule thread which requires a lock */
   ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
   p->waitid = -1;
   check_pendings(p);
   sip_pvt_unlock(p);
   dialog_unref(p, "unref the dialog ptr from sip_reinvite_retry, because it held a dialog ptr");
   return 0;
}

static char * sip_reload	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Force reload of module from cli.

Definition at line 23323 of file chan_sip.c.

References ast_mutex_lock(), ast_mutex_unlock(), ast_verbose, CHANNEL_CLI_RELOAD, CHANNEL_MODULE_RELOAD, CLI_GENERATE, CLI_INIT, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, restart_monitor(), sip_reload_lock, TRUE, and ast_cli_entry::usage.

Referenced by reload().

{
   
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip reload";
      e->usage =
         "Usage: sip reload\n"
         "       Reloads SIP configuration from sip.conf\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   ast_mutex_lock(&sip_reload_lock);
   if (sip_reloading) 
      ast_verbose("Previous SIP reload not yet done\n");
   else {
      sip_reloading = TRUE;
      sip_reloadreason = (a && a->fd) ? CHANNEL_CLI_RELOAD : CHANNEL_MODULE_RELOAD;
   }
   ast_mutex_unlock(&sip_reload_lock);
   restart_monitor();

   return CLI_SUCCESS;
}

static struct ast_channel * sip_request_call	(	const char *	type,
		int	format,
		void *	data,
		int *	cause
	)			[static]

PBX interface function -build SIP pvt structure SIP calls initiated by the PBX arrive here.

 * 	SIP Dial string syntax
 *		SIP/exten@host!dnid
 *	or	SIP/host/exten!dnid
 *	or	SIP/host!dnid
 * 

Definition at line 20988 of file chan_sip.c.

References ao2_t_link, ao2_t_unlink, ast_calloc, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_SWITCH_CONGESTION, AST_CAUSE_UNREGISTERED, ast_copy_string(), ast_debug, AST_FORMAT_AUDIO_MASK, ast_getformatname(), ast_getformatname_multiple(), ast_log(), ast_sip_ouraddrfor(), AST_STATE_DOWN, ast_string_field_set, ast_strlen_zero(), ast_update_use_count(), build_callid_pvt(), build_via(), create_addr(), dialog_unlink_all(), dialog_unref(), dialogs, EVENT_FLAG_SYSTEM, ext, sip_socket::fd, LOG_ERROR, LOG_NOTICE, LOG_WARNING, manager_event, sip_pvt::options, sip_pvt::outgoing_call, sip_pvt::peername, restart_monitor(), secret, sip_alloc(), SIP_INVITE, sip_new(), sip_pvt_lock, sip_pvt_unlock, SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, SIP_TRANSPORT_UDP, sip_pvt::socket, TRUE, and sip_socket::type.

{
   struct sip_pvt *p;
   struct ast_channel *tmpc = NULL;
   char *ext = NULL, *host;
   char tmp[256];
   char *dest = data;
   char *dnid;
   char *secret = NULL;
   char *md5secret = NULL;
   char *authname = NULL;
   char *trans = NULL;
   enum sip_transport transport = 0;
   int oldformat = format;

   /* mask request with some set of allowed formats.
    * XXX this needs to be fixed.
    * The original code uses AST_FORMAT_AUDIO_MASK, but it is
    * unclear what to use here. We have global_capabilities, which is
    * configured from sip.conf, and sip_tech.capabilities, which is
    * hardwired to all audio formats.
    */
   format &= AST_FORMAT_AUDIO_MASK;
   if (!format) {
      ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(global_capability));
      *cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;   /* Can't find codec to connect to host */
      return NULL;
   }
   ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));

   if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
      ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest);
      *cause = AST_CAUSE_SWITCH_CONGESTION;
      return NULL;
   }

   p->outgoing_call = TRUE;

   if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
      dialog_unlink_all(p, TRUE, TRUE);
      dialog_unref(p, "unref dialog p from mem fail");
      /* sip_destroy(p); */
      ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");
      *cause = AST_CAUSE_SWITCH_CONGESTION;
      return NULL;
   }

   /* Save the destination, the SIP dial string */
   ast_copy_string(tmp, dest, sizeof(tmp));


   /* Find DNID and take it away */
   dnid = strchr(tmp, '!');
   if (dnid != NULL) {
      *dnid++ = '\0';
      ast_string_field_set(p, todnid, dnid);
   }

   /* Find at sign - @ */
   host = strchr(tmp, '@');
   if (host) {
      *host++ = '\0';
      ext = tmp;
      secret = strchr(ext, ':');
   }
   if (secret) {
      *secret++ = '\0';
      md5secret = strchr(secret, ':');
   }
   if (md5secret) {
      *md5secret++ = '\0';
      authname = strchr(md5secret, ':');
   }
   if (authname) {
      *authname++ = '\0';
      trans = strchr(authname, ':');
   }
   if (trans) {
      *trans++ = '\0';
      if (!strcasecmp(trans, "tcp"))
         transport = SIP_TRANSPORT_TCP;
      else if (!strcasecmp(trans, "tls"))
         transport = SIP_TRANSPORT_TLS;
      else {
         if (strcasecmp(trans, "udp"))
            ast_log(LOG_WARNING, "'%s' is not a valid transport option to Dial() for SIP calls, using udp by default.\n", trans);
         transport = SIP_TRANSPORT_UDP;
      }
   } else { /* use default */
      transport = SIP_TRANSPORT_UDP;
   }

   if (!host) {
      ext = strchr(tmp, '/');
      if (ext) 
         *ext++ = '\0';
      host = tmp;
   }

   p->socket.fd = -1;
   p->socket.type = transport;

   /* We now have 
      host = peer name, DNS host name or DNS domain (for SRV) 
      ext = extension (user part of URI)
      dnid = destination of the call (applies to the To: header)
   */
   if (create_addr(p, host, NULL, 1)) {
      *cause = AST_CAUSE_UNREGISTERED;
      ast_debug(3, "Cant create SIP call - target device not registered\n");
      dialog_unlink_all(p, TRUE, TRUE);
      dialog_unref(p, "unref dialog p UNREGISTERED");
      /* sip_destroy(p); */
      return NULL;
   }
   if (ast_strlen_zero(p->peername) && ext)
      ast_string_field_set(p, peername, ext);
   /* Recalculate our side, and recalculate Call ID */
   ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
   build_via(p);
   ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
   build_callid_pvt(p);
   ao2_t_link(dialogs, p, "Linking in under new name");
   
   /* We have an extension to call, don't use the full contact here */
   /* This to enable dialing registered peers with extension dialling,
      like SIP/peername/extension   
      SIP/peername will still use the full contact 
    */
   if (ext) {
      ast_string_field_set(p, username, ext);
      ast_string_field_set(p, fullcontact, NULL);
   }
   if (secret && !ast_strlen_zero(secret))
      ast_string_field_set(p, peersecret, secret);

   if (md5secret && !ast_strlen_zero(md5secret))
      ast_string_field_set(p, peermd5secret, md5secret);

   if (authname && !ast_strlen_zero(authname))
      ast_string_field_set(p, authname, authname);
#if 0
   printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
#endif
   p->prefcodec = oldformat;           /* Format for this call */
   p->jointcapability = oldformat;
   sip_pvt_lock(p);
   tmpc = sip_new(p, AST_STATE_DOWN, host);  /* Place the call */
   if (global_callevents)
      manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
         "Channel: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
         p->owner? p->owner->name : "", "SIP", p->callid, p->fullcontact, p->peername);
   sip_pvt_unlock(p);
   if (!tmpc) {
      dialog_unlink_all(p, TRUE, TRUE);
      /* sip_destroy(p); */
   }
   dialog_unref(p, "toss pvt ptr at end of sip_request_call");
   ast_update_use_count();
   restart_monitor();
   return tmpc;
}

static int sip_reregister

(

const void *

data

)

[static]

Update registration with SIP Proxy. Called from the scheduler when the previous registration expires, so we don't have to cancel the pending event. We assume the reference so the sip_registry is valid, since it is stored in the scheduled event anyways.

Definition at line 9869 of file chan_sip.c.

References __sip_do_register(), append_history, ast_log(), sip_registry::call, sip_pvt::do_history, sip_registry::expire, sip_registry::hostname, LOG_NOTICE, registry_unref(), and sip_registry::username.

Referenced by handle_response_register(), sip_send_all_registers(), and sip_show_sched().

{
   /* if we are here, we know that we need to reregister. */
   struct sip_registry *r= (struct sip_registry *) data;

   /* if we couldn't get a reference to the registry object, punt */
   if (!r)
      return 0;

   if (r->call && r->call->do_history)
      append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
   /* Since registry's are only added/removed by the the monitor thread, this
      may be overkill to reference/dereference at all here */
   if (sipdebug)
      ast_log(LOG_NOTICE, "   -- Re-registration for  %s@%s\n", r->username, r->hostname);

   r->expire = -1;
   __sip_do_register(r);
   registry_unref(r, "unreg the re-registered");
   return 0;
}

static struct ast_frame * sip_rtp_read	(	struct ast_channel *	ast,
		struct sip_pvt *	p,
		int *	faxdetect
	)			[static]

Read RTP from network.

Definition at line 6033 of file chan_sip.c.

References ast_debug, ast_dsp_process(), AST_FORMAT_AUDIO_MASK, AST_FORMAT_TEXT_MASK, AST_FORMAT_VIDEO_MASK, AST_FRAME_DTMF, AST_FRAME_VOICE, ast_getformatname(), ast_null_frame, ast_rtcp_read(), ast_rtp_read(), ast_set_read_format(), ast_set_write_format(), ast_test_flag, ast_udptl_read(), ast_verbose, f, ast_channel::fdno, sip_pvt::flags, sip_pvt::jointcapability, ast_channel::name, ast_channel::nativeformats, sip_pvt::owner, ast_channel::readformat, sip_pvt::rtp, SIP_DTMF, SIP_DTMF_INBAND, SIP_DTMF_RFC2833, SIP_PAGE2_T38SUPPORT_UDPTL, sip_pvt::t38, t38properties::t38support, sip_pvt::trtp, sip_pvt::udptl, sip_pvt::vad, sip_pvt::vrtp, and ast_channel::writeformat.

Referenced by sip_read().

{
   /* Retrieve audio/etc from channel.  Assumes p->lock is already held. */
   struct ast_frame *f;
   
   if (!p->rtp) {
      /* We have no RTP allocated for this channel */
      return &ast_null_frame;
   }

   switch(ast->fdno) {
   case 0:
      f = ast_rtp_read(p->rtp);  /* RTP Audio */
      break;
   case 1:
      f = ast_rtcp_read(p->rtp); /* RTCP Control Channel */
      break;
   case 2:
      f = ast_rtp_read(p->vrtp); /* RTP Video */
      break;
   case 3:
      f = ast_rtcp_read(p->vrtp);   /* RTCP Control Channel for video */
      break;
   case 4:
      f = ast_rtp_read(p->trtp); /* RTP Text */
      if (sipdebug_text) {
         int i;
         unsigned char* arr = f->data.ptr;
         for (i=0; i < f->datalen; i++)
            ast_verbose("%c", (arr[i] > ' ' && arr[i] < '}') ? arr[i] : '.');
         ast_verbose(" -> ");
         for (i=0; i < f->datalen; i++)
            ast_verbose("%02X ", arr[i]);
         ast_verbose("\n");
      }
      break;
   case 5:
      f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */
      break;
   default:
      f = &ast_null_frame;
   }
   /* Don't forward RFC2833 if we're not supposed to */
   if (f && (f->frametype == AST_FRAME_DTMF) &&
       (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833))
      return &ast_null_frame;

   /* We already hold the channel lock */
   if (!p->owner || (f && f->frametype != AST_FRAME_VOICE))
      return f;

   if (f && f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
      if (!(f->subclass & p->jointcapability)) {
         ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n",
            ast_getformatname(f->subclass), p->owner->name);
         return &ast_null_frame;
      }
      ast_debug(1, "Oooh, format changed to %d %s\n",
         f->subclass, ast_getformatname(f->subclass));
      p->owner->nativeformats = (p->owner->nativeformats & (AST_FORMAT_VIDEO_MASK | AST_FORMAT_TEXT_MASK)) | f->subclass;
      ast_set_read_format(p->owner, p->owner->readformat);
      ast_set_write_format(p->owner, p->owner->writeformat);
   }

   if (f && (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
      f = ast_dsp_process(p->owner, p->vad, f);
      if (f && f->frametype == AST_FRAME_DTMF) {
         if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') {
            ast_debug(1, "Fax CNG detected on %s\n", ast->name);
            *faxdetect = 1;
         } else {
            ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass);
         }
      }
   }
   
   return f;
}

static void sip_scheddestroy	(	struct sip_pvt *	p,
		int	ms
	)			[static]

Schedule destruction of SIP dialog.

Definition at line 3267 of file chan_sip.c.

References __sip_autodestruct(), append_history, ast_log(), ast_sched_add(), ast_verbose, sip_pvt::autokillid, sip_pvt::callid, dialog_ref(), sip_pvt::do_history, LOG_WARNING, sip_pvt::method, sched, sip_cancel_destroy(), sip_debug_test_pvt(), sip_methods, sip_st_dlg::st_active, sip_st_dlg::st_schedid, sip_pvt::stimer, stop_session_timer(), sip_pvt::timer_b, sip_pvt::timer_t1, and TRUE.

Referenced by __sip_autodestruct(), cb_extensionstate(), check_auth(), check_pendings(), handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_notify(), handle_request_options(), handle_request_register(), handle_request_subscribe(), handle_response_invite(), handle_response_register(), manager_sipnotify(), receive_message(), sip_cli_notify(), sip_hangup(), sip_send_mwi_to_peer(), sip_sipredirect(), and transmit_fake_auth_response().

{
   if (ms < 0) {
      if (p->timer_t1 == 0) {
         p->timer_t1 = global_t1;   /* Set timer T1 if not set (RFC 3261) */
         p->timer_b = global_timer_b;  /* Set timer B if not set (RFC 3261) */
      }
      ms = p->timer_t1 * 64;
   }
   if (sip_debug_test_pvt(p))
      ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
   if (sip_cancel_destroy(p))
      ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");

   if (p->do_history)
      append_history(p, "SchedDestroy", "%d ms", ms);
   p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, dialog_ref(p, "setting ref as passing into ast_sched_add for __sip_autodestruct"));

   if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_schedid > 0)
      stop_session_timer(p);
}

static void sip_send_all_registers

(

void

)

[static]

Send all known registrations.

Definition at line 23271 of file chan_sip.c.

References AST_SCHED_REPLACE_UNREF, ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_UNLOCK, ASTOBJ_WRLOCK, registry_addref(), registry_unref(), regl, sched, and sip_reregister().

Referenced by load_module(), and sip_do_reload().

{
   int ms;
   int regspacing;
   if (!regobjs)
      return;
   regspacing = default_expiry * 1000/regobjs;
   if (regspacing > 100)
      regspacing = 100;
   ms = regspacing;
   ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
      ASTOBJ_WRLOCK(iterator);
      ms += regspacing;
      AST_SCHED_REPLACE_UNREF(iterator->expire, sched, ms, sip_reregister, iterator, 
                        registry_unref(_data, "REPLACE sched del decs the refcount"),
                        registry_unref(iterator, "REPLACE sched add failure decs the refcount"),
                        registry_addref(iterator, "REPLACE sched add incs the refcount"));
      ASTOBJ_UNLOCK(iterator);
   } while (0)
   );
}

static int sip_send_mwi_to_peer	(	struct sip_peer *	peer,
		const struct ast_event *	event,
		int	cache_only
	)			[static]

Send message waiting indication to alert peer that they've got voicemail.

Definition at line 20211 of file chan_sip.c.

References sip_peer::addr, ao2_t_link, ao2_t_unlink, ast_app_inboxcount(), ast_event_get_ie_uint(), AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_OLDMSGS, ast_set_flag, ast_sip_ouraddrfor(), ast_str_alloca, ast_test_flag, build_callid_pvt(), build_via(), create_addr_from_peer(), sip_peer::defaddr, DEFAULT_TRANS_TIMEOUT, dialog_ref(), dialog_unlink_all(), dialog_unref(), dialogs, sip_pvt::flags, sip_peer::flags, get_cached_mwi(), sip_peer::mwipvt, sip_pvt::ourip, peer_mailboxes_to_str(), sip_pvt::sa, sip_alloc(), SIP_NOTIFY, SIP_OUTGOING, SIP_PAGE2_SUBSCRIBEMWIONLY, sip_scheddestroy(), sip_pvt::socket, ast_str::str, transmit_notify_with_mwi(), TRUE, sip_socket::type, and sip_peer::vmexten.

Referenced by build_peer(), handle_request_subscribe(), and mwi_event_cb().

{
   /* Called with peerl lock, but releases it */
   struct sip_pvt *p;
   int newmsgs = 0, oldmsgs = 0;

   if (ast_test_flag((&peer->flags[1]), SIP_PAGE2_SUBSCRIBEMWIONLY) && !peer->mwipvt)
      return 0;

   /* Do we have an IP address? If not, skip this peer */
   if (!peer->addr.sin_addr.s_addr && !peer->defaddr.sin_addr.s_addr) 
      return 0;

   if (event) {
      newmsgs = ast_event_get_ie_uint(event, AST_EVENT_IE_NEWMSGS);
      oldmsgs = ast_event_get_ie_uint(event, AST_EVENT_IE_OLDMSGS);
   } else if (!get_cached_mwi(peer, &newmsgs, &oldmsgs)) {
      /* got it!  Don't keep looking. */
   } else if (cache_only) {
      return 0;
   } else { /* Fall back to manually checking the mailbox */
      struct ast_str *mailbox_str = ast_str_alloca(512);
      peer_mailboxes_to_str(&mailbox_str, peer);
      ast_app_inboxcount(mailbox_str->str, &newmsgs, &oldmsgs);
   }
   
   if (peer->mwipvt) {
      /* Base message on subscription */
      p = dialog_ref(peer->mwipvt, "sip_send_mwi_to_peer: Setting dialog ptr p from peer->mwipvt-- should this be done?");
   } else {
      /* Build temporary dialog for this message */
      if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY))) 
         return -1;
      /* If we don't set the socket type to 0, then create_addr_from_peer will fail immediately if the peer
       * uses any transport other than UDP. We set the type to 0 here and then let create_addr_from_peer copy 
       * the peer's socket information to the sip_pvt we just allocated
       */
      p->socket.type = 0;
      if (create_addr_from_peer(p, peer)) {
         /* Maybe they're not registered, etc. */
         dialog_unlink_all(p, TRUE, TRUE);
         dialog_unref(p, "unref dialog p just created via sip_alloc");
         /* sip_destroy(p); */
         return 0;
      }
      /* Recalculate our side, and recalculate Call ID */
      ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
      build_via(p);
      ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
      build_callid_pvt(p);
      ao2_t_link(dialogs, p, "Linking in under new name");
      /* Destroy this session after 32 secs */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   }

   /* Send MWI */
   ast_set_flag(&p->flags[0], SIP_OUTGOING);
   /* the following will decrement the refcount on p as it finishes */
   transmit_notify_with_mwi(p, newmsgs, oldmsgs, peer->vmexten);
   dialog_unref(p, "unref dialog ptr p just before it goes out of scope at the end of sip_send_mwi_to_peer.");
   return 0;
}

static int sip_senddigit_begin	(	struct ast_channel *	ast,
		char	digit
	)			[static]

Definition at line 5511 of file chan_sip.c.

References ast_rtp_senddigit_begin(), ast_test_flag, sip_pvt::flags, sip_pvt::rtp, SIP_DTMF, SIP_DTMF_INBAND, SIP_DTMF_RFC2833, sip_pvt_lock, sip_pvt_unlock, and ast_channel::tech_pvt.

{
   struct sip_pvt *p = ast->tech_pvt;
   int res = 0;

   sip_pvt_lock(p);
   switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
   case SIP_DTMF_INBAND:
      res = -1; /* Tell Asterisk to generate inband indications */
      break;
   case SIP_DTMF_RFC2833:
      if (p->rtp)
         ast_rtp_senddigit_begin(p->rtp, digit);
      break;
   default:
      break;
   }
   sip_pvt_unlock(p);

   return res;
}

static int sip_senddigit_end	(	struct ast_channel *	ast,
		char	digit,
		unsigned int	duration
	)			[static]

Send DTMF character on SIP channel within one call, we're able to transmit in many methods simultaneously.

Definition at line 5535 of file chan_sip.c.

References ast_rtp_senddigit_end(), ast_test_flag, sip_pvt::flags, SIP_DTMF, SIP_DTMF_INBAND, SIP_DTMF_INFO, SIP_DTMF_RFC2833, SIP_DTMF_SHORTINFO, sip_pvt_lock, sip_pvt_unlock, ast_channel::tech_pvt, and transmit_info_with_digit().

{
   struct sip_pvt *p = ast->tech_pvt;
   int res = 0;

   sip_pvt_lock(p);
   switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
   case SIP_DTMF_INFO:
   case SIP_DTMF_SHORTINFO:
      transmit_info_with_digit(p, digit, duration);
      break;
   case SIP_DTMF_RFC2833:
      if (p->rtp)
         ast_rtp_senddigit_end(p->rtp, digit);
      break;
   case SIP_DTMF_INBAND:
      res = -1; /* Tell Asterisk to stop inband indications */
      break;
   }
   sip_pvt_unlock(p);

   return res;
}

static int sip_sendhtml	(	struct ast_channel *	chan,
		int	subclass,
		const char *	data,
		int	datalen
	)			[static]

Send message with Access-URL header, if this is an HTML URL only!

Definition at line 3684 of file chan_sip.c.

References ast_channel::_state, ast_debug, AST_HTML_URL, ast_log(), ast_set_flag, AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, ast_string_field_build, ast_test_flag, chan, FALSE, sip_pvt::flags, sip_pvt::initreq, LOG_WARNING, sip_pvt::pendinginvite, sip_debug_test_pvt(), SIP_NEEDREINVITE, SIP_PENDINGBYE, ast_channel::tech_pvt, transmit_reinvite_with_sdp(), transmit_response(), and url.

{
   struct sip_pvt *p = chan->tech_pvt;

   if (subclass != AST_HTML_URL)
      return -1;

   ast_string_field_build(p, url, "<%s>;mode=active", data);

   if (sip_debug_test_pvt(p))
      ast_debug(1, "Send URL %s, state = %d!\n", data, chan->_state);

   switch (chan->_state) {
   case AST_STATE_RING:
      transmit_response(p, "100 Trying", &p->initreq);
      break;
   case AST_STATE_RINGING:
      transmit_response(p, "180 Ringing", &p->initreq);
      break;
   case AST_STATE_UP:
      if (!p->pendinginvite) {      /* We are up, and have no outstanding invite */
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
         ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);   
      }  
      break;
   default:
      ast_log(LOG_WARNING, "Don't know how to send URI when state is %d!\n", chan->_state);
   }

   return 0;
}

static int sip_sendtext	(	struct ast_channel *	ast,
		const char *	text
	)			[static]

Send SIP MESSAGE text within a call Called from PBX core sendtext() application.

Definition at line 3725 of file chan_sip.c.

References ast_verbose, sip_request::debug, ast_channel::name, sip_debug_test_pvt(), ast_channel::tech_pvt, and transmit_message_with_text().

{
   struct sip_pvt *p = ast->tech_pvt;
   int debug = sip_debug_test_pvt(p);

   if (debug)
      ast_verbose("Sending text %s on %s\n", text, ast->name);
   if (!p)
      return -1;
   /* NOT ast_strlen_zero, because a zero-length message is specifically
    * allowed by RFC 3428 (See section 10, Examples) */
   if (!text)
      return 0;
   if (debug)
      ast_verbose("Really sending text %s on %s\n", text, ast->name);
   transmit_message_with_text(p, text);
   return 0;   
}

static char * sip_set_history	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Enable/Disable SIP History logging (CLI).

Definition at line 15208 of file chan_sip.c.

References ast_cli_args::argc, ast_cli_entry::args, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, FALSE, ast_cli_args::fd, TRUE, and ast_cli_entry::usage.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip set history {on|off}";
      e->usage =
         "Usage: sip set history {on|off}\n"
         "       Enables/Disables recording of SIP dialog history for debugging purposes.\n"
         "       Use 'sip show history' to view the history of a call number.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != e->args)
      return CLI_SHOWUSAGE;

   if (!strncasecmp(a->argv[e->args - 1], "on", 2)) {
      recordhistory = TRUE;
      ast_cli(a->fd, "SIP History Recording Enabled (use 'sip show history')\n");
   } else if (!strncasecmp(a->argv[e->args - 1], "off", 3)) {
      recordhistory = FALSE;
      ast_cli(a->fd, "SIP History Recording Disabled\n");
   } else {
      return CLI_SHOWUSAGE;
   }
   return CLI_SUCCESS;
}

static void sip_set_redirstr	(	struct sip_pvt *	p,
		char *	reason
	)			[static]

Translate referring cause.

Definition at line 11459 of file chan_sip.c.

References ast_string_field_set.

Referenced by get_rdnis().

                                                              {

   if (!strcmp(reason, "unknown")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "user-busy")) {
      ast_string_field_set(p, redircause, "BUSY");
   } else if (!strcmp(reason, "no-answer")) {
      ast_string_field_set(p, redircause, "NOANSWER");
   } else if (!strcmp(reason, "unavailable")) {
      ast_string_field_set(p, redircause, "UNREACHABLE");
   } else if (!strcmp(reason, "unconditional")) {
      ast_string_field_set(p, redircause, "UNCONDITIONAL");
   } else if (!strcmp(reason, "time-of-day")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "do-not-disturb")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "deflection")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "follow-me")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "out-of-service")) {
      ast_string_field_set(p, redircause, "UNREACHABLE");
   } else if (!strcmp(reason, "away")) {
      ast_string_field_set(p, redircause, "UNREACHABLE");
   } else {
      ast_string_field_set(p, redircause, "UNKNOWN");
   }
}

static int sip_set_rtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp *	rtp,
		struct ast_rtp *	vrtp,
		struct ast_rtp *	trtp,
		int	codecs,
		int	nat_active
	)			[static]

Set the RTP peer for this call.

Definition at line 22993 of file chan_sip.c.

References ast_channel::_state, sip_pvt::alreadygone, append_history, ast_debug, ast_inet_ntoa(), ast_rtp_get_peer(), ast_set_flag, AST_STATE_UP, ast_test_flag, sip_pvt::callid, chan, sip_pvt::do_history, FALSE, sip_pvt::flags, sip_pvt::lastrtprx, sip_pvt::lastrtptx, sip_pvt::ourip, sip_pvt::pendinginvite, sip_pvt::redircodecs, sip_pvt::redirip, sip_pvt::rtp, SIP_CAN_REINVITE_NAT, SIP_DEFER_BYE_ON_TRANSFER, SIP_GOTREFER, SIP_NEEDREINVITE, SIP_PENDINGBYE, sip_pvt_lock, sip_pvt_unlock, ast_channel::tech_pvt, transmit_reinvite_with_sdp(), sip_pvt::tredirip, sip_pvt::trtp, sip_pvt::vredirip, and sip_pvt::vrtp.

Referenced by sip_fixup().

{
   struct sip_pvt *p;
   int changed = 0;

   p = chan->tech_pvt;
   if (!p) 
      return -1;

   /* Disable early RTP bridge  */
   if (chan->_state != AST_STATE_UP && !global_directrtpsetup)    /* We are in early state */
      return 0;

   sip_pvt_lock(p);
   if (p->alreadygone) {
      /* If we're destroyed, don't bother */
      sip_pvt_unlock(p);
      return 0;
   }

   /* if this peer cannot handle reinvites of the media stream to devices
      that are known to be behind a NAT, then stop the process now
   */
   if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
      sip_pvt_unlock(p);
      return 0;
   }

   if (rtp) {
      changed |= ast_rtp_get_peer(rtp, &p->redirip);
   } else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) {
      memset(&p->redirip, 0, sizeof(p->redirip));
      changed = 1;
   }
   if (vrtp) {
      changed |= ast_rtp_get_peer(vrtp, &p->vredirip);
   } else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) {
      memset(&p->vredirip, 0, sizeof(p->vredirip));
      changed = 1;
   }
   if (trtp) {
      changed |= ast_rtp_get_peer(trtp, &p->tredirip);
   } else if (p->tredirip.sin_addr.s_addr || ntohs(p->tredirip.sin_port) != 0) {
      memset(&p->tredirip, 0, sizeof(p->tredirip));
      changed = 1;
   }
   if (codecs && (p->redircodecs != codecs)) {
      p->redircodecs = codecs;
      changed = 1;
   }
   if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
      if (chan->_state != AST_STATE_UP) { /* We are in early state */
         if (p->do_history)
            append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
         ast_debug(1, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip.sin_addr));
      } else if (!p->pendinginvite) {     /* We are up, and have no outstanding invite */
         ast_debug(3, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip.sin_addr));
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
         ast_debug(3, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip.sin_addr));
         /* We have a pending Invite. Send re-invite when we're done with the invite */
         ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);   
      }
   }
   /* Reset lastrtprx timer */
   p->lastrtprx = p->lastrtptx = time(NULL);
   sip_pvt_unlock(p);
   return 0;
}

static int sip_set_udptl_peer	(	struct ast_channel *	chan,
		struct ast_udptl *	udptl
	)			[static]

Definition at line 22886 of file chan_sip.c.

References ast_debug, ast_inet_ntoa(), ast_set_flag, ast_test_flag, ast_udptl_get_peer(), sip_pvt::callid, chan, FALSE, sip_pvt::flags, sip_pvt::lastrtprx, sip_pvt::lastrtptx, sip_pvt::ourip, sip_pvt::pendinginvite, SIP_GOTREFER, SIP_NEEDREINVITE, SIP_PENDINGBYE, sip_pvt_lock, sip_pvt_unlock, ast_channel::tech_pvt, transmit_reinvite_with_sdp(), TRUE, sip_pvt::udptl, and sip_pvt::udptlredirip.

{
   struct sip_pvt *p;
   
   p = chan->tech_pvt;
   if (!p)
      return -1;
   sip_pvt_lock(p);
   if (udptl)
      ast_udptl_get_peer(udptl, &p->udptlredirip);
   else
      memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
   if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
      if (!p->pendinginvite) {
         ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip.sin_addr), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
         transmit_reinvite_with_sdp(p, TRUE, FALSE);
      } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
         ast_debug(3, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip.sin_addr), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
         ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
      }
   }
   /* Reset lastrtprx timer */
   p->lastrtprx = p->lastrtptx = time(NULL);
   sip_pvt_unlock(p);
   return 0;
}

static char * sip_show_channel	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Show details of one active dialog.

Definition at line 14635 of file chan_sip.c.

References sip_pvt::allowtransfer, ao2_iterator_init(), ao2_t_iterator_next, ao2_t_ref, ast_cli_args::argc, ast_cli_args::argv, ARRAY_LEN, ast_cli(), ast_getformatname_multiple(), ast_inet_ntoa(), ast_strlen_zero(), ast_test_flag, sip_pvt::callid, sip_pvt::capability, sip_pvt::cid_num, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, cli_yesno(), ast_cli_entry::command, complete_sipch(), dialogs, dtmfmode2str(), ast_cli_args::fd, sip_pvt::flags, sip_route::hop, cfsip_options::id, sip_pvt::jointcapability, sip_pvt::lastmsg, len(), ast_cli_args::line, sip_pvt::maxcallbitrate, ast_cli_args::n, ast_channel::name, nat2str(), ast_channel::nativeformats, sip_pvt::needdestroy, sip_pvt::noncodeccapability, NONE, sip_pvt::ourip, sip_pvt::owner, sip_pvt::peercapability, sip_pvt::peername, ast_cli_args::pos, sip_pvt::recv, sip_pvt::redirip, sip_pvt::route, sip_pvt::sa, SIP_DTMF, SIP_NAT, sip_options, SIP_OUTGOING, SIP_PROMISCREDIR, sip_pvt_lock, sip_pvt_unlock, SIPBUFSIZE, sip_pvt::sipoptions, sip_st_dlg::st_active, sip_st_dlg::st_active_peer_ua, sip_st_dlg::st_cached_max_se, sip_st_dlg::st_cached_min_se, sip_st_dlg::st_cached_mode, sip_st_dlg::st_cached_ref, sip_st_dlg::st_expirys, sip_st_dlg::st_interval, sip_st_dlg::st_ref, sip_st_dlg::st_schedid, sip_pvt::stimer, stmode2str(), strefresher2str(), sip_pvt::subscribed, subscription_type2str(), sip_pvt::tag, cfsip_options::text, sip_pvt::theirtag, transfermode2str(), TRUE, sip_pvt::udptl, sip_pvt::uri, ast_cli_entry::usage, sip_pvt::useragent, sip_pvt::username, sip_pvt::vrtp, and ast_cli_args::word.

{
   struct sip_pvt *cur;
   size_t len;
   int found = 0;
   struct ao2_iterator i;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show channel";
      e->usage =
         "Usage: sip show channel <call-id>\n"
         "       Provides detailed status on a given SIP dialog (identified by SIP call-id).\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sipch(a->line, a->word, a->pos, a->n);
   }

   if (a->argc != 4)
      return CLI_SHOWUSAGE;
   len = strlen(a->argv[3]);
   
   i = ao2_iterator_init(dialogs, 0);

   while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      sip_pvt_lock(cur);

      if (!strncasecmp(cur->callid, a->argv[3], len)) {
         char formatbuf[SIPBUFSIZE/2];
         ast_cli(a->fd, "\n");
         if (cur->subscribed != NONE)
            ast_cli(a->fd, "  * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
         else
            ast_cli(a->fd, "  * SIP Call\n");
         ast_cli(a->fd, "  Curr. trans. direction:  %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming");
         ast_cli(a->fd, "  Call-ID:                %s\n", cur->callid);
         ast_cli(a->fd, "  Owner channel ID:       %s\n", cur->owner ? cur->owner->name : "<none>");
         ast_cli(a->fd, "  Our Codec Capability:   %d\n", cur->capability);
         ast_cli(a->fd, "  Non-Codec Capability (DTMF):   %d\n", cur->noncodeccapability);
         ast_cli(a->fd, "  Their Codec Capability:   %d\n", cur->peercapability);
         ast_cli(a->fd, "  Joint Codec Capability:   %d\n", cur->jointcapability);
         ast_cli(a->fd, "  Format:                 %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0) );
         ast_cli(a->fd, "  T.38 support            %s\n", cli_yesno(cur->udptl != NULL));
         ast_cli(a->fd, "  Video support           %s\n", cli_yesno(cur->vrtp != NULL));
         ast_cli(a->fd, "  MaxCallBR:              %d kbps\n", cur->maxcallbitrate);
         ast_cli(a->fd, "  Theoretical Address:    %s:%d\n", ast_inet_ntoa(cur->sa.sin_addr), ntohs(cur->sa.sin_port));
         ast_cli(a->fd, "  Received Address:       %s:%d\n", ast_inet_ntoa(cur->recv.sin_addr), ntohs(cur->recv.sin_port));
         ast_cli(a->fd, "  SIP Transfer mode:      %s\n", transfermode2str(cur->allowtransfer));
         ast_cli(a->fd, "  NAT Support:            %s\n", nat2str(ast_test_flag(&cur->flags[0], SIP_NAT)));
         ast_cli(a->fd, "  Audio IP:               %s %s\n", ast_inet_ntoa(cur->redirip.sin_addr.s_addr ? cur->redirip.sin_addr : cur->ourip.sin_addr), cur->redirip.sin_addr.s_addr ? "(Outside bridge)" : "(local)" );
         ast_cli(a->fd, "  Our Tag:                %s\n", cur->tag);
         ast_cli(a->fd, "  Their Tag:              %s\n", cur->theirtag);
         ast_cli(a->fd, "  SIP User agent:         %s\n", cur->useragent);
         if (!ast_strlen_zero(cur->username))
            ast_cli(a->fd, "  Username:               %s\n", cur->username);
         if (!ast_strlen_zero(cur->peername))
            ast_cli(a->fd, "  Peername:               %s\n", cur->peername);
         if (!ast_strlen_zero(cur->uri))
            ast_cli(a->fd, "  Original uri:           %s\n", cur->uri);
         if (!ast_strlen_zero(cur->cid_num))
            ast_cli(a->fd, "  Caller-ID:              %s\n", cur->cid_num);
         ast_cli(a->fd, "  Need Destroy:           %s\n", cli_yesno(cur->needdestroy));
         ast_cli(a->fd, "  Last Message:           %s\n", cur->lastmsg);
         ast_cli(a->fd, "  Promiscuous Redir:      %s\n", cli_yesno(ast_test_flag(&cur->flags[0], SIP_PROMISCREDIR)));
         ast_cli(a->fd, "  Route:                  %s\n", cur->route ? cur->route->hop : "N/A");
         ast_cli(a->fd, "  DTMF Mode:              %s\n", dtmfmode2str(ast_test_flag(&cur->flags[0], SIP_DTMF)));
         ast_cli(a->fd, "  SIP Options:            ");
         if (cur->sipoptions) {
            int x;
            for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
               if (cur->sipoptions & sip_options[x].id)
                  ast_cli(a->fd, "%s ", sip_options[x].text);
            }
            ast_cli(a->fd, "\n");
         } else
            ast_cli(a->fd, "(none)\n");

         if (!cur->stimer)
            ast_cli(a->fd, "  Session-Timer:          Uninitiallized\n");
         else {
            ast_cli(a->fd, "  Session-Timer:          %s\n", cur->stimer->st_active ? "Active" : "Inactive");
            if (cur->stimer->st_active == TRUE) {
               ast_cli(a->fd, "  S-Timer Interval:       %d\n", cur->stimer->st_interval);
               ast_cli(a->fd, "  S-Timer Refresher:      %s\n", strefresher2str(cur->stimer->st_ref));
               ast_cli(a->fd, "  S-Timer Expirys:        %d\n", cur->stimer->st_expirys);
               ast_cli(a->fd, "  S-Timer Sched Id:       %d\n", cur->stimer->st_schedid);
               ast_cli(a->fd, "  S-Timer Peer Sts:       %s\n", cur->stimer->st_active_peer_ua ? "Active" : "Inactive");
               ast_cli(a->fd, "  S-Timer Cached Min-SE:  %d\n", cur->stimer->st_cached_min_se);
               ast_cli(a->fd, "  S-Timer Cached SE:      %d\n", cur->stimer->st_cached_max_se);
               ast_cli(a->fd, "  S-Timer Cached Ref:     %s\n", strefresher2str(cur->stimer->st_cached_ref));
               ast_cli(a->fd, "  S-Timer Cached Mode:    %s\n", stmode2str(cur->stimer->st_cached_mode));
            }
         }

         ast_cli(a->fd, "\n\n");

         found++;
      }

      sip_pvt_unlock(cur);

      ao2_t_ref(cur, -1, "toss dialog ptr set by iterator_next");
   }

   if (!found) 
      ast_cli(a->fd, "No such SIP Call ID starting with '%s'\n", a->argv[3]);

   return CLI_SUCCESS;
}

static char* sip_show_channels	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

CLI for show channels or subscriptions. This is a new-style CLI handler so a single function contains the prototype for the function, the 'generator' to produce multiple entries in case it is required, and the actual handler for the command.

Definition at line 14458 of file chan_sip.c.

References ao2_t_callback, ast_cli_args::argc, ast_cli_entry::args, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, dialogs, ESS, ast_cli_args::fd, __show_chan_arg::fd, FORMAT2, FORMAT3, __show_chan_arg::numchans, OBJ_NODATA, show_channels_cb(), __show_chan_arg::subscriptions, and ast_cli_entry::usage.

{
   struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };


   if (cmd == CLI_INIT) {
      e->command = "sip show {channels|subscriptions}";
      e->usage =
         "Usage: sip show channels\n"
         "       Lists all currently active SIP calls (dialogs).\n"
         "Usage: sip show subscriptions\n"
         "       Lists active SIP subscriptions.\n";
      return NULL;
   } else if (cmd == CLI_GENERATE)
      return NULL;

   if (a->argc != e->args)
      return CLI_SHOWUSAGE;
   arg.subscriptions = !strcasecmp(a->argv[e->args - 1], "subscriptions");
   if (!arg.subscriptions)
      ast_cli(arg.fd, FORMAT2, "Peer", "User/ANR", "Call ID", "Format", "Hold", "Last Message", "Expiry");
   else
      ast_cli(arg.fd, FORMAT3, "Peer", "User", "Call ID", "Extension", "Last state", "Type", "Mailbox", "Expiry");

   /* iterate on the container and invoke the callback on each item */
   ao2_t_callback(dialogs, OBJ_NODATA, show_channels_cb, &arg, "callback to show channels");
   
   /* print summary information */
   ast_cli(arg.fd, "%d active SIP %s%s\n", arg.numchans,
      (arg.subscriptions ? "subscription" : "dialog"),
      ESS(arg.numchans));  /* ESS(n) returns an "s" if n>1 */
   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
#undef FORMAT3
}

static char * sip_show_channelstats	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

SIP show channelstats CLI (main function).

Definition at line 14150 of file chan_sip.c.

References ao2_t_callback, ast_cli_args::argc, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, dialogs, ast_cli_args::fd, __show_chan_arg::fd, FORMAT2, __show_chan_arg::numchans, OBJ_NODATA, show_chanstats_cb(), and ast_cli_entry::usage.

{
   struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show channelstats";
      e->usage =
         "Usage: sip show channelstats\n"
         "       Lists all currently active SIP channel's RTCP statistics.\n"
         "       Note that calls in the much optimized RTP P2P bridge mode will not show any packets here.";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != 3)
      return CLI_SHOWUSAGE;

   ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "%", "Jitter", "Send: Pack", "Lost", "Jitter");
   /* iterate on the container and invoke the callback on each item */
   ao2_t_callback(dialogs, OBJ_NODATA, show_chanstats_cb, &arg, "callback to sip show chanstats");
   ast_cli(a->fd, "%d active SIP channel%s\n", arg.numchans, (arg.numchans != 1) ? "s" : ""); 
   return CLI_SUCCESS;
}

static char * sip_show_domains	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

CLI command to list local domains.

Definition at line 13470 of file chan_sip.c.

References ast_cli(), AST_LIST_EMPTY, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, CLI_GENERATE, CLI_INIT, CLI_SUCCESS, ast_cli_entry::command, domain::context, domain::domain, domain_mode_to_text(), ast_cli_args::fd, FORMAT, domain::list, domain::mode, S_OR, and ast_cli_entry::usage.

{
   struct domain *d;
#define FORMAT "%-40.40s %-20.20s %-16.16s\n"

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show domains";
      e->usage =
         "Usage: sip show domains\n"
         "       Lists all configured SIP local domains.\n"
         "       Asterisk only responds to SIP messages to local domains.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (AST_LIST_EMPTY(&domain_list)) {
      ast_cli(a->fd, "SIP Domain support not enabled.\n\n");
      return CLI_SUCCESS;
   } else {
      ast_cli(a->fd, FORMAT, "Our local SIP domains:", "Context", "Set by");
      AST_LIST_LOCK(&domain_list);
      AST_LIST_TRAVERSE(&domain_list, d, list)
         ast_cli(a->fd, FORMAT, d->domain, S_OR(d->context, "(default)"),
            domain_mode_to_text(d->mode));
      AST_LIST_UNLOCK(&domain_list);
      ast_cli(a->fd, "\n");
      return CLI_SUCCESS;
   }
}

static char * sip_show_history	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Show history details of one dialog.

Definition at line 14746 of file chan_sip.c.

References ao2_iterator_init(), ao2_t_iterator_next, ao2_t_ref, ast_cli_args::argc, ast_cli_args::argv, ast_cli(), AST_LIST_TRAVERSE, sip_pvt::callid, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_show_history(), dialogs, sip_history::event, ast_cli_args::fd, sip_pvt::history, len(), ast_cli_args::line, sip_history::list, ast_cli_args::n, NONE, ast_cli_args::pos, sip_pvt_lock, sip_pvt_unlock, sip_pvt::subscribed, ast_cli_entry::usage, and ast_cli_args::word.

{
   struct sip_pvt *cur;
   size_t len;
   int found = 0;
   struct ao2_iterator i;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show history";
      e->usage =
         "Usage: sip show history <call-id>\n"
         "       Provides detailed dialog history on a given SIP call (specified by call-id).\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_show_history(a->line, a->word, a->pos, a->n);
   }

   if (a->argc != 4)
      return CLI_SHOWUSAGE;

   if (!recordhistory)
      ast_cli(a->fd, "\n***Note: History recording is currently DISABLED.  Use 'sip set history on' to ENABLE.\n");

   len = strlen(a->argv[3]);

   i = ao2_iterator_init(dialogs, 0);
   while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      sip_pvt_lock(cur);
      if (!strncasecmp(cur->callid, a->argv[3], len)) {
         struct sip_history *hist;
         int x = 0;

         ast_cli(a->fd, "\n");
         if (cur->subscribed != NONE)
            ast_cli(a->fd, "  * Subscription\n");
         else
            ast_cli(a->fd, "  * SIP Call\n");
         if (cur->history)
            AST_LIST_TRAVERSE(cur->history, hist, list)
               ast_cli(a->fd, "%d. %s\n", ++x, hist->event);
         if (x == 0)
            ast_cli(a->fd, "Call '%s' has no history\n", cur->callid);
         found++;
      }
      sip_pvt_unlock(cur);
      ao2_t_ref(cur, -1, "toss dialog ptr from iterator_next");
   }

   if (!found) 
      ast_cli(a->fd, "No such SIP Call ID starting with '%s'\n", a->argv[3]);

   return CLI_SUCCESS;
}

static char * sip_show_inuse	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

CLI Command to show calls within limits set by call_limit.

Definition at line 12559 of file chan_sip.c.

References ao2_iterator_init(), ao2_lock(), ao2_t_iterator_next, ao2_unlock(), ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_copy_string(), sip_peer::call_limit, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, FALSE, ast_cli_args::fd, FORMAT, FORMAT2, sip_peer::inRinging, sip_peer::inUse, sip_peer::name, sip_peer::onHold, TRUE, unref_peer(), and ast_cli_entry::usage.

{
#define FORMAT "%-25.25s %-15.15s %-15.15s \n"
#define FORMAT2 "%-25.25s %-15.15s %-15.15s \n"
   char ilimits[40];
   char iused[40];
   int showall = FALSE;
   struct ao2_iterator i;
   struct sip_peer *peer;
   
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show inuse";
      e->usage =
         "Usage: sip show inuse [all]\n"
         "       List all SIP devices usage counters and limits.\n"
         "       Add option \"all\" to show all devices, not only those with a limit.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc < 3) 
      return CLI_SHOWUSAGE;

   if (a->argc == 4 && !strcmp(a->argv[3], "all")) 
      showall = TRUE;
   
   ast_cli(a->fd, FORMAT, "* Peer name", "In use", "Limit");

   i = ao2_iterator_init(peers, 0);

   while ((peer = ao2_t_iterator_next(&i, "iterate thru peer table"))) {
      ao2_lock(peer);
      if (peer->call_limit)
         snprintf(ilimits, sizeof(ilimits), "%d", peer->call_limit);
      else 
         ast_copy_string(ilimits, "N/A", sizeof(ilimits));
      snprintf(iused, sizeof(iused), "%d/%d/%d", peer->inUse, peer->inRinging, peer->onHold);
      if (showall || peer->call_limit)
         ast_cli(a->fd, FORMAT2, peer->name, iused, ilimits);
      ao2_unlock(peer);
      unref_peer(peer, "toss iterator pointer");
   }

   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static char * sip_show_objects	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

List all allocated SIP Objects (realtime or static).

Definition at line 13138 of file chan_sip.c.

References ao2_t_callback, ast_cli_args::argc, ast_cli(), ASTOBJ_CONTAINER_DUMP, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, dialog_dump_func(), dialogs, ast_cli_args::fd, OBJ_NODATA, peer_dump_func(), regl, and ast_cli_entry::usage.

{
   char tmp[256];
   
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show objects";
      e->usage =
         "Usage: sip show objects\n"
         "       Lists status of known SIP objects\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }  

   if (a->argc != 3)
      return CLI_SHOWUSAGE;
   ast_cli(a->fd, "-= Peer objects: %d static, %d realtime, %d autocreate =-\n\n", speerobjs, rpeerobjs, apeerobjs);
   ao2_t_callback(peers, OBJ_NODATA, peer_dump_func, &a->fd, "initiate ao2_callback to dump peers");
   ast_cli(a->fd, "-= Registry objects: %d =-\n\n", regobjs);
   ASTOBJ_CONTAINER_DUMP(a->fd, tmp, sizeof(tmp), &regl);
   ast_cli(a->fd, "-= Dialog objects:\n\n");
   ao2_t_callback(dialogs, OBJ_NODATA, dialog_dump_func, &a->fd, "initiate ao2_callback to dump dialogs");
   return CLI_SUCCESS;
}

static char * sip_show_peer	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Show one peer in detail.

Definition at line 13531 of file chan_sip.c.

References _sip_show_peer(), ast_cli_args::argc, ast_cli_args::argv, CLI_GENERATE, CLI_INIT, ast_cli_entry::command, complete_sip_show_peer(), ast_cli_args::fd, ast_cli_args::line, ast_cli_args::n, ast_cli_args::pos, ast_cli_entry::usage, and ast_cli_args::word.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show peer";
      e->usage =
         "Usage: sip show peer <name> [load]\n"
         "       Shows all details on one SIP peer and the current status.\n"
         "       Option \"load\" forces lookup of peer in realtime storage.\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_show_peer(a->line, a->word, a->pos, a->n);
   }
   return _sip_show_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
}

static char * sip_show_peers	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

CLI Show Peers command.

Definition at line 12915 of file chan_sip.c.

References _sip_show_peers(), ast_cli_args::argc, ast_cli_args::argv, CLI_GENERATE, CLI_INIT, ast_cli_entry::command, ast_cli_args::fd, and ast_cli_entry::usage.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show peers";
      e->usage =
         "Usage: sip show peers [like <pattern>]\n"
         "       Lists all known SIP peers.\n"
         "       Optional regular expression pattern is used to filter the peer list.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   return _sip_show_peers(a->fd, NULL, NULL, NULL, a->argc, (const char **) a->argv);
}

static char * sip_show_registry	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Show SIP Registry (registrations with other SIP proxies.

Definition at line 14011 of file chan_sip.c.

References ast_cli_args::argc, ast_cli(), ast_localtime(), ast_strftime(), ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_RDLOCK, ASTOBJ_UNLOCK, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, FORMAT, FORMAT2, regl, regstate2str(), STANDARD_SIP_PORT, and ast_cli_entry::usage.

{
#define FORMAT2 "%-30.30s %-6.6s %-12.12s  %8.8s %-20.20s %-25.25s\n"
#define FORMAT  "%-30.30s %-6.6s %-12.12s  %8d %-20.20s %-25.25s\n"
   char host[80];
   char tmpdat[256];
   struct ast_tm tm;
   int counter = 0;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show registry";
      e->usage =
         "Usage: sip show registry\n"
         "       Lists all registration requests and status.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != 3)
      return CLI_SHOWUSAGE;
   ast_cli(a->fd, FORMAT2, "Host", "dnsmgr", "Username", "Refresh", "State", "Reg.Time");
   
   ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
      ASTOBJ_RDLOCK(iterator);
      snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT);
      if (iterator->regtime.tv_sec) {
         ast_localtime(&iterator->regtime, &tm, NULL);
         ast_strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T", &tm);
      } else 
         tmpdat[0] = '\0';
      ast_cli(a->fd, FORMAT, host, (iterator->dnsmgr) ? "Y" : "N", iterator->username, iterator->refresh, regstate2str(iterator->regstate), tmpdat);
      ASTOBJ_UNLOCK(iterator);
      counter++;
   } while(0));
   ast_cli(a->fd, "%d SIP registrations.\n", counter);
   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static char * sip_show_sched	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Definition at line 13968 of file chan_sip.c.

References __sip_autodestruct(), ast_cli(), ast_sched_report(), ast_str_alloca, auto_congest(), CLI_GENERATE, CLI_INIT, CLI_SUCCESS, ast_cli_entry::command, expire_register(), ast_cli_args::fd, retrans_pkt(), sched, sip_poke_noanswer(), sip_poke_peer_s(), sip_reg_timeout(), sip_reinvite_retry(), sip_reregister(), ast_str::str, and ast_cli_entry::usage.

{
   struct ast_str *cbuf;
   struct ast_cb_names cbnames = {9, { "retrans_pkt",
                                        "__sip_autodestruct",
                                        "expire_register",
                                        "auto_congest",
                                        "sip_reg_timeout",
                                        "sip_poke_peer_s",
                                        "sip_poke_noanswer",
                                        "sip_reregister",
                                        "sip_reinvite_retry"},
                           { retrans_pkt,
                                     __sip_autodestruct,
                                     expire_register,
                                     auto_congest,
                                     sip_reg_timeout,
                                     sip_poke_peer_s,
                                     sip_poke_noanswer,
                                     sip_reregister,
                                     sip_reinvite_retry}};
   
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show sched";
      e->usage =
         "Usage: sip show sched\n"
         "       Shows stats on what's in the sched queue at the moment\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   cbuf = ast_str_alloca(2048);

   ast_cli(a->fd, "\n");
   ast_sched_report(sched, &cbuf, &cbnames);
   ast_cli(a->fd, "%s", cbuf->str);

   return CLI_SUCCESS;
}

static char * sip_show_settings	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

List global settings for the SIP channel.

Definition at line 14179 of file chan_sip.c.

References ast_cli_args::argc, ast_check_realtime(), ast_cli(), ast_getformatname_multiple(), ast_inet_ntoa(), AST_JB_ENABLED, AST_JB_FORCED, AST_JB_LOG, AST_LIST_EMPTY, ast_strlen_zero(), ast_test_flag, ast_tos2str(), authl, buf, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, cli_yesno(), ast_cli_entry::command, default_prefs, default_tls_cfg, dtmfmode2str(), ast_tls_config::enabled, externhost, externrefresh, FALSE, ast_cli_args::fd, sip_proxy::force, global_flags, global_jbconf, global_outboundproxy, sip_settings::ignore_regexpire, ast_jb_conf::impl, ast_tcptls_session_args::local_address, localaddr, ast_jb_conf::max_size, msg, sip_proxy::name, nat2str(), ast_ha::netaddr, ast_ha::netmask, ast_ha::next, sip_settings::peer_rtupdate, prefix, print_codec_to_cli(), ast_jb_conf::resync_threshold, sip_settings::rtsave_sysname, S_OR, sip_cfg, SIP_DTMF, SIP_NAT, SIP_PAGE2_ALLOWOVERLAP, SIP_PAGE2_ALLOWSUBSCRIBE, SIP_PAGE2_RFC2833_COMPENSATE, SIP_PAGE2_RTCACHEFRIENDS, SIP_PAGE2_T38SUPPORT_RTP, SIP_PAGE2_T38SUPPORT_TCP, SIP_PAGE2_T38SUPPORT_UDPTL, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_VIDEOSUPPORT, SIP_PROG_INBAND, SIP_PROG_INBAND_NEVER, SIP_PROG_INBAND_NO, SIP_PROMISCREDIR, sip_tcp_desc, sip_tls_desc, SIP_USECLIENTCODE, SIP_USEREQPHONE, SIPBUFSIZE, stmode2str(), strefresher2str(), transfermode2str(), and ast_cli_entry::usage.

{
   int realtimepeers;
   int realtimeregs;
   char codec_buf[SIPBUFSIZE];
   const char *msg;  /* temporary msg pointer */

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show settings";
      e->usage =
         "Usage: sip show settings\n"
         "       Provides detailed list of the configuration of the SIP channel.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }


   realtimepeers = ast_check_realtime("sippeers");
   realtimeregs = ast_check_realtime("sipregs");

   if (a->argc != 3)
      return CLI_SHOWUSAGE;
   ast_cli(a->fd, "\n\nGlobal Settings:\n");
   ast_cli(a->fd, "----------------\n");
   ast_cli(a->fd, "  UDP SIP Port:           %d\n", ntohs(bindaddr.sin_port));
   ast_cli(a->fd, "  UDP Bindaddress:        %s\n", ast_inet_ntoa(bindaddr.sin_addr));
   ast_cli(a->fd, "  TCP SIP Port:           ");
   if (sip_tcp_desc.local_address.sin_family == AF_INET) {
      ast_cli(a->fd, "%d\n", ntohs(sip_tcp_desc.local_address.sin_port));
      ast_cli(a->fd, "  TCP Bindaddress:        %s\n", ast_inet_ntoa(sip_tcp_desc.local_address.sin_addr));
   } else {
      ast_cli(a->fd, "Disabled\n");
   }
   ast_cli(a->fd, "  TLS SIP Port:           ");
   if (default_tls_cfg.enabled != FALSE) {
      ast_cli(a->fd, "%d\n", ntohs(sip_tls_desc.local_address.sin_port));
      ast_cli(a->fd, "  TLS Bindaddress:        %s\n", ast_inet_ntoa(sip_tls_desc.local_address.sin_addr));
   } else {
      ast_cli(a->fd, "Disabled\n");
   }
   ast_cli(a->fd, "  Videosupport:           %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT)));
   ast_cli(a->fd, "  Textsupport:            %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT)));
   ast_cli(a->fd, "  AutoCreate Peer:        %s\n", cli_yesno(autocreatepeer));
   ast_cli(a->fd, "  Match Auth Username:    %s\n", cli_yesno(global_match_auth_username));
   ast_cli(a->fd, "  Allow unknown access:   %s\n", cli_yesno(global_allowguest));
   ast_cli(a->fd, "  Allow subscriptions:    %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
   ast_cli(a->fd, "  Allow overlap dialing:  %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
   ast_cli(a->fd, "  Allow promsic. redir:   %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
   ast_cli(a->fd, "  Enable call counters:   %s\n", cli_yesno(global_callcounter));
   ast_cli(a->fd, "  SIP domain support:     %s\n", cli_yesno(!AST_LIST_EMPTY(&domain_list)));
   ast_cli(a->fd, "  Realm. auth:            %s\n", cli_yesno(authl != NULL));
   ast_cli(a->fd, "  Our auth realm          %s\n", global_realm);
   ast_cli(a->fd, "  Call to non-local dom.: %s\n", cli_yesno(allow_external_domains));
   ast_cli(a->fd, "  URI user is phone no:   %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_USEREQPHONE)));
   ast_cli(a->fd, "  Always auth rejects:    %s\n", cli_yesno(global_alwaysauthreject));
   ast_cli(a->fd, "  Direct RTP setup:       %s\n", cli_yesno(global_directrtpsetup));
   ast_cli(a->fd, "  User Agent:             %s\n", global_useragent);
   ast_cli(a->fd, "  SDP Session Name:       %s\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
   ast_cli(a->fd, "  SDP Owner Name:         %s\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner);
   ast_cli(a->fd, "  Reg. context:           %s\n", S_OR(global_regcontext, "(not set)"));
   ast_cli(a->fd, "  Regexten on Qualify:    %s\n", cli_yesno(global_regextenonqualify));
   ast_cli(a->fd, "  Caller ID:              %s\n", default_callerid);
   ast_cli(a->fd, "  From: Domain:           %s\n", default_fromdomain);
   ast_cli(a->fd, "  Record SIP history:     %s\n", recordhistory ? "On" : "Off");
   ast_cli(a->fd, "  Call Events:            %s\n", global_callevents ? "On" : "Off");
   ast_cli(a->fd, "  Auth. Failure Events:   %s\n", global_authfailureevents ? "On" : "Off");

   ast_cli(a->fd, "  T38 fax pt UDPTL:       %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
   ast_cli(a->fd, "  T38 fax pt RTP:         %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP)));
   ast_cli(a->fd, "  T38 fax pt TCP:         %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP)));
#endif
   if (!realtimepeers && !realtimeregs)
      ast_cli(a->fd, "  SIP realtime:           Disabled\n" );
   else
      ast_cli(a->fd, "  SIP realtime:           Enabled\n" );
   ast_cli(a->fd, "  Qualify Freq :          %d ms\n", global_qualifyfreq);
   ast_cli(a->fd, "\nNetwork QoS Settings:\n");
   ast_cli(a->fd, "---------------------------\n");
   ast_cli(a->fd, "  IP ToS SIP:             %s\n", ast_tos2str(global_tos_sip));
   ast_cli(a->fd, "  IP ToS RTP audio:       %s\n", ast_tos2str(global_tos_audio));
   ast_cli(a->fd, "  IP ToS RTP video:       %s\n", ast_tos2str(global_tos_video));
   ast_cli(a->fd, "  IP ToS RTP text:        %s\n", ast_tos2str(global_tos_text));
   ast_cli(a->fd, "  802.1p CoS SIP:         %d\n", global_cos_sip);
   ast_cli(a->fd, "  802.1p CoS RTP audio:   %d\n", global_cos_audio);
   ast_cli(a->fd, "  802.1p CoS RTP video:   %d\n", global_cos_video);
   ast_cli(a->fd, "  802.1p CoS RTP text:    %d\n", global_cos_text);
   ast_cli(a->fd, "  Jitterbuffer enabled:   %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_ENABLED)));
   ast_cli(a->fd, "  Jitterbuffer forced:    %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_FORCED)));
   ast_cli(a->fd, "  Jitterbuffer max size:  %ld\n", global_jbconf.max_size);
   ast_cli(a->fd, "  Jitterbuffer resync:    %ld\n", global_jbconf.resync_threshold);
   ast_cli(a->fd, "  Jitterbuffer impl:      %s\n", global_jbconf.impl);
   ast_cli(a->fd, "  Jitterbuffer log:       %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_LOG)));

   ast_cli(a->fd, "\nNetwork Settings:\n");
   ast_cli(a->fd, "---------------------------\n");
   /* determine if/how SIP address can be remapped */
   if (localaddr == NULL)
      msg = "Disabled, no localnet list";
   else if (externip.sin_addr.s_addr == 0)
      msg = "Disabled, externip is 0.0.0.0";
   else if (stunaddr.sin_addr.s_addr != 0)
      msg = "Enabled using STUN";
   else if (!ast_strlen_zero(externhost))
      msg = "Enabled using externhost";
   else
      msg = "Enabled using externip";
   ast_cli(a->fd, "  SIP address remapping:  %s\n", msg);
   ast_cli(a->fd, "  Externhost:             %s\n", S_OR(externhost, "<none>"));
   ast_cli(a->fd, "  Externip:               %s:%d\n", ast_inet_ntoa(externip.sin_addr), ntohs(externip.sin_port));
   ast_cli(a->fd, "  Externrefresh:          %d\n", externrefresh);
   ast_cli(a->fd, "  Internal IP:            %s:%d\n", ast_inet_ntoa(internip.sin_addr), ntohs(internip.sin_port));
   {
      struct ast_ha *d;
      const char *prefix = "Localnet:";
      char buf[INET_ADDRSTRLEN]; /* need to print two addresses */

      for (d = localaddr; d ; prefix = "", d = d->next) {
         ast_cli(a->fd, "  %-24s%s/%s\n",
             prefix, ast_inet_ntoa(d->netaddr),
             inet_ntop(AF_INET, &d->netmask, buf, sizeof(buf)) );
      }
   }
   ast_cli(a->fd, "  STUN server:            %s:%d\n", ast_inet_ntoa(stunaddr.sin_addr), ntohs(stunaddr.sin_port));
 
   ast_cli(a->fd, "\nGlobal Signalling Settings:\n");
   ast_cli(a->fd, "---------------------------\n");
   ast_cli(a->fd, "  Codecs:                 ");
   ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, global_capability);
   ast_cli(a->fd, "%s\n", codec_buf);
   ast_cli(a->fd, "  Codec Order:            ");
   print_codec_to_cli(a->fd, &default_prefs);
   ast_cli(a->fd, "\n");
   ast_cli(a->fd, "  Relax DTMF:             %s\n", cli_yesno(global_relaxdtmf));
   ast_cli(a->fd, "  RFC2833 Compensation:   %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE)));
   ast_cli(a->fd, "  Compact SIP headers:    %s\n", cli_yesno(compactheaders));
   ast_cli(a->fd, "  RTP Keepalive:          %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
   ast_cli(a->fd, "  RTP Timeout:            %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
   ast_cli(a->fd, "  RTP Hold Timeout:       %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
   ast_cli(a->fd, "  MWI NOTIFY mime type:   %s\n", default_notifymime);
   ast_cli(a->fd, "  DNS SRV lookup:         %s\n", cli_yesno(global_srvlookup));
   ast_cli(a->fd, "  Pedantic SIP support:   %s\n", cli_yesno(pedanticsipchecking));
   ast_cli(a->fd, "  Reg. min duration       %d secs\n", min_expiry);
   ast_cli(a->fd, "  Reg. max duration:      %d secs\n", max_expiry);
   ast_cli(a->fd, "  Reg. default duration:  %d secs\n", default_expiry);
   ast_cli(a->fd, "  Outbound reg. timeout:  %d secs\n", global_reg_timeout);
   ast_cli(a->fd, "  Outbound reg. attempts: %d\n", global_regattempts_max);
   ast_cli(a->fd, "  Notify ringing state:   %s\n", cli_yesno(global_notifyringing));
   ast_cli(a->fd, "  Notify hold state:      %s\n", cli_yesno(global_notifyhold));
   ast_cli(a->fd, "  SIP Transfer mode:      %s\n", transfermode2str(global_allowtransfer));
   ast_cli(a->fd, "  Max Call Bitrate:       %d kbps\n", default_maxcallbitrate);
   ast_cli(a->fd, "  Auto-Framing:           %s\n", cli_yesno(global_autoframing));
   ast_cli(a->fd, "  Outb. proxy:            %s %s\n", ast_strlen_zero(global_outboundproxy.name) ? "<not set>" : global_outboundproxy.name,
                     global_outboundproxy.force ? "(forced)" : "");
   ast_cli(a->fd, "  Session Timers:         %s\n", stmode2str(global_st_mode));
   ast_cli(a->fd, "  Session Refresher:      %s\n", strefresher2str (global_st_refresher));
   ast_cli(a->fd, "  Session Expires:        %d secs\n", global_max_se);
   ast_cli(a->fd, "  Session Min-SE:         %d secs\n", global_min_se);
   ast_cli(a->fd, "  Timer T1:               %d\n", global_t1);
   ast_cli(a->fd, "  Timer T1 minimum:       %d\n", global_t1min);
   ast_cli(a->fd, "  Timer B:                %d\n", global_timer_b);

   ast_cli(a->fd, "\nDefault Settings:\n");
   ast_cli(a->fd, "-----------------\n");
   ast_cli(a->fd, "  Context:                %s\n", default_context);
   ast_cli(a->fd, "  Nat:                    %s\n", nat2str(ast_test_flag(&global_flags[0], SIP_NAT)));
   ast_cli(a->fd, "  DTMF:                   %s\n", dtmfmode2str(ast_test_flag(&global_flags[0], SIP_DTMF)));
   ast_cli(a->fd, "  Qualify:                %d\n", default_qualify);
   ast_cli(a->fd, "  Use ClientCode:         %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_USECLIENTCODE)));
   ast_cli(a->fd, "  Progress inband:        %s\n", (ast_test_flag(&global_flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER) ? "Never" : (ast_test_flag(&global_flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NO) ? "No" : "Yes" );
   ast_cli(a->fd, "  Language:               %s\n", default_language);
   ast_cli(a->fd, "  MOH Interpret:          %s\n", default_mohinterpret);
   ast_cli(a->fd, "  MOH Suggest:            %s\n", default_mohsuggest);
   ast_cli(a->fd, "  Voice Mail Extension:   %s\n", default_vmexten);

   
   if (realtimepeers || realtimeregs) {
      ast_cli(a->fd, "\nRealtime SIP Settings:\n");
      ast_cli(a->fd, "----------------------\n");
      ast_cli(a->fd, "  Realtime Peers:         %s\n", cli_yesno(realtimepeers));
      ast_cli(a->fd, "  Realtime Regs:          %s\n", cli_yesno(realtimeregs));
      ast_cli(a->fd, "  Cache Friends:          %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)));
      ast_cli(a->fd, "  Update:                 %s\n", cli_yesno(sip_cfg.peer_rtupdate));
      ast_cli(a->fd, "  Ignore Reg. Expire:     %s\n", cli_yesno(sip_cfg.ignore_regexpire));
      ast_cli(a->fd, "  Save sys. name:         %s\n", cli_yesno(sip_cfg.rtsave_sysname));
      ast_cli(a->fd, "  Auto Clear:             %d\n", global_rtautoclear);
   }
   ast_cli(a->fd, "\n----\n");
   return CLI_SUCCESS;
}

static char* sip_show_tcp	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Show active TCP connections.

Definition at line 12730 of file chan_sip.c.

References ast_cli_args::argc, ast_cli(), ast_inet_ntoa(), AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_tcptls_session_instance::client, ast_cli_entry::command, ast_cli_args::fd, FORMAT, FORMAT2, get_transport(), sip_threadinfo::list, ast_tcptls_session_instance::remote_address, sip_threadinfo::tcptls_session, sip_threadinfo::type, and ast_cli_entry::usage.

{
   struct sip_threadinfo *th;

#define FORMAT2 "%-30.30s %3.6s %9.9s %6.6s\n"
#define FORMAT  "%-30.30s %-6d %-9.9s %-6.6s\n"

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show tcp";
      e->usage =
         "Usage: sip show tcp\n"
         "       Lists all active TCP/TLS sessions.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != 3)
      return CLI_SHOWUSAGE;

   ast_cli(a->fd, FORMAT2, "Host", "Port", "Transport", "Type");
   AST_LIST_LOCK(&threadl);
   AST_LIST_TRAVERSE(&threadl, th, list) {
      ast_cli(a->fd, FORMAT, ast_inet_ntoa(th->tcptls_session->remote_address.sin_addr), 
         ntohs(th->tcptls_session->remote_address.sin_port), 
         get_transport(th->type), 
         (th->tcptls_session->client ? "Client" : "Server"));

   }
   AST_LIST_UNLOCK(&threadl);
   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static char* sip_show_user	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Show one user in detail.

Definition at line 13893 of file chan_sip.c.

References ao2_lock(), ao2_unlock(), ast_cli_args::argc, ast_cli_args::argv, ast_callerid_merge(), ast_cdr_flags2str(), ast_cli(), ast_describe_caller_presentation(), ast_strlen_zero(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, cli_yesno(), ast_cli_entry::command, complete_sip_show_user(), FALSE, ast_cli_args::fd, find_peer(), FINDUSERS, ast_cli_args::line, ast_cli_args::n, ast_variable::name, ast_variable::next, ast_cli_args::pos, print_codec_to_cli(), print_group(), stmode2str(), strefresher2str(), transfermode2str(), TRUE, unref_peer(), ast_cli_entry::usage, user, ast_variable::value, and ast_cli_args::word.

{
   char cbuf[256];
   struct sip_peer *user;
   struct ast_variable *v;
   int load_realtime;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show user";
      e->usage =
         "Usage: sip show user <name> [load]\n"
         "       Shows all details on one SIP user and the current status.\n"
         "       Option \"load\" forces lookup of peer in realtime storage.\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_show_user(a->line, a->word, a->pos, a->n);
   }

   if (a->argc < 4)
      return CLI_SHOWUSAGE;

   /* Load from realtime storage? */
   load_realtime = (a->argc == 5 && !strcmp(a->argv[4], "load")) ? TRUE : FALSE;

   if ((user = find_peer(a->argv[3], NULL, load_realtime, FINDUSERS, FALSE))) {
      ao2_lock(user);
      ast_cli(a->fd, "\n\n");
      ast_cli(a->fd, "  * Name       : %s\n", user->name);
      ast_cli(a->fd, "  Secret       : %s\n", ast_strlen_zero(user->secret)?"<Not set>":"<Set>");
      ast_cli(a->fd, "  MD5Secret    : %s\n", ast_strlen_zero(user->md5secret)?"<Not set>":"<Set>");
      ast_cli(a->fd, "  Context      : %s\n", user->context);
      ast_cli(a->fd, "  Language     : %s\n", user->language);
      if (!ast_strlen_zero(user->accountcode))
         ast_cli(a->fd, "  Accountcode  : %s\n", user->accountcode);
      ast_cli(a->fd, "  AMA flags    : %s\n", ast_cdr_flags2str(user->amaflags));
      ast_cli(a->fd, "  Transfer mode: %s\n", transfermode2str(user->allowtransfer));
      ast_cli(a->fd, "  MaxCallBR    : %d kbps\n", user->maxcallbitrate);
      ast_cli(a->fd, "  CallingPres  : %s\n", ast_describe_caller_presentation(user->callingpres));
      ast_cli(a->fd, "  Call limit   : %d\n", user->call_limit);
      ast_cli(a->fd, "  Callgroup    : ");
      print_group(a->fd, user->callgroup, 0);
      ast_cli(a->fd, "  Pickupgroup  : ");
      print_group(a->fd, user->pickupgroup, 0);
      ast_cli(a->fd, "  Callerid     : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), user->cid_name, user->cid_num, "<unspecified>"));
      ast_cli(a->fd, "  ACL          : %s\n", cli_yesno(user->ha != NULL));
      ast_cli(a->fd, "  Sess-Timers  : %s\n", stmode2str(user->stimer.st_mode_oper));
      ast_cli(a->fd, "  Sess-Refresh : %s\n", strefresher2str(user->stimer.st_ref));
      ast_cli(a->fd, "  Sess-Expires : %d secs\n", user->stimer.st_max_se);
      ast_cli(a->fd, "  Sess-Min-SE  : %d secs\n", user->stimer.st_min_se);

      ast_cli(a->fd, "  Codec Order  : (");
      print_codec_to_cli(a->fd, &user->prefs);
      ast_cli(a->fd, ")\n");

      ast_cli(a->fd, "  Auto-Framing:  %s \n", cli_yesno(user->autoframing));
      if (user->chanvars) {
         ast_cli(a->fd, "  Variables    :\n");
         for (v = user->chanvars ; v ; v = v->next)
            ast_cli(a->fd, "                 %s = %s\n", v->name, v->value);
      }

      ast_cli(a->fd, "\n");

      ao2_unlock(user);
      unref_peer(user, "sip show user");
   } else {
      ast_cli(a->fd, "User %s not found.\n", a->argv[3]);
      ast_cli(a->fd, "\n");
   }

   return CLI_SUCCESS;
}

static char* sip_show_users	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

CLI Command 'SIP Show Users'.

Definition at line 12767 of file chan_sip.c.

References ao2_iterator_init(), ao2_iterator_next, ao2_lock(), ao2_unlock(), ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_test_flag, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, cli_yesno(), ast_cli_entry::command, FALSE, ast_cli_args::fd, FORMAT, nat2str(), SIP_NAT, SIP_TYPE_USER, TRUE, unref_peer(), ast_cli_entry::usage, and user.

{
   regex_t regexbuf;
   int havepattern = FALSE;
   struct ao2_iterator user_iter;
   struct sip_peer *user;

#define FORMAT  "%-25.25s  %-15.15s  %-15.15s  %-15.15s  %-5.5s%-10.10s\n"

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show users";
      e->usage =
         "Usage: sip show users [like <pattern>]\n"
         "       Lists all known SIP users.\n"
         "       Optional regular expression pattern is used to filter the user list.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   switch (a->argc) {
   case 5:
      if (!strcasecmp(a->argv[3], "like")) {
         if (regcomp(&regexbuf, a->argv[4], REG_EXTENDED | REG_NOSUB))
            return CLI_SHOWUSAGE;
         havepattern = TRUE;
      } else
         return CLI_SHOWUSAGE;
   case 3:
      break;
   default:
      return CLI_SHOWUSAGE;
   }

   ast_cli(a->fd, FORMAT, "Username", "Secret", "Accountcode", "Def.Context", "ACL", "NAT");

   user_iter = ao2_iterator_init(peers, 0);
   while ((user = ao2_iterator_next(&user_iter))) {
      ao2_lock(user);
      if (!(user->type & SIP_TYPE_USER)) {
         ao2_unlock(user);
         unref_peer(user, "sip show users");
         continue;
      }

      if (havepattern && regexec(&regexbuf, user->name, 0, NULL, 0)) {
         ao2_unlock(user);
         unref_peer(user, "sip show users");
         continue;
      }

      ast_cli(a->fd, FORMAT, user->name, 
         user->secret, 
         user->accountcode,
         user->context,
         cli_yesno(user->ha != NULL),
         nat2str(ast_test_flag(&user->flags[0], SIP_NAT)));
      ao2_unlock(user);
      unref_peer(user, "sip show users");
   }

   if (havepattern)
      regfree(&regexbuf);

   return CLI_SUCCESS;
#undef FORMAT
}

static int sip_sipredirect	(	struct sip_pvt *	p,
		const char *	dest
	)			[static]

Transfer call before connect with a 302 redirect.

Note:

Called by the transfer() dialplan application through the sip_transfer() pbx interface function if the call is in ringing state

Todo:

Fix this function so that we wait for reply to the REFER and react to errors, denials or other issues the other end might have.

Definition at line 23182 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_strdupa, ast_string_field_build, ast_strlen_zero(), get_header(), sip_pvt::initreq, LOG_ERROR, sip_pvt::our_contact, sip_alreadygone(), sip_scheddestroy(), SIP_TRANS_TIMEOUT, strcasestr(), strsep(), and transmit_response_reliable().

Referenced by sip_transfer().

{
   char *cdest;
   char *extension, *host, *port;
   char tmp[80];

   cdest = ast_strdupa(dest);
   
   extension = strsep(&cdest, "@");
   host = strsep(&cdest, ":");
   port = strsep(&cdest, ":");
   if (ast_strlen_zero(extension)) {
      ast_log(LOG_ERROR, "Missing mandatory argument: extension\n");
      return 0;
   }

   /* we'll issue the redirect message here */
   if (!host) {
      char *localtmp;

      ast_copy_string(tmp, get_header(&p->initreq, "To"), sizeof(tmp));
      if (ast_strlen_zero(tmp)) {
         ast_log(LOG_ERROR, "Cannot retrieve the 'To' header from the original SIP request!\n");
         return 0;
      }
      if ( ( (localtmp = strcasestr(tmp, "sip:")) || (localtmp = strcasestr(tmp, "sips:")) ) 
         && (localtmp = strchr(localtmp, '@'))) {
         char lhost[80], lport[80];

         memset(lhost, 0, sizeof(lhost));
         memset(lport, 0, sizeof(lport));
         localtmp++;
         /* This is okey because lhost and lport are as big as tmp */
         sscanf(localtmp, "%[^<>:; ]:%[^<>:; ]", lhost, lport);
         if (ast_strlen_zero(lhost)) {
            ast_log(LOG_ERROR, "Can't find the host address\n");
            return 0;
         }
         host = ast_strdupa(lhost);
         if (!ast_strlen_zero(lport)) {
            port = ast_strdupa(lport);
         }
      }
   }

   ast_string_field_build(p, our_contact, "Transfer <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : "");
   transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq);

   sip_scheddestroy(p, SIP_TRANS_TIMEOUT);   /* Make sure we stop send this reply. */
   sip_alreadygone(p);
   /* hangup here */
   return 0;
}

static struct sip_st_dlg * sip_st_alloc

(

struct sip_pvt *const

p

)

[static]

Allocate Session-Timers struct w/in dialog.

Definition at line 6190 of file chan_sip.c.

References ast_calloc, ast_log(), LOG_ERROR, sip_st_dlg::st_schedid, and sip_pvt::stimer.

Referenced by handle_request_invite(), and st_get_mode().

{
   struct sip_st_dlg *stp;

   if (p->stimer) {
      ast_log(LOG_ERROR, "Session-Timer struct already allocated\n");
      return p->stimer;
   }

   if (!(stp = ast_calloc(1, sizeof(struct sip_st_dlg))))
      return NULL;

   p->stimer = stp;

   stp->st_schedid = -1;           /* Session-Timers ast_sched scheduler id */

   return p->stimer;
}

static int sip_standard_port	(	enum sip_transport	type,
		int	port
	)			[static]

Returns the port to use for this socket.

Parameters:

	type	The type of transport used
	port	Port we are checking to see if it's the standard port.

Note:

port is expected in host byte order

Definition at line 20029 of file chan_sip.c.

References SIP_TRANSPORT_TLS, STANDARD_SIP_PORT, and STANDARD_TLS_PORT.

Referenced by build_contact(), and initreqprep().

{
   if (type & SIP_TRANSPORT_TLS)
      return port == STANDARD_TLS_PORT;
   else
      return port == STANDARD_SIP_PORT;
}

static int sip_t38_abort

(

const void *

data

)

[static]

Definition at line 17793 of file chan_sip.c.

References change_t38_state(), dialog_unref(), T38_DISABLED, and transmit_response_reliable().

Referenced by handle_request_invite().

{
   struct sip_pvt *p = (struct sip_pvt *) data;

   change_t38_state(p, T38_DISABLED);
   transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
   p->t38id = -1;
   dialog_unref(p, "unref the dialog ptr from sip_t38_abort, because it held a dialog ptr");

   return 0;
}

static struct ast_tcptls_session_instance* sip_tcp_locate

(

struct sockaddr_in *

s

)

[static]

Find thread for TCP/TLS session (based on IP/Port.

Note:

This function returns an astobj2 reference

Definition at line 20042 of file chan_sip.c.

References ao2_ref, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_tcptls_session_instance::remote_address, and sip_threadinfo::tcptls_session.

Referenced by sip_prepare_socket().

{
   struct sip_threadinfo *th;
   struct ast_tcptls_session_instance *tcptls_instance = NULL;

   AST_LIST_LOCK(&threadl);
   AST_LIST_TRAVERSE(&threadl, th, list) {
      if ((s->sin_family == th->tcptls_session->remote_address.sin_family) &&
         (s->sin_addr.s_addr == th->tcptls_session->remote_address.sin_addr.s_addr) &&
         (s->sin_port == th->tcptls_session->remote_address.sin_port))  {
            tcptls_instance = (ao2_ref(th->tcptls_session, +1), th->tcptls_session);
            break;
         }
   }
   AST_LIST_UNLOCK(&threadl);

   return tcptls_instance;
}

static void * sip_tcp_worker_fn

(

void *

)

[static]

SIP TCP connection handler.

Definition at line 2360 of file chan_sip.c.

References _sip_tcp_helper_thread().

Referenced by sip_prepare_socket().

{
   struct ast_tcptls_session_instance *tcptls_session = data;

   return _sip_tcp_helper_thread(NULL, tcptls_session);
}

static int sip_transfer	(	struct ast_channel *	ast,
		const char *	dest
	)			[static]

Transfer SIP call.

Definition at line 5560 of file chan_sip.c.

References ast_channel::_state, AST_STATE_RING, sip_pvt_lock, sip_pvt_unlock, sip_sipredirect(), ast_channel::tech_pvt, and transmit_refer().

{
   struct sip_pvt *p = ast->tech_pvt;
   int res;

   if (dest == NULL) /* functions below do not take a NULL */
      dest = "";
   sip_pvt_lock(p);
   if (ast->_state == AST_STATE_RING)
      res = sip_sipredirect(p, dest);
   else
      res = transmit_refer(p, dest);
   sip_pvt_unlock(p);
   return res;
}

static char * sip_unregister	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)			[static]

Unregister (force expiration) a SIP peer in the registry via CLI.

Note:

This function does not tell the SIP device what's going on, so use it with great care.

Definition at line 14057 of file chan_sip.c.

References ast_cli_args::argc, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_unregister(), sip_peer::expire, expire_register(), ast_cli_args::fd, find_peer(), FINDPEERS, ast_cli_args::line, ast_cli_args::n, ast_cli_args::pos, ref_peer(), TRUE, unref_peer(), ast_cli_entry::usage, and ast_cli_args::word.

{
   struct sip_peer *peer;
   int load_realtime = 0;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip unregister";
      e->usage =
         "Usage: sip unregister <peer>\n"
         "       Unregister (force expiration) a SIP peer from the registry\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_unregister(a->line, a->word, a->pos, a->n);
   }
   
   if (a->argc != 3)
      return CLI_SHOWUSAGE;
   
   if ((peer = find_peer(a->argv[2], NULL, load_realtime, FINDPEERS, TRUE))) {
      if (peer->expire > 0) {
         expire_register(ref_peer(peer, "ref for expire_register"));
         ast_cli(a->fd, "Unregistered peer \'%s\'\n\n", a->argv[2]);
      } else {
         ast_cli(a->fd, "Peer %s not registered\n", a->argv[2]);
      }
      unref_peer(peer, "sip_unregister: unref_peer via sip_unregister: done with peer from find_peer call");
   } else {
      ast_cli(a->fd, "Peer unknown: \'%s\'. Not unregistered.\n", a->argv[2]);
   }
   
   return CLI_SUCCESS;
}

static int sip_uri_cmp	(	const char *	input1,
		const char *	input2
	)			[static]

Definition at line 17710 of file chan_sip.c.

References ast_strdupa, S_OR, sip_uri_headers_cmp(), sip_uri_params_cmp(), and strsep().

Referenced by handle_request_invite().

{
   char *uri1 = ast_strdupa(input1);
   char *uri2 = ast_strdupa(input2);
   char *host1;
   char *host2;
   char *params1;
   char *params2;
   char *headers1;
   char *headers2;

   /* Strip off "sip:" from the URI. We know this is present
    * because it was checked back in parse_request()
    */
   strsep(&uri1, ":");
   strsep(&uri2, ":");

   if ((host1 = strchr(uri1, '@'))) {
      *host1++ = '\0';
   }
   if ((host2 = strchr(uri2, '@'))) {
      *host2++ = '\0';
   }

   /* Check for mismatched username and passwords. This is the
    * only case-sensitive comparison of a SIP URI
    */
   if ((host1 && !host2) ||
         (host2 && !host1) ||
         (host1 && host2 && strcmp(uri1, uri2))) {
      return 1;
   }

   if (!host1)
      host1 = uri1;
   if (!host2)
      host2 = uri2;

   /* Strip off the parameters and headers so we can compare
    * host and port
    */

   if ((params1 = strchr(host1, ';'))) {
      *params1++ = '\0';
   }
   if ((params2 = strchr(host2, ';'))) {
      *params2++ = '\0';
   }

   /* Headers come after parameters, but there may be headers without
    * parameters, thus the S_OR
    */
   if ((headers1 = strchr(S_OR(params1, host1), '?'))) {
      *headers1++ = '\0';
   }
   if ((headers2 = strchr(S_OR(params2, host2), '?'))) {
      *headers2++ = '\0';
   }

   /* Now the host/port are properly isolated. We can get by with a string comparison
    * because the SIP URI checking rules have some interesting exceptions that make
    * this possible. I will note 2 in particular
    * 1. hostnames which resolve to the same IP address as well as a hostname and its
    *    IP address are not considered a match with SIP URI's.
    * 2. If one URI specifies a port and the other does not, then the URIs do not match.
    *    This includes if one URI explicitly contains port 5060 and the other implies it
    *    by not having a port specified.
    */

   if (strcasecmp(host1, host2)) {
      return 1;
   }

   /* Headers have easier rules to follow, so do those first */
   if (sip_uri_headers_cmp(headers1, headers2)) {
      return 1;
   }

   /* And now the parameters. Ugh */
   return sip_uri_params_cmp(params1, params2);
}

static int sip_uri_headers_cmp	(	const char *	input1,
		const char *	input2
	)			[static]

helper routine for sip_uri_cmp

This takes the "headers" from two SIP URIs and determines if the URIs match. The rules for headers is simple. If a header appears in one URI, then it must also appear in the other URI. The order in which the headers appear does not matter.

Parameters:

	input1	Headers from URI 1
	input2	Headers from URI 2

Returns:

Return 0 if the URIs' headers match, 1 if they do not

Definition at line 17664 of file chan_sip.c.

References ast_strdupa, ast_strlen_zero(), strcasestr(), and strsep().

Referenced by sip_uri_cmp().

{
   char *headers1 = NULL;
   char *headers2 = NULL;
   int zerolength1 = 0;
   int zerolength2 = 0;
   int different = 0;
   char *header1;

   if (ast_strlen_zero(input1)) {
      zerolength1 = 1;
   } else {
      headers1 = ast_strdupa(input1);
   }
   
   if (ast_strlen_zero(input2)) {
      zerolength2 = 1;
   } else {
      headers2 = ast_strdupa(input2);
   }

   if ((zerolength1 && !zerolength2) ||
         (zerolength2 && !zerolength1))
      return 1;

   if (zerolength1 && zerolength2)
      return 0;

   /* At this point, we can definitively state that both inputs are
    * not zero-length. First, one more optimization. If the length
    * of the headers is not equal, then we definitely have no match
    */
   if (strlen(headers1) != strlen(headers2)) {
      return 1;
   }

   for (header1 = strsep(&headers1, "&"); header1; header1 = strsep(&headers1, "&")) {
      if (!strcasestr(headers2, header1)) {
         different = 1;
         break;
      }
   }

   return different;
}

static int sip_uri_params_cmp	(	const char *	input1,
		const char *	input2
	)			[static]

helper routine for sip_uri_cmp

This takes the parameters from two SIP URIs and determines if the URIs match. The rules for parameters *suck*. Here's a breakdown 1. If a parameter appears in both URIs, then they must have the same value in order for the URIs to match 2. If one URI has a user, maddr, ttl, or method parameter, then the other URI must also have that parameter and must have the same value in order for the URIs to match 3. All other headers appearing in only one URI are not considered when determining if URIs match

Parameters:

	input1	Parameters from URI 1
	input2	Parameters from URI 2

Returns:

Return 0 if the URIs' parameters match, 1 if they do not

Definition at line 17524 of file chan_sip.c.

References ast_strdupa, and ast_strlen_zero().

Referenced by sip_uri_cmp().

{
   char *params1 = NULL;
   char *params2 = NULL;
   char *pos1;
   char *pos2;
   int zerolength1 = 0;
   int zerolength2 = 0;
   int maddrmatch = 0;
   int ttlmatch = 0;
   int usermatch = 0;
   int methodmatch = 0;

   if (ast_strlen_zero(input1)) {
      zerolength1 = 1;
   } else {
      params1 = ast_strdupa(input1);
   }
   if (ast_strlen_zero(input2)) {
      zerolength2 = 1;
   } else {
      params2 = ast_strdupa(input2);
   }

   /*Quick optimization. If both params are zero-length, then
    * they match
    */
   if (zerolength1 && zerolength2) {
      return 0;
   }

   pos1 = params1;
   while (!ast_strlen_zero(pos1)) {
      char *name1 = pos1;
      char *value1 = strchr(pos1, '=');
      char *semicolon1 = strchr(pos1, ';');
      int matched = 0;
      if (semicolon1) {
         *semicolon1++ = '\0';
      }
      if (!value1) {
         goto fail;
      }
      *value1++ = '\0';
      /* Checkpoint reached. We have the name and value parsed for param1 
       * We have to duplicate params2 each time through the second loop
       * or else we can't search and replace the semicolons with \0 each
       * time
       */
      pos2 = ast_strdupa(params2);
      while (!ast_strlen_zero(pos2)) {
         char *name2 = pos2;
         char *value2 = strchr(pos2, '=');
         char *semicolon2 = strchr(pos2, ';');
         if (semicolon2) {
            *semicolon2++ = '\0';
         }
         if (!value2) {
            goto fail;
         }
         *value2++ = '\0';
         if (!strcasecmp(name1, name2)) {
            if (strcasecmp(value1, value2)) {
               goto fail;
            } else {
               matched = 1;
               break;
            }
         }
         pos2 = semicolon2;
      }
      /* Need to see if the parameter we're looking at is one of the 'must-match' parameters */
      if (!strcasecmp(name1, "maddr")) {
         if (matched) {
            maddrmatch = 1;
         } else {
            goto fail;
         }
      } else if (!strcasecmp(name1, "ttl")) {
         if (matched) {
            ttlmatch = 1;
         } else {
            goto fail;
         }
      } else if (!strcasecmp(name1, "user")) {
         if (matched) {
            usermatch = 1;
         } else {
            goto fail;
         }
      } else if (!strcasecmp(name1, "method")) {
         if (matched) {
            methodmatch = 1;
         } else {
            goto fail;
         }
      }
      pos1 = semicolon1;
   }

   /* We've made it out of that horrible O(m*n) construct and there are no
    * failures yet. We're not done yet, though, because params2 could have
    * an maddr, ttl, user, or method header and params1 did not.
    */
   pos2 = params2;
   while (!ast_strlen_zero(pos2)) {
      char *name2 = pos2;
      char *value2 = strchr(pos2, '=');
      char *semicolon2 = strchr(pos2, ';');
      if (semicolon2) {
         *semicolon2++ = '\0';
      }
      if (!value2) {
         goto fail;
      }
      *value2++ = '\0';
      if ((!strcasecmp(name2, "maddr") && !maddrmatch) ||
            (!strcasecmp(name2, "ttl") && !ttlmatch) ||
            (!strcasecmp(name2, "user") && !usermatch) ||
            (!strcasecmp(name2, "method") && !methodmatch)) {
         goto fail;
      }
   }
   return 0;

fail:
   return 1;
}

static int sip_write	(	struct ast_channel *	ast,
		struct ast_frame *	frame
	)			[static]

Send frame to media channel (rtp).

Definition at line 5356 of file chan_sip.c.

References ast_channel::_state, AST_FORMAT_AUDIO_MASK, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname_multiple(), ast_log(), ast_rtp_new_source(), ast_rtp_write(), ast_set_flag, AST_STATE_UP, ast_test_flag, ast_udptl_write(), change_t38_state(), FALSE, ast_frame::frametype, INV_EARLY_MEDIA, LOG_WARNING, ast_channel::nativeformats, ast_channel::readformat, red_buffer_t140(), SIP_OUTGOING, SIP_PAGE2_T38SUPPORT, SIP_PROGRESS_SENT, sip_pvt_lock, sip_pvt_unlock, ast_frame::subclass, T38_DISABLED, T38_ENABLED, T38_LOCAL_REINVITE, ast_channel::tech_pvt, transmit_reinvite_with_sdp(), transmit_response_with_sdp(), TRUE, ast_channel::writeformat, and XMIT_UNRELIABLE.

{
   struct sip_pvt *p = ast->tech_pvt;
   int res = 0;

   switch (frame->frametype) {
   case AST_FRAME_VOICE:
      if (!(frame->subclass & ast->nativeformats)) {
         char s1[512], s2[512], s3[512];
         ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %s(%d) read/write = %s(%d)/%s(%d)\n",
            frame->subclass, 
            ast_getformatname_multiple(s1, sizeof(s1) - 1, ast->nativeformats & AST_FORMAT_AUDIO_MASK),
            ast->nativeformats & AST_FORMAT_AUDIO_MASK,
            ast_getformatname_multiple(s2, sizeof(s2) - 1, ast->readformat),
            ast->readformat,
            ast_getformatname_multiple(s3, sizeof(s3) - 1, ast->writeformat),
            ast->writeformat);
         return 0;
      }
      if (p) {
         sip_pvt_lock(p);
         if (p->rtp) {
            /* If channel is not up, activate early media session */
            if ((ast->_state != AST_STATE_UP) &&
                !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
                !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
               ast_rtp_new_source(p->rtp);
               p->invitestate = INV_EARLY_MEDIA;
               transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
               ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);  
            } else if (p->t38.state == T38_ENABLED && !p->t38.direct) {
               change_t38_state(p, T38_DISABLED);
               transmit_reinvite_with_sdp(p, FALSE, FALSE);
            } else {
               p->lastrtptx = time(NULL);
               res = ast_rtp_write(p->rtp, frame);
            }
         }
         sip_pvt_unlock(p);
      }
      break;
   case AST_FRAME_VIDEO:
      if (p) {
         sip_pvt_lock(p);
         if (p->vrtp) {
            /* Activate video early media */
            if ((ast->_state != AST_STATE_UP) &&
                !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
                !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
               p->invitestate = INV_EARLY_MEDIA;
               transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
               ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);  
            }
            p->lastrtptx = time(NULL);
            res = ast_rtp_write(p->vrtp, frame);
         }
         sip_pvt_unlock(p);
      }
      break;
   case AST_FRAME_TEXT:
      if (p) {
         sip_pvt_lock(p);
         if (p->red) {
            red_buffer_t140(p->trtp, frame);
         } else {
            if (p->trtp) {
               /* Activate text early media */
               if ((ast->_state != AST_STATE_UP) &&
                   !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
                   !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
                  p->invitestate = INV_EARLY_MEDIA;
                  transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
                  ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);  
               }
               p->lastrtptx = time(NULL);
               res = ast_rtp_write(p->trtp, frame);
            }
         }
         sip_pvt_unlock(p);
      }
      break;
   case AST_FRAME_IMAGE:
      return 0;
      break;
   case AST_FRAME_MODEM:
      if (p) {
         sip_pvt_lock(p);
         /* UDPTL requires two-way communication, so early media is not needed here.
            we simply forget the frames if we get modem frames before the bridge is up.
            Fax will re-transmit.
         */
         if (ast->_state == AST_STATE_UP) {
            if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && p->t38.state == T38_DISABLED) {
               if (!p->pendinginvite) {
                  change_t38_state(p, T38_LOCAL_REINVITE);
                  transmit_reinvite_with_sdp(p, TRUE, FALSE);
               }
            } else if (p->udptl && p->t38.state == T38_ENABLED) {
               res = ast_udptl_write(p->udptl, frame);
            }
         }
         sip_pvt_unlock(p);
      }
      break;
   default: 
      ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype);
      return 0;
   }

   return res;
}

static int sipsock_read	(	int *	id,
		int	fd,
		short	events,
		void *	ignore
	)			[static]

Read data from SIP socket.

Note:

sipsock_read locks the owner channel while we are processing the SIP message

Returns:

1 on error, 0 on success

Note:

Successful messages is connected to SIP call and forwarded to handle_incoming()

Definition at line 19868 of file chan_sip.c.

References AST_DYNSTR_BUILD_FAILED, ast_free, ast_log(), ast_str_create(), ast_str_set(), errno, handle_request_do(), len(), LOG_NOTICE, LOG_WARNING, SIP_MIN_PACKET, SIP_TRANSPORT_UDP, and sipsock.

Referenced by do_monitor().

{
   struct sip_request req;
   struct sockaddr_in sin = { 0, };
   int res;
   socklen_t len = sizeof(sin);
   static char readbuf[65535];

   memset(&req, 0, sizeof(req));
   res = recvfrom(fd, readbuf, sizeof(readbuf) - 1, 0, (struct sockaddr *)&sin, &len);
   if (res < 0) {
#if !defined(__FreeBSD__)
      if (errno == EAGAIN)
         ast_log(LOG_NOTICE, "SIP: Received packet with bad UDP checksum\n");
      else 
#endif
      if (errno != ECONNREFUSED)
         ast_log(LOG_WARNING, "Recv error: %s\n", strerror(errno));
      return 1;
   }

   readbuf[res] = '\0';

   if (!(req.data = ast_str_create(SIP_MIN_PACKET))) {
      return 1;
   }

   if (ast_str_set(&req.data, 0, "%s", readbuf) == AST_DYNSTR_BUILD_FAILED) {
      return -1;
   }

   req.len = res;
   req.socket.fd  = sipsock;
   req.socket.type = SIP_TRANSPORT_UDP;
   req.socket.tcptls_session  = NULL;
   req.socket.port = bindaddr.sin_port;

   handle_request_do(&req, &sin);
   if (req.data) {
      ast_free(req.data);
      req.data = NULL;
   }

   return 1;
}

enum st_mode st_get_mode

(

struct sip_pvt *

p

)

[static]

Get the session-timer mode.

Parameters:

p

pointer to the SIP dialog

Definition at line 20731 of file chan_sip.c.

References FALSE, find_peer(), FINDPEERS, sip_pvt::peername, pp, SESSION_TIMER_MODE_INVALID, sip_st_alloc(), sip_st_dlg::st_cached_mode, sip_pvt::stimer, TRUE, and unref_peer().

Referenced by handle_request_invite(), handle_response_invite(), and transmit_invite().

{
   if (!p->stimer) 
      sip_st_alloc(p);

   if (p->stimer->st_cached_mode != SESSION_TIMER_MODE_INVALID) 
      return p->stimer->st_cached_mode;

   if (p->peername) {
      struct sip_peer *pp = find_peer(p->peername, NULL, TRUE, FINDPEERS, FALSE);
      if (pp) {
         p->stimer->st_cached_mode = pp->stimer.st_mode_oper;
         unref_peer(pp, "unref peer pointer from find_peer call in st_get_mode");
         return pp->stimer.st_mode_oper;
      }
   }

   p->stimer->st_cached_mode = global_st_mode;
   return global_st_mode;
}

enum st_refresher st_get_refresher

(

struct sip_pvt *

p

)

[static]

Get the entity (UAC or UAS) that's acting as the session-timer refresher.

Parameters:

p

pointer to the SIP dialog

Definition at line 20709 of file chan_sip.c.

References FALSE, find_peer(), FINDPEERS, sip_pvt::peername, pp, SESSION_TIMER_REFRESHER_AUTO, sip_st_dlg::st_cached_ref, sip_pvt::stimer, TRUE, and unref_peer().

Referenced by handle_request_invite().

{
   if (p->stimer->st_cached_ref != SESSION_TIMER_REFRESHER_AUTO) 
      return p->stimer->st_cached_ref;

   if (p->peername) {
      struct sip_peer *pp = find_peer(p->peername, NULL, TRUE, FINDPEERS, FALSE);
      if (pp) {
         p->stimer->st_cached_ref = pp->stimer.st_ref;
         unref_peer(pp, "unref peer pointer from find_peer call in st_get_refresher");
         return pp->stimer.st_ref;
      }
   }
   
   p->stimer->st_cached_ref = global_st_refresher;
   return global_st_refresher;
}

int st_get_se	(	struct sip_pvt *	p,
		int	max
	)			[static]

Get Max or Min SE (session timer expiry).

Parameters:

	p	pointer to the SIP dialog
	max	if true, get max se, otherwise min se

Definition at line 20674 of file chan_sip.c.

References FALSE, find_peer(), FINDPEERS, sip_pvt::peername, pp, sip_st_dlg::st_cached_max_se, sip_st_dlg::st_cached_min_se, sip_pvt::stimer, TRUE, and unref_peer().

Referenced by handle_request_invite(), reqprep(), and transmit_invite().

{
   if (max == TRUE) {
      if (p->stimer->st_cached_max_se) {
         return p->stimer->st_cached_max_se;
      } else if (p->peername) {
         struct sip_peer *pp = find_peer(p->peername, NULL, TRUE, FINDPEERS, FALSE);
         if (pp) {
            p->stimer->st_cached_max_se = pp->stimer.st_max_se;
            unref_peer(pp, "unref peer pointer from find_peer call in st_get_se");
            return (p->stimer->st_cached_max_se);
         }
      }
      p->stimer->st_cached_max_se = global_max_se;
      return (p->stimer->st_cached_max_se);
   } else {
      if (p->stimer->st_cached_min_se) {
         return p->stimer->st_cached_min_se;
      } else if (p->peername) {
         struct sip_peer *pp = find_peer(p->peername, NULL, TRUE, FINDPEERS, FALSE);
         if (pp) {
            p->stimer->st_cached_min_se = pp->stimer.st_min_se;
            unref_peer(pp, "unref peer pointer from find_peer call in st_get_se (2)");
            return (p->stimer->st_cached_min_se);
         }
      }
      p->stimer->st_cached_min_se = global_min_se;
      return (p->stimer->st_cached_min_se);
   }
}

static void start_session_timer

(

struct sip_pvt *

p

)

[static]

Session-Timers: Start session timer.

Definition at line 20481 of file chan_sip.c.

References ast_debug, ast_log(), ast_sched_add(), sip_pvt::callid, dialog_ref(), dialog_unref(), LOG_ERROR, LOG_WARNING, proc_session_timer(), sched, sip_st_dlg::st_interval, sip_st_dlg::st_schedid, and sip_pvt::stimer.

Referenced by handle_request_invite(), handle_response_invite(), and restart_session_timer().

{
   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in start_session_timer - %s\n", p->callid);
      return;
   }

   p->stimer->st_schedid  = ast_sched_add(sched, p->stimer->st_interval * 1000 / 2, proc_session_timer, 
         dialog_ref(p, "adding session timer ref"));
   if (p->stimer->st_schedid < 0) {
      dialog_unref(p, "removing session timer ref");
      ast_log(LOG_ERROR, "ast_sched_add failed.\n");
   }
   ast_debug(2, "Session timer started: %d - %s\n", p->stimer->st_schedid, p->callid);
}

static const char* stmode2str

(

enum st_mode

m

)

[static]

Definition at line 12667 of file chan_sip.c.

References map_x_s(), and stmodes.

Referenced by _sip_show_peer(), sip_show_channel(), sip_show_settings(), and sip_show_user().

{
   return map_x_s(stmodes, m, "Unknown");
}

static void stop_media_flows

(

struct sip_pvt *

p

)

[static]

Immediately stop RTP, VRTP and UDPTL as applicable.

Definition at line 16469 of file chan_sip.c.

References ast_rtp_stop(), ast_udptl_stop(), sip_pvt::rtp, sip_pvt::trtp, sip_pvt::udptl, and sip_pvt::vrtp.

Referenced by handle_request_bye(), handle_request_cancel(), handle_response(), and sip_hangup().

{
   /* Immediately stop RTP, VRTP and UDPTL as applicable */
   if (p->rtp)
      ast_rtp_stop(p->rtp);
   if (p->vrtp)
      ast_rtp_stop(p->vrtp);
   if (p->trtp)
      ast_rtp_stop(p->trtp);
   if (p->udptl)
      ast_udptl_stop(p->udptl);
}

static void stop_session_timer

(

struct sip_pvt *

p

)

[static]

Session-Timers: Stop session timer.

Definition at line 20464 of file chan_sip.c.

References ast_debug, ast_log(), AST_SCHED_DEL_UNREF, sip_pvt::callid, dialog_unref(), FALSE, LOG_WARNING, sched, sip_st_dlg::st_active, sip_st_dlg::st_schedid, sip_pvt::stimer, and TRUE.

Referenced by handle_request_bye(), proc_session_timer(), sip_hangup(), and sip_scheddestroy().

{
   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in stop_session_timer - %s\n", p->callid);
      return;
   }

   if (p->stimer->st_active == TRUE) {
      p->stimer->st_active = FALSE;
      AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
            dialog_unref(p, "removing session timer ref"));
      ast_debug(2, "Session timer stopped: %d - %s\n", p->stimer->st_schedid, p->callid);
   }
}

static int str2dtmfmode

(

const char *

str

)

[static]

maps a string to dtmfmode, returns -1 on error

Definition at line 13187 of file chan_sip.c.

References dtmfstr, and map_s_x().

{
   return map_s_x(dtmfstr, str, -1);
}

static enum st_mode str2stmode

(

const char *

s

)

[static]

Definition at line 12672 of file chan_sip.c.

References map_s_x(), and stmodes.

Referenced by build_peer().

{
   return map_s_x(stmodes, s, -1);
}

static enum st_refresher str2strefresher

(

const char *

s

)

[static]

Definition at line 12690 of file chan_sip.c.

References map_s_x(), and strefreshers.

Referenced by build_peer().

{
   return map_s_x(strefreshers, s, -1);
}

static const char * strefresher2str

(

enum st_refresher

r

)

[static]

Definition at line 12685 of file chan_sip.c.

References map_x_s(), and strefreshers.

Referenced by _sip_show_peer(), reqprep(), respprep(), sip_show_channel(), sip_show_settings(), and sip_show_user().

{
   return map_x_s(strefreshers, r, "Unknown");
}

static const char * subscription_type2str

(

enum subscriptiontype

subtype

)

[static]

Show subscription type in string format.

Definition at line 14373 of file chan_sip.c.

References ARRAY_LEN, subscription_types, and cfsubscription_types::text.

Referenced by show_channels_cb(), and sip_show_channel().

{
   int i;

   for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
      if (subscription_types[i].type == subtype) {
         return subscription_types[i].text;
      }
   }
   return subscription_types[0].text;
}

static int t38_get_rate

(

int

t38cap

)

[static]

Get Max T.38 Transmission rate from T38 capabilities.

Definition at line 8507 of file chan_sip.c.

References ast_debug, T38FAX_RATE_12000, T38FAX_RATE_14400, T38FAX_RATE_2400, T38FAX_RATE_4800, T38FAX_RATE_7200, and T38FAX_RATE_9600.

Referenced by add_sdp().

{
   int maxrate = (t38cap & (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400));
   
   if (maxrate & T38FAX_RATE_14400) {
      ast_debug(2, "T38MaxBitRate 14400 found\n");
      return 14400;
   } else if (maxrate & T38FAX_RATE_12000) {
      ast_debug(2, "T38MaxBitRate 12000 found\n");
      return 12000;
   } else if (maxrate & T38FAX_RATE_9600) {
      ast_debug(2, "T38MaxBitRate 9600 found\n");
      return 9600;
   } else if (maxrate & T38FAX_RATE_7200) {
      ast_debug(2, "T38MaxBitRate 7200 found\n");
      return 7200;
   } else if (maxrate & T38FAX_RATE_4800) {
      ast_debug(2, "T38MaxBitRate 4800 found\n");
      return 4800;
   } else if (maxrate & T38FAX_RATE_2400) {
      ast_debug(2, "T38MaxBitRate 2400 found\n");
      return 2400;
   } else {
      ast_debug(2, "Strange, T38MaxBitRate NOT found in peers T38 SDP.\n");
      return 0;
   }
}

static struct sip_peer * temp_peer

(

const char *

name

)

[static]

Create temporary peer (used in autocreatepeer mode).

Definition at line 21526 of file chan_sip.c.

References ao2_t_alloc, ast_atomic_fetchadd_int(), ast_copy_string(), default_prefs, reg_source_db(), set_peer_defaults(), sip_destroy_peer_fn(), and TRUE.

Referenced by register_verify().

{
   struct sip_peer *peer;

   if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct")))
      return NULL;

   ast_atomic_fetchadd_int(&apeerobjs, 1);
   set_peer_defaults(peer);

   ast_copy_string(peer->name, name, sizeof(peer->name));

   peer->selfdestruct = TRUE;
   peer->host_dynamic = TRUE;
   peer->prefs = default_prefs;
   reg_source_db(peer);

   return peer;
}

static void temp_pvt_cleanup

(

void *

)

[static]

Definition at line 8191 of file chan_sip.c.

References ast_free, and ast_string_field_free_memory.

{
   struct sip_pvt *p = data;

   ast_string_field_free_memory(p);

   ast_free(data);
}

static int temp_pvt_init

(

void *

)

[static]

Definition at line 8183 of file chan_sip.c.

References ast_string_field_init, and sip_pvt::do_history.

{
   struct sip_pvt *p = data;

   p->do_history = 0;   /* XXX do we need it ? isn't already all 0 ? */
   return ast_string_field_init(p, 512);
}

static char* terminate_uri

(

char *

uri

)

[static]

Terminate the uri at the first ';' or space. Technically we should ignore escaped space per RFC3261 (19.1.1 etc) but don't do it for the time being. Remember the uri format is:

 *
 *	sip:user:password@host:port;uri-parameters?headers
 *	sips:user:password@host:port;uri-parameters?headers
 *
 *

Definition at line 11239 of file chan_sip.c.

Referenced by check_user_full(), and register_verify().

{
   char *t = uri;
   while (*t && *t > ' ' && *t != ';')
      t++;
   *t = '\0';
   return uri;
}

static char * transfermode2str

(

enum transfermodes

mode

)

const [static]

Convert transfer mode to text string.

Definition at line 12611 of file chan_sip.c.

References TRANSFER_CLOSED, and TRANSFER_OPENFORALL.

Referenced by _sip_show_peer(), sip_show_channel(), sip_show_settings(), and sip_show_user().

{
   if (mode == TRANSFER_OPENFORALL)
      return "open";
   else if (mode == TRANSFER_CLOSED)
      return "closed";
   return "strict";
}

static void transmit_fake_auth_response	(	struct sip_pvt *	p,
		int	sipmethod,
		struct sip_request *	req,
		enum xmittype	reliable
	)			[static]

Send a fake 401 Unauthorized response when the administrator wants to hide the names of local devices from fishers.

Definition at line 11136 of file chan_sip.c.

References AST_DYNSTR_BUILD_FAILED, ast_random(), ast_skip_blanks(), ast_str_set(), ast_str_thread_get(), ast_string_field_build, ast_strlen_zero(), buf, check_auth_buf, CHECK_AUTH_BUF_INITLEN, DEFAULT_TRANS_TIMEOUT, get_header(), sip_request::ignore, sip_pvt::initreq, sip_pvt::randdata, s, SIP_REGISTER, sip_scheddestroy(), SIP_SUBSCRIBE, strsep(), transmit_response(), and transmit_response_with_auth().

Referenced by handle_request_invite(), handle_request_subscribe(), and register_verify().

{
   /* We have to emulate EXACTLY what we'd get with a good peer
    * and a bad password, or else we leak information. */
   const char *response = "407 Proxy Authentication Required";
   const char *reqheader = "Proxy-Authorization";
   const char *respheader = "Proxy-Authenticate";
   const char *authtoken;
   struct ast_str *buf;
   char *c;

   /* table of recognised keywords, and their value in the digest */
   enum keys { K_NONCE, K_LAST };
   struct x {
      const char *key;
      const char *s;
   } *i, keys[] = {
      [K_NONCE] = { "nonce=", "" },
      [K_LAST] = { NULL, NULL}
   };

   if (sipmethod == SIP_REGISTER || sipmethod == SIP_SUBSCRIBE) {
      response = "401 Unauthorized";
      reqheader = "Authorization";
      respheader = "WWW-Authenticate";
   }
   authtoken = get_header(req, reqheader);
   if (req->ignore && !ast_strlen_zero(p->randdata) && ast_strlen_zero(authtoken)) {
      /* This is a retransmitted invite/register/etc, don't reconstruct authentication
       * information */
      transmit_response_with_auth(p, response, req, p->randdata, 0, respheader, 0);
      /* Schedule auto destroy in 32 seconds (according to RFC 3261) */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   } else if (ast_strlen_zero(p->randdata) || ast_strlen_zero(authtoken)) {
      /* We have no auth, so issue challenge and request authentication */
      ast_string_field_build(p, randdata, "%08lx", ast_random()); /* Create nonce for challenge */
      transmit_response_with_auth(p, response, req, p->randdata, 0, respheader, 0);
      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   }

   if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN))) {
      transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq);
      return;
   }

   /* Make a copy of the response and parse it */
   if (ast_str_set(&buf, 0, "%s", authtoken) == AST_DYNSTR_BUILD_FAILED) {
      transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq);
      return;
   }

   c = buf->str;

   while (c && *(c = ast_skip_blanks(c))) { /* lookup for keys */
      for (i = keys; i->key != NULL; i++) {
         const char *separator = ",";  /* default */

         if (strncasecmp(c, i->key, strlen(i->key)) != 0) {
            continue;
         }
         /* Found. Skip keyword, take text in quotes or up to the separator. */
         c += strlen(i->key);
         if (*c == '"') { /* in quotes. Skip first and look for last */
            c++;
            separator = "\"";
         }
         i->s = c;
         strsep(&c, separator);
         break;
      }
      if (i->key == NULL) { /* not found, jump after space or comma */
         strsep(&c, " ,");
      }
   }

   /* Verify nonce from request matches our nonce.  If not, send 401 with new nonce */
   if (strcasecmp(p->randdata, keys[K_NONCE].s)) {
      if (!req->ignore) {
         ast_string_field_build(p, randdata, "%08lx", ast_random());
      }
      transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, FALSE);

      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else {
      transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq);
   }
}

static int transmit_info_with_digit	(	struct sip_pvt *	p,
		const char	digit,
		unsigned int	duration
	)			[static]

Send SIP INFO dtmf message, see Cisco documentation on cisco.com.

Definition at line 10292 of file chan_sip.c.

References add_digit(), ast_test_flag, sip_pvt::flags, sip_pvt::ocseq, reqprep(), send_request(), SIP_DTMF, SIP_DTMF_SHORTINFO, SIP_INFO, and XMIT_RELIABLE.

Referenced by sip_senddigit_end().

{
   struct sip_request req;
   
   reqprep(&req, p, SIP_INFO, 0, 1);
   add_digit(&req, digit, duration, (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO));
   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_info_with_vidupdate

(

struct sip_pvt *

p

)

[static]

Send SIP INFO with video update request.

Definition at line 10302 of file chan_sip.c.

References add_vidupdate(), sip_pvt::ocseq, reqprep(), send_request(), SIP_INFO, and XMIT_RELIABLE.

Referenced by sip_indicate().

{
   struct sip_request req;
   
   reqprep(&req, p, SIP_INFO, 0, 1);
   add_vidupdate(&req);
   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_invite	(	struct sip_pvt *	p,
		int	sipmethod,
		int	sdp,
		int	init
	)			[static]

Build REFER/INVITE/OPTIONS message and transmit it.

Parameters:

	init	0 = Prepare request within dialog, 1= prepare request, new branch, 2= prepare new request and new dialog. do_proxy_auth calls this with init!=2
	p	sip_pvt structure
	sdp	unknown
	sipmethod	unknown

Definition at line 9390 of file chan_sip.c.

References add_header(), add_header_contentLength(), add_sdp(), sip_invite_param::addsipheaders, ALLOWED_METHODS, append_date(), ast_channel_lock, ast_channel_unlock, ast_copy_string(), ast_debug, AST_LIST_TRAVERSE, ast_log(), ast_random(), ast_skip_blanks(), ast_strdupa, ast_strlen_zero(), ast_udptl_offered_from_local(), ast_unescape_semicolon(), ast_var_name(), ast_var_value(), sip_invite_param::auth, sip_invite_param::authheader, sip_pvt::branch, buf, build_via(), chan, ast_var_t::entries, FALSE, sip_request::header, sip_request::headers, initialize_initreq(), sip_pvt::initreq, initreqprep(), sip_pvt::invite_branch, sip_pvt::lastinvite, LOG_WARNING, sip_request::method, ast_channel::name, sip_pvt::notify_headers, sip_pvt::ocseq, sip_pvt::options, sip_pvt::owner, sip_pvt::refer, sip_refer::refer_to, sip_refer::referred_by, sip_refer::referred_by_name, sip_invite_param::replaces, reqprep(), sip_pvt::rtp, send_request(), SESSION_TIMER_MODE_ORIGINATE, SIP_REFER, SIPBUFSIZE, sip_st_dlg::st_active, st_get_mode(), st_get_se(), sip_st_dlg::st_interval, t38properties::state, sip_pvt::stimer, SUPPORTED_EXTENSIONS, sip_pvt::t38, T38_LOCAL_REINVITE, TRUE, sip_pvt::udptl, var, ast_channel::varshead, XMIT_CRITICAL, and XMIT_RELIABLE.

Referenced by do_proxy_auth(), handle_request_invite(), proc_422_rsp(), sip_call(), and sip_poke_peer().

{
   struct sip_request req;
   struct ast_variable *var;
   
   req.method = sipmethod;
   if (init) {/* Bump branch even on initial requests */
      p->branch ^= ast_random();
      p->invite_branch = p->branch;
      build_via(p);
   }
   if (init > 1)
      initreqprep(&req, p, sipmethod);
   else
      /* If init=1, we should not generate a new branch. If it's 0, we need a new branch. */
      reqprep(&req, p, sipmethod, 0, init ? 0 : 1);
      
   if (p->options && p->options->auth)
      add_header(&req, p->options->authheader, p->options->auth);
   append_date(&req);
   if (sipmethod == SIP_REFER) { /* Call transfer */
      if (p->refer) {
         char buf[SIPBUFSIZE];
         if (!ast_strlen_zero(p->refer->refer_to))
            add_header(&req, "Refer-To", p->refer->refer_to);
         if (!ast_strlen_zero(p->refer->referred_by)) {
            snprintf(buf, sizeof(buf), "%s <%s>", p->refer->referred_by_name, p->refer->referred_by);
            add_header(&req, "Referred-By", buf);
         }
      }
   }
   /* This new INVITE is part of an attended transfer. Make sure that the
   other end knows and replace the current call with this new call */
   if (p->options && !ast_strlen_zero(p->options->replaces)) {
      add_header(&req, "Replaces", p->options->replaces);
      add_header(&req, "Require", "replaces");
   }

   /* Add Session-Timers related headers */
   if (st_get_mode(p) == SESSION_TIMER_MODE_ORIGINATE) {
      char i2astr[10];

      if (!p->stimer->st_interval)
         p->stimer->st_interval = st_get_se(p, TRUE);

      p->stimer->st_active = TRUE;
      
      snprintf(i2astr, sizeof(i2astr), "%d", p->stimer->st_interval);
      add_header(&req, "Session-Expires", i2astr);
      snprintf(i2astr, sizeof(i2astr), "%d", st_get_se(p, FALSE));
      add_header(&req, "Min-SE", i2astr);
   }

   add_header(&req, "Allow", ALLOWED_METHODS);
   add_header(&req, "Supported", SUPPORTED_EXTENSIONS);

   if(p->notify_headers) {
      char buf[512];
      for (var = p->notify_headers; var; var = var->next) {
         ast_copy_string(buf, var->value, sizeof(buf));
         add_header(&req, var->name, ast_unescape_semicolon(buf));
      }
   }
   if (p->options && p->options->addsipheaders && p->owner) {
      struct ast_channel *chan = p->owner; /* The owner channel */
      struct varshead *headp;
   
      ast_channel_lock(chan);

      headp = &chan->varshead;

      if (!headp)
         ast_log(LOG_WARNING, "No Headp for the channel...ooops!\n");
      else {
         const struct ast_var_t *current;
         AST_LIST_TRAVERSE(headp, current, entries) {  
            /* SIPADDHEADER: Add SIP header to outgoing call */
            if (!strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
               char *content, *end;
               const char *header = ast_var_value(current);
               char *headdup = ast_strdupa(header);

               /* Strip of the starting " (if it's there) */
               if (*headdup == '"')
                  headdup++;
               if ((content = strchr(headdup, ':'))) {
                  *content++ = '\0';
                  content = ast_skip_blanks(content); /* Skip white space */
                  /* Strip the ending " (if it's there) */
                  end = content + strlen(content) -1; 
                  if (*end == '"')
                     *end = '\0';
               
                  add_header(&req, headdup, content);
                  if (sipdebug)
                     ast_debug(1, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content);
               }
            }
         }
      }

      ast_channel_unlock(chan);
   }
   if (sdp) {
      if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
         ast_udptl_offered_from_local(p->udptl, 1);
         ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
         add_sdp(&req, p, FALSE, FALSE, TRUE);
      } else if (p->rtp) 
         add_sdp(&req, p, FALSE, TRUE, FALSE);
   } else {
      if (!p->notify_headers) {
         add_header_contentLength(&req, 0);
      }
   }

   if (!p->initreq.headers || init > 2)
      initialize_initreq(p, &req);
   p->lastinvite = p->ocseq;
   return send_request(p, &req, init ? XMIT_CRITICAL : XMIT_RELIABLE, p->ocseq);
}

static int transmit_message_with_text	(	struct sip_pvt *	p,
		const char *	text
	)			[static]

Transmit text with SIP MESSAGE method.

Definition at line 10198 of file chan_sip.c.

References add_text(), sip_pvt::ocseq, reqprep(), send_request(), SIP_MESSAGE, and XMIT_RELIABLE.

Referenced by sip_park_thread(), and sip_sendtext().

{
   struct sip_request req;
   
   reqprep(&req, p, SIP_MESSAGE, 0, 1);
   add_text(&req, text);
   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_notify_custom	(	struct sip_pvt *	p,
		struct ast_variable *	vars
	)			[static]

Notify device with custom headers from sip_notify.conf.

Definition at line 9761 of file chan_sip.c.

References add_header(), ast_copy_string(), ast_debug, ast_unescape_semicolon(), ast_variable_new(), buf, sip_request::headers, initialize_initreq(), sip_pvt::initreq, initreqprep(), ast_variable::name, ast_variable::next, sip_pvt::notify_headers, sip_pvt::ocseq, send_request(), SIP_NOTIFY, ast_variable::value, var, and XMIT_UNRELIABLE.

Referenced by manager_sipnotify(), and sip_cli_notify().

                                                                                {
   struct sip_request req;
   struct ast_variable *var, *newvar;

   initreqprep(&req, p, SIP_NOTIFY);

   /* Copy notify vars and add headers */
   p->notify_headers = newvar = ast_variable_new("Subscription-State", "terminated", "");
   add_header(&req, newvar->name, newvar->value);
   for (var = vars; var; var = var->next) {
      char buf[512];
      ast_debug(2, "  Adding pair %s=%s\n", var->name, var->value);
      ast_copy_string(buf, var->value, sizeof(buf));
      add_header(&req, var->name, ast_unescape_semicolon(buf));
      newvar->next = ast_variable_new(var->name, var->value, "");
      newvar = newvar->next;
   }

   if (!p->initreq.headers) { /* Initialize first request before sending */
      initialize_initreq(p, &req);
   }

   return send_request(p, &req, XMIT_UNRELIABLE, p->ocseq);
}

static int transmit_notify_with_mwi	(	struct sip_pvt *	p,
		int	newmsgs,
		int	oldmsgs,
		char *	vmexten
	)			[static]

Notify user of messages waiting in voicemail (RFC3842).

Note:

- Notification only works for registered peers with mailbox= definitions in sip.conf

• We use the SIP Event package message-summary MIME type defaults to "application/simple-message-summary";

Definition at line 9701 of file chan_sip.c.

References add_header(), add_header_contentLength(), add_line(), ast_inet_ntoa(), ast_str_alloca, ast_str_append(), ast_test_flag, sip_pvt::expiry, sip_pvt::flags, sip_pvt::fromdomain, sip_request::headers, initialize_initreq(), sip_pvt::initreq, initreqprep(), sip_pvt::ocseq, sip_pvt::ourip, S_OR, send_request(), SIP_NOTIFY, SIP_PAGE2_BUGGY_MWI, ast_str::str, sip_pvt::subscribed, ast_str::used, and XMIT_RELIABLE.

Referenced by sip_send_mwi_to_peer().

{
   struct sip_request req;
   struct ast_str *out = ast_str_alloca(500);
   
   initreqprep(&req, p, SIP_NOTIFY);
   add_header(&req, "Event", "message-summary");
   add_header(&req, "Content-Type", default_notifymime);

   ast_str_append(&out, 0, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no");
   ast_str_append(&out, 0, "Message-Account: sip:%s@%s\r\n",
      S_OR(vmexten, default_vmexten), S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr)));
   /* Cisco has a bug in the SIP stack where it can't accept the
      (0/0) notification. This can temporarily be disabled in
      sip.conf with the "buggymwi" option */
   ast_str_append(&out, 0, "Voice-Message: %d/%d%s\r\n",
      newmsgs, oldmsgs, (ast_test_flag(&p->flags[1], SIP_PAGE2_BUGGY_MWI) ? "" : " (0/0)"));

   if (p->subscribed) {
      if (p->expiry)
         add_header(&req, "Subscription-State", "active");
      else  /* Expired */
         add_header(&req, "Subscription-State", "terminated;reason=timeout");
   }

   add_header_contentLength(&req, out->used);
   add_line(&req, out->str);

   if (!p->initreq.headers) 
      initialize_initreq(p, &req);
   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_notify_with_sipfrag	(	struct sip_pvt *	p,
		int	cseq,
		char *	message,
		int	terminate
	)			[static]

Notify a transferring party of the status of transfer (RFC3515).

Definition at line 9735 of file chan_sip.c.

References add_header(), add_header_contentLength(), add_line(), ALLOWED_METHODS, sip_request::headers, initialize_initreq(), sip_pvt::initreq, sip_pvt::lastnoninvite, sip_pvt::ocseq, reqprep(), send_request(), SIP_NOTIFY, SIPBUFSIZE, SUPPORTED_EXTENSIONS, and XMIT_RELIABLE.

Referenced by handle_request_refer(), local_attended_transfer(), and sip_park_thread().

{
   struct sip_request req;
   char tmp[SIPBUFSIZE/2];
   
   reqprep(&req, p, SIP_NOTIFY, 0, 1);
   snprintf(tmp, sizeof(tmp), "refer;id=%d", cseq);
   add_header(&req, "Event", tmp);
   add_header(&req, "Subscription-state", terminate ? "terminated;reason=noresource" : "active");
   add_header(&req, "Content-Type", "message/sipfrag;version=2.0");
   add_header(&req, "Allow", ALLOWED_METHODS);
   add_header(&req, "Supported", SUPPORTED_EXTENSIONS);

   snprintf(tmp, sizeof(tmp), "SIP/2.0 %s\r\n", message);
   add_header_contentLength(&req, strlen(tmp));
   add_line(&req, tmp);

   if (!p->initreq.headers)
      initialize_initreq(p, &req);

   p->lastnoninvite = p->ocseq;

   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_refer	(	struct sip_pvt *	p,
		const char *	dest
	)			[static]

Transmit SIP REFER message (initiated by the transfer() dialplan application.

Note:

this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails.

Todo:

Fix the transfer() dialplan function so that a transfer may fail

Todo:

In theory, we should hang around and wait for a reply, before returning to the dial plan here. Don't know really how that would affect the transfer() app or the pbx, but, well, to make this useful we should have a STATUS code on transfer().

Definition at line 10219 of file chan_sip.c.

References add_header(), ALLOWED_METHODS, ast_copy_string(), ast_debug, ast_log(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, sip_pvt::callid, sip_pvt::flags, get_header(), get_in_brackets(), sip_request::headers, sip_pvt::initreq, LOG_NOTICE, sip_pvt::ocseq, sip_pvt::our_contact, sip_pvt::refer, REFER_SENT, sip_refer::refer_to, sip_refer::referred_by, reqprep(), send_request(), SIP_OUTGOING, SIP_REFER, sip_refer_allocate(), sip_refer::status, SUPPORTED_EXTENSIONS, sip_pvt::tag, sip_pvt::theirtag, and XMIT_RELIABLE.

Referenced by sip_transfer().

{
   struct sip_request req = { 
      .headers = 0,  
   };
   char from[256];
   const char *of;
   char *c;
   char referto[256];
   char *ttag, *ftag;
   char *theirtag = ast_strdupa(p->theirtag);

   if (sipdebug)
      ast_debug(1, "SIP transfer of %s to %s\n", p->callid, dest);

   /* Are we transfering an inbound or outbound call ? */
   if (ast_test_flag(&p->flags[0], SIP_OUTGOING))  {
      of = get_header(&p->initreq, "To");
      ttag = theirtag;
      ftag = p->tag;
   } else {
      of = get_header(&p->initreq, "From");
      ftag = theirtag;
      ttag = p->tag;
   }

   ast_copy_string(from, of, sizeof(from));
   of = get_in_brackets(from);
   ast_string_field_set(p, from, of);
   if (!strncasecmp(of, "sip:", 4))
      of += 4;
   else if (!strncasecmp(of, "sips:", 5))
      of += 5;
   else
      ast_log(LOG_NOTICE, "From address missing 'sip(s):', using it anyway\n");
   /* Get just the username part */
   if ((c = strchr(dest, '@')))
      c = NULL;
   else if ((c = strchr(of, '@')))
      *c++ = '\0';
   if (c) 
      snprintf(referto, sizeof(referto), "<sip:%s@%s>", dest, c);
   else
      snprintf(referto, sizeof(referto), "<sip:%s>", dest);

   /* save in case we get 407 challenge */
   sip_refer_allocate(p);
   ast_copy_string(p->refer->refer_to, referto, sizeof(p->refer->refer_to));
   ast_copy_string(p->refer->referred_by, p->our_contact, sizeof(p->refer->referred_by));
   p->refer->status = REFER_SENT;   /* Set refer status */

   reqprep(&req, p, SIP_REFER, 0, 1);

   add_header(&req, "Refer-To", referto);
   add_header(&req, "Allow", ALLOWED_METHODS);
   add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
   if (!ast_strlen_zero(p->our_contact))
      add_header(&req, "Referred-By", p->our_contact);

   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);

   /* We should propably wait for a NOTIFY here until we ack the transfer */
   /* Maybe fork a new thread and wait for a STATUS of REFER_200OK on the refer status before returning to app_transfer */

   /*! \todo In theory, we should hang around and wait for a reply, before
   returning to the dial plan here. Don't know really how that would
   affect the transfer() app or the pbx, but, well, to make this
   useful we should have a STATUS code on transfer().
   */
}

static int transmit_register	(	struct sip_registry *	r,
		int	sipmethod,
		const char *	auth,
		const char *	authheader
	)			[static]

Transmit register to SIP proxy or UA auth = NULL on the initial registration (from sip_reregister()).

Definition at line 9964 of file chan_sip.c.

References add_header(), add_header_contentLength(), append_history, ast_debug, ast_dnsmgr_lookup(), ast_log(), ast_random(), ast_sched_add(), AST_SCHED_REPLACE_UNREF, ast_set_flag, ast_sip_ouraddrfor(), ast_string_field_set, ast_strlen_zero(), ast_verbose, sip_pvt::authname, sip_registry::authuser, sip_pvt::branch, build_callid_registry(), build_contact(), build_reply_digest(), build_via(), sip_registry::call, sip_registry::callback, sip_pvt::callid, sip_registry::callid, sip_registry::callid_valid, create_addr(), DEFAULT_MAX_FORWARDS, dialog_ref(), dialog_unlink_all(), dialog_unref(), sip_registry::dnsmgr, sip_pvt::do_history, sip_registry::domain, sip_registry::expiry, exten, sip_pvt::flags, sip_pvt::fromdomain, sip_pvt::fromuser, get_transport(), sip_request::headers, sip_registry::hostname, init_req(), initialize_initreq(), sip_pvt::initreq, sip_request::lines, ast_tcptls_session_args::local_address, LOG_NOTICE, LOG_WARNING, make_our_tag(), MAXHOSTNAMELEN, sip_registry::md5secret, sip_pvt::nonce, sip_registry::nonce, sip_registry::noncecount, sip_pvt::noncecount, obproxy_get(), sip_pvt::ocseq, sip_registry::ocseq, sip_registry::opaque, sip_pvt::opaque, sip_pvt::our_contact, sip_pvt::ourip, sip_pvt::peermd5secret, sip_pvt::peername, sip_pvt::peersecret, sip_socket::port, sip_registry::portno, sip_registry::qop, sip_pvt::qop, sip_registry::realm, sip_pvt::realm, sip_pvt::recv, ref_proxy(), REG_STATE_AUTHSENT, REG_STATE_REGSENT, sip_registry::regattempts, sip_pvt::registry, registry_addref(), registry_unref(), sip_registry::regstate, sip_pvt::sa, sched, sip_registry::secret, send_request(), sip_alloc(), sip_debug_test_pvt(), sip_methods, SIP_OUTGOING, sip_reg_timeout(), SIP_REGISTER, sip_tcp_desc, SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, sip_pvt::socket, STANDARD_SIP_PORT, sip_pvt::tag, cfsip_methods::text, sip_pvt::theirtag, sip_registry::timeout, sip_pvt::tohost, sip_registry::transport, TRUE, sip_socket::type, sip_pvt::uri, sip_registry::us, sip_pvt::username, sip_registry::username, sip_pvt::via, and XMIT_CRITICAL.

Referenced by __sip_do_register(), do_register_auth(), handle_response_register(), and sip_reg_timeout().

{
   struct sip_request req;
   char from[256];
   char to[256];
   char tmp[80];
   char addr[80];
   struct sip_pvt *p;
   int res;
   char *fromdomain;

   /* exit if we are already in process with this registrar ?*/
   if (r == NULL || ((auth == NULL) && (r->regstate == REG_STATE_REGSENT || r->regstate == REG_STATE_AUTHSENT))) {
      if (r) {
         ast_log(LOG_NOTICE, "Strange, trying to register %s@%s when registration already pending\n", r->username, r->hostname);
      }
      return 0;
   }

   if (r->dnsmgr == NULL) {
      char transport[MAXHOSTNAMELEN];
      snprintf(transport, sizeof(transport), "_sip._%s", get_transport(r->transport)); /* have to use static get_transport function */
      ast_dnsmgr_lookup(r->hostname, &r->us, &r->dnsmgr, global_srvlookup ? transport : NULL);
   }

   if (r->call) { /* We have a registration */
      if (!auth) {
         ast_log(LOG_WARNING, "Already have a REGISTER going on to %s@%s?? \n", r->username, r->hostname);
         return 0;
      } else {
         p = dialog_ref(r->call, "getting a copy of the r->call dialog in transmit_register");
         make_our_tag(p->tag, sizeof(p->tag));  /* create a new local tag for every register attempt */
         ast_string_field_set(p, theirtag, NULL);  /* forget their old tag, so we don't match tags when getting response */
      }
   } else {
      /* Build callid for registration if we haven't registered before */
      if (!r->callid_valid) {
         build_callid_registry(r, internip.sin_addr, default_fromdomain);
         r->callid_valid = TRUE;
      }
      /* Allocate SIP dialog for registration */
      if (!(p = sip_alloc( r->callid, NULL, 0, SIP_REGISTER))) {
         ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
         return 0;
      }
      
      if (p->do_history)
         append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);

      ref_proxy(p, obproxy_get(p, NULL));

      /* Use port number specified if no SRV record was found */
      if (!r->us.sin_port && r->portno)
         r->us.sin_port = htons(r->portno);

      /* Find address to hostname */
      if (create_addr(p, r->hostname, &r->us, 0)) {
         /* we have what we hope is a temporary network error,
          * probably DNS.  We need to reschedule a registration try */
         dialog_unlink_all(p, TRUE, TRUE);
         p = dialog_unref(p, "unref dialog after unlink_all");
         if (r->timeout > -1) {
            AST_SCHED_REPLACE_UNREF(r->timeout, sched, global_reg_timeout * 1000, sip_reg_timeout, r,
                              registry_unref(_data, "del for REPLACE of registry ptr"), 
                              registry_unref(r, "object ptr dec when SCHED_REPLACE add failed"),
                              registry_addref(r,"add for REPLACE registry ptr"));
            ast_log(LOG_WARNING, "Still have a registration timeout for %s@%s (create_addr() error), %d\n", r->username, r->hostname, r->timeout);
         } else {
            r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, registry_addref(r, "add for REPLACE registry ptr"));
            ast_log(LOG_WARNING, "Probably a DNS error for registration to %s@%s, trying REGISTER again (after %d seconds)\n", r->username, r->hostname, global_reg_timeout);
         }
         r->regattempts++;
         return 0;
      }

      /* Copy back Call-ID in case create_addr changed it */
      ast_string_field_set(r, callid, p->callid);
      if (!r->dnsmgr && r->portno) {
         p->sa.sin_port = htons(r->portno);
         p->recv.sin_port = htons(r->portno);
      } else { /* Set registry port to the port set from the peer definition/srv or default */
         r->portno = ntohs(p->sa.sin_port);
      }
      ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Registration is outgoing call */
      r->call = dialog_ref(p, "copying dialog into registry r->call");     /* Save pointer to SIP dialog */
      p->registry = registry_addref(r, "transmit_register: addref to p->registry in transmit_register"); /* Add pointer to registry in packet */
      if (!ast_strlen_zero(r->secret)) /* Secret (password) */
         ast_string_field_set(p, peersecret, r->secret);
      if (!ast_strlen_zero(r->md5secret))
         ast_string_field_set(p, peermd5secret, r->md5secret);
      /* User name in this realm  
      - if authuser is set, use that, otherwise use username */
      if (!ast_strlen_zero(r->authuser)) {   
         ast_string_field_set(p, peername, r->authuser);
         ast_string_field_set(p, authname, r->authuser);
      } else if (!ast_strlen_zero(r->username)) {
         ast_string_field_set(p, peername, r->username);
         ast_string_field_set(p, authname, r->username);
         ast_string_field_set(p, fromuser, r->username);
      }
      if (!ast_strlen_zero(r->username))
         ast_string_field_set(p, username, r->username);
      /* Save extension in packet */
      if (!ast_strlen_zero(r->callback))
         ast_string_field_set(p, exten, r->callback);

      /* Set transport and port so the correct contact is built */
      p->socket.type = r->transport;
      if (r->transport == SIP_TRANSPORT_TLS || r->transport == SIP_TRANSPORT_TCP) {
         p->socket.port = sip_tcp_desc.local_address.sin_port;
      }

      /*
        check which address we should use in our contact header 
        based on whether the remote host is on the external or
        internal network so we can register through nat
       */
      ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
      build_contact(p);
   }

   /* set up a timeout */
   if (auth == NULL)  {
      if (r->timeout > -1)
         ast_log(LOG_WARNING, "Still have a registration timeout, #%d - deleting it\n", r->timeout);
      AST_SCHED_REPLACE_UNREF(r->timeout, sched, global_reg_timeout * 1000, sip_reg_timeout, r,
                        registry_unref(_data,"reg ptr unrefed from del in SCHED_REPLACE"),
                        registry_unref(r,"reg ptr unrefed from add failure in SCHED_REPLACE"),
                        registry_addref(r,"reg ptr reffed from add in SCHED_REPLACE"));
      ast_debug(1, "Scheduled a registration timeout for %s id  #%d \n", r->hostname, r->timeout);
   }

   if ((fromdomain = strchr(r->username, '@'))) {
      /* the domain name is just behind '@' */
      fromdomain++ ;
      /* We have a domain in the username for registration */
      snprintf(from, sizeof(from), "<sip:%s>;tag=%s", r->username, p->tag);
      if (!ast_strlen_zero(p->theirtag))
         snprintf(to, sizeof(to), "<sip:%s>;tag=%s", r->username, p->theirtag);
      else
         snprintf(to, sizeof(to), "<sip:%s>", r->username);

      /* If the registration username contains '@', then the domain should be used as
         the equivalent of "fromdomain" for the registration */
      if (ast_strlen_zero(p->fromdomain)) {
         ast_string_field_set(p, fromdomain, fromdomain);
      }
   } else {
      snprintf(from, sizeof(from), "<sip:%s@%s>;tag=%s", r->username, p->tohost, p->tag);
      if (!ast_strlen_zero(p->theirtag))
         snprintf(to, sizeof(to), "<sip:%s@%s>;tag=%s", r->username, p->tohost, p->theirtag);
      else
         snprintf(to, sizeof(to), "<sip:%s@%s>", r->username, p->tohost);
   }
   
   /* Fromdomain is what we are registering to, regardless of actual
      host name from SRV */
   if (!ast_strlen_zero(p->fromdomain)) {
      if (r->portno && r->portno != STANDARD_SIP_PORT)
         snprintf(addr, sizeof(addr), "sip:%s:%d", p->fromdomain, r->portno);
      else
         snprintf(addr, sizeof(addr), "sip:%s", p->fromdomain);
   } else {
      if (r->portno && r->portno != STANDARD_SIP_PORT)
         snprintf(addr, sizeof(addr), "sip:%s:%d", r->hostname, r->portno);
      else
         snprintf(addr, sizeof(addr), "sip:%s", r->hostname);
   }
   ast_string_field_set(p, uri, addr);

   p->branch ^= ast_random();

   init_req(&req, sipmethod, addr);

   /* Add to CSEQ */
   snprintf(tmp, sizeof(tmp), "%u %s", ++r->ocseq, sip_methods[sipmethod].text);
   p->ocseq = r->ocseq;

   build_via(p);
   add_header(&req, "Via", p->via);
   add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
   add_header(&req, "From", from);
   add_header(&req, "To", to);
   add_header(&req, "Call-ID", p->callid);
   add_header(&req, "CSeq", tmp);
   if (!ast_strlen_zero(global_useragent))
      add_header(&req, "User-Agent", global_useragent);

   
   if (auth)   /* Add auth header */
      add_header(&req, authheader, auth);
   else if (!ast_strlen_zero(r->nonce)) {
      char digest[1024];

      /* We have auth data to reuse, build a digest header.
       * Note, this is not always useful because some parties do not
       * like nonces to be reused (for good reasons!) so they will
       * challenge us anyways.
       */
      if (sipdebug)
         ast_debug(1, "   >>> Re-using Auth data for %s@%s\n", r->username, r->hostname);
      ast_string_field_set(p, realm, r->realm);
      ast_string_field_set(p, nonce, r->nonce);
      ast_string_field_set(p, domain, r->domain);
      ast_string_field_set(p, opaque, r->opaque);
      ast_string_field_set(p, qop, r->qop);
      p->noncecount = ++r->noncecount;

      memset(digest, 0, sizeof(digest));
      if(!build_reply_digest(p, sipmethod, digest, sizeof(digest)))
         add_header(&req, "Authorization", digest);
      else
         ast_log(LOG_NOTICE, "No authorization available for authentication of registration to %s@%s\n", r->username, r->hostname);
   
   }

   snprintf(tmp, sizeof(tmp), "%d", r->expiry);
   add_header(&req, "Expires", tmp);
   add_header(&req, "Contact", p->our_contact);
   add_header_contentLength(&req, 0);

   initialize_initreq(p, &req);
   if (sip_debug_test_pvt(p)) {
      ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
   }
   r->regstate = auth ? REG_STATE_AUTHSENT : REG_STATE_REGSENT;
   r->regattempts++; /* Another attempt */
   ast_debug(4, "REGISTER attempt %d to %s@%s\n", r->regattempts, r->username, r->hostname);
   res = send_request(p, &req, XMIT_CRITICAL, p->ocseq);
   dialog_unref(p, "p is finished here at the end of transmit_register");
   return res;
}

static int transmit_reinvite_with_sdp	(	struct sip_pvt *	p,
		int	t38version,
		int	oldsdp
	)			[static]

Transmit reinvite with SDP.

Note:

A re-invite is basically a new INVITE with the same CALL-ID and TAG as the INVITE that opened the SIP dialogue We reinvite so that the audio stream (RTP) go directly between the SIP UAs. SIP Signalling stays with * in the path.

If t38version is TRUE, we send T38 SDP for re-invite from audio/video to T38 UDPTL transmission on the channel

If oldsdp is TRUE then the SDP version number is not incremented. This is needed for Session-Timers so we can send a re-invite to refresh the SIP session without modifying the media session.

Definition at line 9081 of file chan_sip.c.

References add_header(), add_sdp(), ALLOWED_METHODS, append_history, ast_set_flag, ast_test_flag, sip_pvt::do_history, FALSE, sip_pvt::flags, initialize_initreq(), sip_pvt::lastinvite, sip_pvt::ocseq, reqprep(), send_request(), SIP_INVITE, SIP_OUTGOING, SIP_REINVITE_UPDATE, SIP_UPDATE, SUPPORTED_EXTENSIONS, TRUE, and XMIT_CRITICAL.

Referenced by check_pendings(), handle_response_invite(), proc_session_timer(), sip_indicate(), sip_read(), sip_sendhtml(), sip_set_rtp_peer(), sip_set_udptl_peer(), and sip_write().

{
   struct sip_request req;
   
   reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ?  SIP_UPDATE : SIP_INVITE, 0, 1);

   add_header(&req, "Allow", ALLOWED_METHODS);
   add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
   if (sipdebug) {
      if (oldsdp == TRUE)
         add_header(&req, "X-asterisk-Info", "SIP re-invite (Session-Timers)");
      else
         add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
   }

   if (p->do_history)
      append_history(p, "ReInv", "Re-invite sent");
   if (t38version)
      add_sdp(&req, p, oldsdp, FALSE, TRUE);
   else
      add_sdp(&req, p, oldsdp, TRUE, FALSE);

   /* Use this as the basis */
   initialize_initreq(p, &req);
   p->lastinvite = p->ocseq;
   ast_set_flag(&p->flags[0], SIP_OUTGOING);       /* Change direction of this dialog */

   return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
}

static int transmit_request	(	struct sip_pvt *	p,
		int	sipmethod,
		int	inc,
		enum xmittype	reliable,
		int	newbranch
	)			[static]

Transmit generic SIP request returns XMIT_ERROR if transmit failed with a critical error (don't retry).

Definition at line 10314 of file chan_sip.c.

References add_header(), add_header_contentLength(), sip_pvt::answered_elsewhere, INV_CONFIRMED, sip_pvt::invitestate, sip_pvt::ocseq, reqprep(), send_request(), SIP_ACK, and SIP_CANCEL.

Referenced by check_pendings(), handle_response(), handle_response_invite(), and sip_hangup().

{
   struct sip_request resp;
   
   if (sipmethod == SIP_ACK)
      p->invitestate = INV_CONFIRMED;

   reqprep(&resp, p, sipmethod, seqno, newbranch);
   if (sipmethod == SIP_CANCEL && p->answered_elsewhere) 
      add_header(&resp, "Reason", "SIP;cause=200;text=\"Call completed elsewhere\"");

   add_header_contentLength(&resp, 0);
   return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
}

static int transmit_request_with_auth	(	struct sip_pvt *	p,
		int	sipmethod,
		int	seqno,
		enum xmittype	reliable,
		int	newbranch
	)			[static]

Transmit SIP request, auth added.

Definition at line 10345 of file chan_sip.c.

References add_header(), add_header_contentLength(), ast_cause2str(), ast_log(), ast_strlen_zero(), auth_headers(), sip_invite_param::auth_type, buf, build_reply_digest(), sip_pvt::callid, dummy(), sip_pvt::hangupcause, LOG_WARNING, sip_pvt::ocseq, sip_pvt::options, PROXY_AUTH, sip_pvt::realm, reqprep(), send_request(), and SIP_BYE.

Referenced by __sip_autodestruct(), check_pendings(), and sip_hangup().

{
   struct sip_request resp;
   
   reqprep(&resp, p, sipmethod, seqno, newbranch);
   if (!ast_strlen_zero(p->realm)) {
      char digest[1024];

      memset(digest, 0, sizeof(digest));
      if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) {
         char *dummy, *response;
         enum sip_auth_type code = p->options ? p->options->auth_type : PROXY_AUTH; /* XXX force 407 if unknown */
         auth_headers(code, &dummy, &response);
         add_header(&resp, response, digest);
      } else
         ast_log(LOG_WARNING, "No authentication available for call %s\n", p->callid);
   }
   /* If we are hanging up and know a cause for that, send it in clear text to make
      debugging easier. */
   if (sipmethod == SIP_BYE)  {
      char buf[10];

      add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->hangupcause));
      snprintf(buf, sizeof(buf), "%d", p->hangupcause);
      add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
   }

   add_header_contentLength(&resp, 0);
   return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);   
}

static int transmit_response	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req
	)			[static]

Transmit response, no retransmits.

Definition at line 8253 of file chan_sip.c.

References __transmit_response(), and XMIT_UNRELIABLE.

{
   return __transmit_response(p, msg, req, XMIT_UNRELIABLE);
}

static int transmit_response_reliable	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req
	)			[static]

Transmit response, Make sure you get an ACK This is only used for responses to INVITEs, where we need to make sure we get an ACK.

Definition at line 8289 of file chan_sip.c.

References __transmit_response(), sip_request::ignore, XMIT_CRITICAL, and XMIT_UNRELIABLE.

Referenced by handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_cancel(), handle_request_invite(), handle_request_subscribe(), sip_hangup(), sip_indicate(), sip_sipredirect(), and sip_t38_abort().

{
   return __transmit_response(p, msg, req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL);
}

static int transmit_response_using_temp	(	ast_string_field	callid,
		struct sockaddr_in *	sin,
		int	useglobal_nat,
		const int	intended_method,
		const struct sip_request *	req,
		const char *	msg
	)			[static]

Transmit response, no retransmits, using a temporary pvt structure.

Definition at line 8201 of file chan_sip.c.

References __transmit_response(), ast_copy_flags, ast_log(), ast_random(), ast_sip_ouraddrfor(), ast_string_field_init, ast_string_field_set, ast_test_flag, ast_threadstorage_get(), build_via(), copy_socket_data(), do_setnat(), sip_pvt::fromdomain, global_flags, INITIAL_CSEQ, LOG_ERROR, make_our_tag(), SIP_NAT, SIP_NAT_ROUTE, sip_request::socket, ts_temp_pvt, and XMIT_UNRELIABLE.

Referenced by find_call().

{
   struct sip_pvt *p = NULL;

   if (!(p = ast_threadstorage_get(&ts_temp_pvt, sizeof(*p)))) {
      ast_log(LOG_ERROR, "Failed to get temporary pvt\n");
      return -1;
   }

   /* XXX the structure may be dirty from previous usage.
    * Here we should state clearly how we should reinitialize it
    * before using it.
    * E.g. certainly the threadstorage should be left alone,
    * but other thihngs such as flags etc. maybe need cleanup ?
    */
    
   /* Initialize the bare minimum */
   p->method = intended_method;

   if (!sin)
      p->ourip = internip;
   else {
      p->sa = *sin;
      ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
   }

   p->branch = ast_random();
   make_our_tag(p->tag, sizeof(p->tag));
   p->ocseq = INITIAL_CSEQ;

   if (useglobal_nat && sin) {
      ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT);
      p->recv = *sin;
      do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
   }

   ast_string_field_set(p, fromdomain, default_fromdomain);
   build_via(p);
   ast_string_field_set(p, callid, callid);

   copy_socket_data(&p->socket, &req->socket);

   /* Use this temporary pvt structure to send the message */
   __transmit_response(p, msg, req, XMIT_UNRELIABLE);

   /* Free the string fields, but not the pool space */
   ast_string_field_init(p, 0);

   return 0;
}

static int transmit_response_with_allow	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		enum xmittype	reliable
	)			[static]

Append Accept header, content length before transmitting response.

Definition at line 8317 of file chan_sip.c.

References add_header(), add_header_contentLength(), respprep(), and send_response().

Referenced by handle_incoming(), and handle_request_options().

{
   struct sip_request resp;
   respprep(&resp, p, msg, req);
   add_header(&resp, "Accept", "application/sdp");
   add_header_contentLength(&resp, 0);
   return send_response(p, &resp, reliable, 0);
}

static int transmit_response_with_auth	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		const char *	rand,
		enum xmittype	reliable,
		const char *	header,
		int	stale
	)			[static]

Respond with authorization request.

Definition at line 8327 of file chan_sip.c.

References add_header(), add_header_contentLength(), append_history, ast_log(), get_header(), LOG_WARNING, sip_pvt::noncecount, respprep(), send_response(), and sip_pvt::username.

Referenced by check_auth(), and transmit_fake_auth_response().

{
   struct sip_request resp;
   char tmp[512];
   int seqno = 0;

   if (reliable && (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1)) {
      ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
      return -1;
   }
   /* Stale means that they sent us correct authentication, but 
      based it on an old challenge (nonce) */
   snprintf(tmp, sizeof(tmp), "Digest algorithm=MD5, realm=\"%s\", nonce=\"%s\"%s", global_realm, randdata, stale ? ", stale=true" : "");
   respprep(&resp, p, msg, req);
   add_header(&resp, header, tmp);
   add_header_contentLength(&resp, 0);
   append_history(p, "AuthChal", "Auth challenge sent for %s - nc %d", p->username, p->noncecount);
   return send_response(p, &resp, reliable, seqno);
}

static int transmit_response_with_date	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req
	)			[static]

Append date and content length before transmitting response.

Definition at line 8307 of file chan_sip.c.

References add_header_contentLength(), append_date(), respprep(), send_response(), and XMIT_UNRELIABLE.

Referenced by register_verify().

{
   struct sip_request resp;
   respprep(&resp, p, msg, req);
   append_date(&resp);
   add_header_contentLength(&resp, 0);
   return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

static int transmit_response_with_minse	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		int	minse_int
	)			[static]

Transmit 422 response with Min-SE header (Session-Timers).

Definition at line 8270 of file chan_sip.c.

References add_header(), add_header_contentLength(), append_date(), respprep(), send_response(), and XMIT_UNRELIABLE.

Referenced by handle_request_invite().

{
   struct sip_request resp;
   char minse_str[20];

   respprep(&resp, p, msg, req);
   append_date(&resp);

   snprintf(minse_str, sizeof(minse_str), "%d", minse_int);
   add_header(&resp, "Min-SE", minse_str);

   add_header_contentLength(&resp, 0);
   return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

static int transmit_response_with_sdp	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		enum xmittype	reliable,
		int	oldsdp
	)			[static]

Used for 200 OK and 183 early media.

Returns:

Will return XMIT_ERROR for network errors.

Definition at line 8998 of file chan_sip.c.

References add_sdp(), ast_debug, ast_log(), ast_rtp_codec_setpref(), ast_test_flag, sip_pvt::autoframing, sip_pvt::callid, FALSE, sip_pvt::flags, get_header(), LOG_ERROR, LOG_WARNING, sip_pvt::pendinginvite, sip_pvt::prefs, respprep(), sip_pvt::rtp, send_response(), SIP_OUTGOING, t38properties::state, sip_pvt::t38, T38_ENABLED, T38_PEER_DIRECT, TRUE, and try_suggested_sip_codec().

Referenced by handle_invite_replaces(), handle_request_invite(), sip_answer(), sip_indicate(), and sip_write().

{
   struct sip_request resp;
   int seqno;
   if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) {
      ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq"));
      return -1;
   }
   respprep(&resp, p, msg, req);
   if (p->rtp) {
      if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
         ast_debug(1, "Setting framing from config on incoming call\n");
         ast_rtp_codec_setpref(p->rtp, &p->prefs);
      }
      try_suggested_sip_codec(p);
      if (p->t38.state == T38_PEER_DIRECT || p->t38.state == T38_ENABLED) {
         add_sdp(&resp, p, oldsdp, TRUE, TRUE);
      } else {
         add_sdp(&resp, p, oldsdp, TRUE, FALSE);
      }
   } else 
      ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
   if (reliable && !p->pendinginvite)
      p->pendinginvite = seqno;     /* Buggy clients sends ACK on RINGING too */
   return send_response(p, &resp, reliable, seqno);
}

static int transmit_response_with_t38_sdp	(	struct sip_pvt *	p,
		char *	msg,
		struct sip_request *	req,
		int	retrans
	)			[static]

Used for 200 OK and 183 early media.

Definition at line 8951 of file chan_sip.c.

References add_sdp(), ast_log(), ast_udptl_offered_from_local(), sip_pvt::callid, get_header(), LOG_ERROR, LOG_WARNING, sip_pvt::pendinginvite, respprep(), send_response(), and sip_pvt::udptl.

Referenced by handle_request_invite(), and sip_indicate().

{
   struct sip_request resp;
   int seqno;
   
   if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) {
      ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq"));
      return -1;
   }
   respprep(&resp, p, msg, req);
   if (p->udptl) {
      ast_udptl_offered_from_local(p->udptl, 0);
      add_sdp(&resp, p, 0, 0, 1);
   } else 
      ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
   if (retrans && !p->pendinginvite)
      p->pendinginvite = seqno;     /* Buggy clients sends ACK on RINGING too */
   return send_response(p, &resp, retrans, seqno);
}

static int transmit_response_with_unsupported	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		const char *	unsupported
	)			[static]

Transmit response, no retransmits.

Definition at line 8259 of file chan_sip.c.

References add_header(), add_header_contentLength(), append_date(), respprep(), send_response(), and XMIT_UNRELIABLE.

Referenced by handle_request_invite().

{
   struct sip_request resp;
   respprep(&resp, p, msg, req);
   append_date(&resp);
   add_header(&resp, "Unsupported", unsupported);
   add_header_contentLength(&resp, 0);
   return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

static int transmit_state_notify	(	struct sip_pvt *	p,
		int	state,
		int	full,
		int	timeout
	)			[static]

Used in the SUBSCRIBE notification subsystem (RFC3265).

Definition at line 9513 of file chan_sip.c.

References add_header(), add_header_contentLength(), add_line(), ast_copy_string(), ast_device_state(), AST_DEVICE_UNAVAILABLE, AST_EXTENSION_BUSY, AST_EXTENSION_DEACTIVATED, AST_EXTENSION_INUSE, AST_EXTENSION_NOT_INUSE, AST_EXTENSION_ONHOLD, AST_EXTENSION_REMOVED, AST_EXTENSION_RINGING, AST_EXTENSION_UNAVAILABLE, ast_get_hint(), ast_log(), AST_MAX_EXTENSION, ast_str_alloca, ast_str_append(), sip_pvt::context, CPIM_PIDF_XML, DIALOG_INFO_XML, sip_pvt::dialogver, sip_pvt::expiry, sip_pvt::exten, find_subscription_type(), get_header(), get_in_brackets(), sip_pvt::initreq, LOG_WARNING, NONE, sip_pvt::ocseq, sip_pvt::pendinginvite, PIDF_XML, remove_uri_parameters(), reqprep(), send_request(), SIP_NOTIFY, ast_str::str, strsep(), sip_pvt::subscribed, ast_str::used, XMIT_RELIABLE, and XPIDF_XML.

Referenced by __sip_autodestruct(), cb_extensionstate(), and handle_request_subscribe().

{
   struct ast_str *tmp = ast_str_alloca(4000);
   char from[256], to[256];
   char *c, *mfrom, *mto;
   struct sip_request req;
   char hint[AST_MAX_EXTENSION];
   char *statestring = "terminated";
   const struct cfsubscription_types *subscriptiontype;
   enum state { NOTIFY_OPEN, NOTIFY_INUSE, NOTIFY_CLOSED } local_state = NOTIFY_OPEN;
   char *pidfstate = "--";
   char *pidfnote= "Ready";
   
   memset(from, 0, sizeof(from));
   memset(to, 0, sizeof(to));

   switch (state) {
   case (AST_EXTENSION_RINGING | AST_EXTENSION_INUSE):
      statestring = (global_notifyringing) ? "early" : "confirmed";
      local_state = NOTIFY_INUSE;
      pidfstate = "busy";
      pidfnote = "Ringing";
      break;
   case AST_EXTENSION_RINGING:
      statestring = "early";
      local_state = NOTIFY_INUSE;
      pidfstate = "busy";
      pidfnote = "Ringing";
      break;
   case AST_EXTENSION_INUSE:
      statestring = "confirmed";
      local_state = NOTIFY_INUSE;
      pidfstate = "busy";
      pidfnote = "On the phone";
      break;
   case AST_EXTENSION_BUSY:
      statestring = "confirmed";
      local_state = NOTIFY_CLOSED;
      pidfstate = "busy";
      pidfnote = "On the phone";
      break;
   case AST_EXTENSION_UNAVAILABLE:
      statestring = "terminated";
      local_state = NOTIFY_CLOSED;
      pidfstate = "away";
      pidfnote = "Unavailable";
      break;
   case AST_EXTENSION_ONHOLD:
      statestring = "confirmed";
      local_state = NOTIFY_CLOSED;
      pidfstate = "busy";
      pidfnote = "On hold";
      break;
   case AST_EXTENSION_NOT_INUSE:
   default:
      /* Default setting */
      break;
   }

   subscriptiontype = find_subscription_type(p->subscribed);
   
   /* Check which device/devices we are watching  and if they are registered */
   if (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, p->exten)) {
      char *hint2 = hint, *individual_hint = NULL;
      int hint_count = 0, unavailable_count = 0;

      while ((individual_hint = strsep(&hint2, "&"))) {
         hint_count++;

         if (ast_device_state(individual_hint) == AST_DEVICE_UNAVAILABLE)
            unavailable_count++;
      }

      /* If none of the hinted devices are registered, we will
       * override notification and show no availability.
       */
      if (hint_count > 0 && hint_count == unavailable_count) {
         local_state = NOTIFY_CLOSED;
         pidfstate = "away";
         pidfnote = "Not online";
      }
   }

   ast_copy_string(from, get_header(&p->initreq, "From"), sizeof(from));
   c = get_in_brackets(from);
   if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
      ast_log(LOG_WARNING, "Huh?  Not a SIP header (%s)?\n", c);
      return -1;
   }
   
   mfrom = remove_uri_parameters(c);

   ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to));
   c = get_in_brackets(to);
   if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
      ast_log(LOG_WARNING, "Huh?  Not a SIP header (%s)?\n", c);
      return -1;
   }
   mto = remove_uri_parameters(c);

   reqprep(&req, p, SIP_NOTIFY, 0, 1);

   
   add_header(&req, "Event", subscriptiontype->event);
   add_header(&req, "Content-Type", subscriptiontype->mediatype);
   switch(state) {
   case AST_EXTENSION_DEACTIVATED:
      if (timeout)
         add_header(&req, "Subscription-State", "terminated;reason=timeout");
      else {
         add_header(&req, "Subscription-State", "terminated;reason=probation");
         add_header(&req, "Retry-After", "60");
      }
      break;
   case AST_EXTENSION_REMOVED:
      add_header(&req, "Subscription-State", "terminated;reason=noresource");
      break;
   default:
      if (p->expiry)
         add_header(&req, "Subscription-State", "active");
      else  /* Expired */
         add_header(&req, "Subscription-State", "terminated;reason=timeout");
   }
   switch (p->subscribed) {
   case XPIDF_XML:
   case CPIM_PIDF_XML:
      ast_str_append(&tmp, 0,
         "<?xml version=\"1.0\"?>\n"
         "<!DOCTYPE presence PUBLIC \"-//IETF//DTD RFCxxxx XPIDF 1.0//EN\" \"xpidf.dtd\">\n"
         "<presence>\n");
      ast_str_append(&tmp, 0, "<presentity uri=\"%s;method=SUBSCRIBE\" />\n", mfrom);
      ast_str_append(&tmp, 0, "<atom id=\"%s\">\n", p->exten);
      ast_str_append(&tmp, 0, "<address uri=\"%s;user=ip\" priority=\"0.800000\">\n", mto);
      ast_str_append(&tmp, 0, "<status status=\"%s\" />\n", (local_state ==  NOTIFY_OPEN) ? "open" : (local_state == NOTIFY_INUSE) ? "inuse" : "closed");
      ast_str_append(&tmp, 0, "<msnsubstatus substatus=\"%s\" />\n", (local_state == NOTIFY_OPEN) ? "online" : (local_state == NOTIFY_INUSE) ? "onthephone" : "offline");
      ast_str_append(&tmp, 0, "</address>\n</atom>\n</presence>\n");
      break;
   case PIDF_XML: /* Eyebeam supports this format */
      ast_str_append(&tmp, 0,
         "<?xml version=\"1.0\" encoding=\"ISO-8859-1\"?>\n"
         "<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" \nxmlns:pp=\"urn:ietf:params:xml:ns:pidf:person\"\nxmlns:es=\"urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status\"\nxmlns:ep=\"urn:ietf:params:xml:ns:pidf:rpid:rpid-person\"\nentity=\"%s\">\n", mfrom);
      ast_str_append(&tmp, 0, "<pp:person><status>\n");
      if (pidfstate[0] != '-')
         ast_str_append(&tmp, 0, "<ep:activities><ep:%s/></ep:activities>\n", pidfstate);
      ast_str_append(&tmp, 0, "</status></pp:person>\n");
      ast_str_append(&tmp, 0, "<note>%s</note>\n", pidfnote); /* Note */
      ast_str_append(&tmp, 0, "<tuple id=\"%s\">\n", p->exten); /* Tuple start */
      ast_str_append(&tmp, 0, "<contact priority=\"1\">%s</contact>\n", mto);
      if (pidfstate[0] == 'b') /* Busy? Still open ... */
         ast_str_append(&tmp, 0, "<status><basic>open</basic></status>\n");
      else
         ast_str_append(&tmp, 0, "<status><basic>%s</basic></status>\n", (local_state != NOTIFY_CLOSED) ? "open" : "closed");
      ast_str_append(&tmp, 0, "</tuple>\n</presence>\n");
      break;
   case DIALOG_INFO_XML: /* SNOM subscribes in this format */
      ast_str_append(&tmp, 0, "<?xml version=\"1.0\"?>\n");
      ast_str_append(&tmp, 0, "<dialog-info xmlns=\"urn:ietf:params:xml:ns:dialog-info\" version=\"%d\" state=\"%s\" entity=\"%s\">\n", p->dialogver++, full ? "full":"partial", mto);
      if ((state & AST_EXTENSION_RINGING) && global_notifyringing)
         ast_str_append(&tmp, 0, "<dialog id=\"%s\" direction=\"recipient\">\n", p->exten);
      else
         ast_str_append(&tmp, 0, "<dialog id=\"%s\">\n", p->exten);
      ast_str_append(&tmp, 0, "<state>%s</state>\n", statestring);
      if (state == AST_EXTENSION_ONHOLD) {
         ast_str_append(&tmp, 0, "<local>\n<target uri=\"%s\">\n"
                                         "<param pname=\"+sip.rendering\" pvalue=\"no\"/>\n"
                                         "</target>\n</local>\n", mto);
      }
      ast_str_append(&tmp, 0, "</dialog>\n</dialog-info>\n");
      break;
   case NONE:
   default:
      break;
   }

   add_header_contentLength(&req, tmp->used);
   add_line(&req, tmp->str);

   p->pendinginvite = p->ocseq;  /* Remember that we have a pending NOTIFY in order not to confuse the NOTIFY subsystem */

   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static void try_suggested_sip_codec

(

struct sip_pvt *

p

)

[static]

Try setting codec suggested by the SIP_CODEC channel variable.

Definition at line 5309 of file chan_sip.c.

References ast_getformatbyname(), ast_log(), sip_pvt::capability, sip_pvt::jointcapability, LOG_NOTICE, sip_pvt::owner, and pbx_builtin_getvar_helper().

Referenced by sip_answer(), and transmit_response_with_sdp().

{
   int fmt;
   const char *codec;

   codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
   if (!codec) 
      return;

   fmt = ast_getformatbyname(codec);
   if (fmt) {
      ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec);
      if (p->jointcapability & fmt) {
         p->jointcapability &= fmt;
         p->capability &= fmt;
      } else
         ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
   } else
      ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
   return;  
}

static int unload_module

(

void

)

[static]

PBX unload module API.

Definition at line 23480 of file chan_sip.c.

References ao2_iterator_init(), ao2_t_iterator_next, ao2_t_ref, ast_channel_unregister(), ast_cli_unregister_multiple(), ast_context_destroy(), ast_context_find(), ast_custom_function_unregister(), ast_free, ast_free_ha(), AST_LIST_LOCK, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_LIST_UNLOCK, ast_manager_unregister(), ast_mutex_lock(), ast_mutex_unlock(), AST_PTHREADT_NULL, AST_PTHREADT_STOP, ast_rtp_proto_unregister(), ast_sched_dump(), ast_softhangup(), AST_SOFTHANGUP_APPUNLOAD, ast_tcptls_server_stop(), ast_udptl_proto_unregister(), ast_unload_realtime(), ast_unregister_application(), ASTOBJ_CONTAINER_DESTROY, ASTOBJ_CONTAINER_DESTROYALL, authl, ast_tls_config::cafile, ast_tls_config::capath, ast_tls_config::certfile, checksipdomain_function, ast_tls_config::cipher, clear_realm_authentication(), clear_sip_domains(), cli_sip, default_tls_cfg, dialog_unlink_all(), dialogs, localaddr, ast_tcptls_session_args::master, monlock, sip_pvt::owner, peers_by_ip, regl, sched, sched_context_destroy(), sip_header_function, sip_registry_destroy(), sip_rtp, sip_tcp_desc, sip_tech, sip_tls_desc, sip_udptl, sipchaninfo_function, sippeer_function, sipsock, sip_threadinfo::stop, thread, sip_threadinfo::threadid, and TRUE.

{
   struct sip_pvt *p;
   struct sip_threadinfo *th;
   struct ast_context *con;
   struct ao2_iterator i;

   ast_sched_dump(sched);
   
   /* First, take us out of the channel type list */
   ast_channel_unregister(&sip_tech);

   /* Unregister dial plan functions */
   ast_custom_function_unregister(&sipchaninfo_function);
   ast_custom_function_unregister(&sippeer_function);
   ast_custom_function_unregister(&sip_header_function);
   ast_custom_function_unregister(&checksipdomain_function);

   /* Unregister dial plan applications */
   ast_unregister_application(app_dtmfmode);
   ast_unregister_application(app_sipaddheader);

   /* Unregister CLI commands */
   ast_cli_unregister_multiple(cli_sip, sizeof(cli_sip) / sizeof(struct ast_cli_entry));

   /* Disconnect from the RTP subsystem */
   ast_rtp_proto_unregister(&sip_rtp);

   /* Disconnect from UDPTL */
   ast_udptl_proto_unregister(&sip_udptl);

   /* Unregister AMI actions */
   ast_manager_unregister("SIPpeers");
   ast_manager_unregister("SIPshowpeer");
   ast_manager_unregister("SIPqualifypeer");
   ast_manager_unregister("SIPshowregistry");
   ast_manager_unregister("SIPnotify");
   
   /* Kill TCP/TLS server threads */
   if (sip_tcp_desc.master)
      ast_tcptls_server_stop(&sip_tcp_desc);
   if (sip_tls_desc.master)
      ast_tcptls_server_stop(&sip_tls_desc);

   /* Kill all existing TCP/TLS threads */
   AST_LIST_LOCK(&threadl);
   AST_LIST_TRAVERSE_SAFE_BEGIN(&threadl, th, list) {
      pthread_t thread = th->threadid;
      th->stop = 1;
      AST_LIST_UNLOCK(&threadl);
      pthread_kill(thread, SIGURG);
      pthread_join(thread, NULL);
      AST_LIST_LOCK(&threadl);
   }
   AST_LIST_TRAVERSE_SAFE_END;
   AST_LIST_UNLOCK(&threadl);

   /* Hangup all dialogs if they have an owner */
   i = ao2_iterator_init(dialogs, 0);
   while ((p = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      if (p->owner)
         ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
      ao2_t_ref(p, -1, "toss dialog ptr from iterator_next");
   }

   ast_mutex_lock(&monlock);
   if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
      pthread_cancel(monitor_thread);
      pthread_kill(monitor_thread, SIGURG);
      pthread_join(monitor_thread, NULL);
   }
   monitor_thread = AST_PTHREADT_STOP;
   ast_mutex_unlock(&monlock);

   /* Destroy all the dialogs and free their memory */
   i = ao2_iterator_init(dialogs, 0);
   while ((p = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      dialog_unlink_all(p, TRUE, TRUE);
      ao2_t_ref(p, -1, "throw away iterator result"); 
   }

   /* Free memory for local network address mask */
   ast_free_ha(localaddr);

   clear_realm_authentication(authl);


   if (default_tls_cfg.certfile)
      ast_free(default_tls_cfg.certfile);
   if (default_tls_cfg.cipher)
      ast_free(default_tls_cfg.cipher);
   if (default_tls_cfg.cafile)
      ast_free(default_tls_cfg.cafile);
   if (default_tls_cfg.capath)
      ast_free(default_tls_cfg.capath);

   ASTOBJ_CONTAINER_DESTROYALL(&regl, sip_registry_destroy);
   ASTOBJ_CONTAINER_DESTROY(&regl);

   ao2_t_ref(peers, -1, "unref the peers table");
   ao2_t_ref(peers_by_ip, -1, "unref the peers_by_ip table");
   ao2_t_ref(dialogs, -1, "unref the dialogs table");

   clear_sip_domains();
   close(sipsock);
   sched_context_destroy(sched);
   con = ast_context_find(used_context);
   if (con)
      ast_context_destroy(con, "SIP");
   ast_unload_realtime("sipregs");
   ast_unload_realtime("sippeers");

   return 0;
}

static void* unref_peer	(	struct sip_peer *	peer,
		char *	tag
	)			[static]

helper functions to unreference various types of objects. By handling them this way, we don't have to declare the destructor on each call, which removes the chance of errors.

Definition at line 2491 of file chan_sip.c.

References ao2_t_ref.

Referenced by __sip_autodestruct(), __sip_destroy(), _sip_qualify_peer(), _sip_show_peer(), _sip_show_peers(), build_peer(), check_peer_ok(), complete_sip_peer(), complete_sip_registered_peer(), complete_sip_user(), create_addr(), expire_register(), function_sippeer(), handle_request_subscribe(), handle_response_peerpoke(), parse_register_contact(), reg_source_db(), register_verify(), sip_devicestate(), sip_do_debug_peer(), sip_peer_hold(), sip_poke_all_peers(), sip_poke_noanswer(), sip_poke_peer(), sip_poke_peer_s(), sip_prune_realtime(), sip_show_inuse(), sip_show_user(), sip_show_users(), sip_unregister(), st_get_mode(), st_get_refresher(), st_get_se(), and update_call_counter().

{
   ao2_t_ref(peer, -1, tag);
   return NULL;
}

static int update_call_counter	(	struct sip_pvt *	fup,
		int	event
	)			[static]

update_call_counter: Handle call_limit for SIP devices Setting a call-limit will cause calls above the limit not to be accepted.

Remember that for a type=friend, there's one limit for the user and another for the peer, not a combined call limit. This will cause unexpected behaviour in subscriptions, since a "friend" is *two* devices in Asterisk, not one.

Thought: For realtime, we should probably update storage with inuse counter...

Returns:

0 if call is ok (no call limit, below threshold) -1 on rejection of call

Definition at line 4804 of file chan_sip.c.

References ao2_lock(), ao2_unlock(), ast_clear_flag, ast_copy_string(), ast_debug, AST_DEVICE_UNKNOWN, ast_devstate_changed(), ast_log(), ast_set_flag, ast_strlen_zero(), ast_test_flag, sip_peer::call_limit, DEC_CALL_LIMIT, DEC_CALL_RINGING, FALSE, find_peer(), FINDALLDEVICES, sip_pvt::flags, INC_CALL_LIMIT, INC_CALL_RINGING, sip_peer::inRinging, sip_peer::inUse, inuse, LOG_ERROR, sip_peer::name, name, sip_pvt::outgoing_call, sip_pvt::peername, SIP_CALL_LIMIT, SIP_INC_COUNT, SIP_INC_RINGING, SIP_PAGE2_CALL_ONHOLD, sip_peer_hold(), sip_pvt_lock, sip_pvt_unlock, TRUE, unref_peer(), and sip_pvt::username.

Referenced by __sip_destroy(), handle_request_cancel(), handle_request_invite(), handle_response_invite(), sip_call(), and sip_hangup().

{
   char name[256];
   int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
   int outgoing = fup->outgoing_call;
   struct sip_peer *p = NULL;

   ast_debug(3, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");


   /* Test if we need to check call limits, in order to avoid 
      realtime lookups if we do not need it */
   if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT) && !ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD))
      return 0;

   ast_copy_string(name, fup->username, sizeof(name));

   /* Check the list of devices */
   if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, TRUE, FINDALLDEVICES, FALSE) ) ) { /* Try to find peer */
      inuse = &p->inUse;
      call_limit = &p->call_limit;
      inringing = &p->inRinging;
      ast_copy_string(name, fup->peername, sizeof(name));
   }
   if (!p) {
      ast_debug(2, "%s is not a local device, no call limit\n", name);
      return 0;
   }

   switch(event) {
   /* incoming and outgoing affects the inUse counter */
   case DEC_CALL_LIMIT:
      /* Decrement inuse count if applicable */
      if (inuse) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (*inuse > 0) {
            if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
               (*inuse)--;
               ast_clear_flag(&fup->flags[0], SIP_INC_COUNT);
            }
         } else {
            *inuse = 0;
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }

      /* Decrement ringing count if applicable */
      if (inringing) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (*inringing > 0) {
            if (ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
               (*inringing)--;
               ast_clear_flag(&fup->flags[0], SIP_INC_RINGING);
            }
         } else {
            *inringing = 0;
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }

      /* Decrement onhold count if applicable */
      sip_pvt_lock(fup);
      ao2_lock(p);
      if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && global_notifyhold) {
         ast_clear_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD);
         ao2_unlock(p);
         sip_pvt_unlock(fup);
         sip_peer_hold(fup, FALSE);
      } else {
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }
      if (sipdebug)
         ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
      break;

   case INC_CALL_RINGING:
   case INC_CALL_LIMIT:
      /* If call limit is active and we have reached the limit, reject the call */
      if (*call_limit > 0 ) {
         if (*inuse >= *call_limit) {
            ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
            unref_peer(p, "update_call_counter: unref peer p, call limit exceeded");
            return -1; 
         }
      }
      if (inringing && (event == INC_CALL_RINGING)) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (!ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
            (*inringing)++;
            ast_set_flag(&fup->flags[0], SIP_INC_RINGING);
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }
      if (inuse) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (!ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
            (*inuse)++;
            ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }
      if (sipdebug) {
         ast_debug(2, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", "peer", name, *inuse, *call_limit);
      }
      break;

   case DEC_CALL_RINGING:
      if (inringing) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
            if (*inringing > 0) {
               (*inringing)--;
            }
            ast_clear_flag(&fup->flags[0], SIP_INC_RINGING);
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }
      break;

   default:
      ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
   }

   if (p) {
      ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", p->name);
      unref_peer(p, "update_call_counter: unref_peer from call counter");
   } 
   return 0;
}

static void update_peer	(	struct sip_peer *	p,
		int	expire
	)			[static]

Update peer data in database (if used).

Definition at line 3903 of file chan_sip.c.

References sip_peer::addr, ast_test_flag, sip_peer::deprecated_username, sip_peer::flags, sip_peer::fullcontact, sip_peer::is_realtime, sip_peer::lastms, sip_peer::name, sip_settings::peer_rtupdate, realtime_update_peer(), sip_cfg, SIP_PAGE2_RTCACHEFRIENDS, sip_peer::useragent, and sip_peer::username.

Referenced by register_verify().

{
   int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
   if (sip_cfg.peer_rtupdate &&
       (p->is_realtime || rtcachefriends)) {
      realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, p->useragent, expire, p->deprecated_username, p->lastms);
   }
}

---

Variable Documentation

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_DEFAULT , .description = "Session Initiation Protocol (SIP)" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "a9c98e5d177805051735cb5b0b16b0a0" , .load = load_module, .unload = unload_module, .reload = reload, } [static]

Definition at line 23599 of file chan_sip.c.

int allow_external_domains [static]

Accept calls to external SIP domains?

Definition at line 790 of file chan_sip.c.

int apeerobjs = 0 [static]

Autocreated peer objects

Definition at line 823 of file chan_sip.c.

char* app_dtmfmode = "SIPDtmfMode" [static]

Definition at line 23065 of file chan_sip.c.

char* app_sipaddheader = "SIPAddHeader" [static]

Definition at line 23067 of file chan_sip.c.

struct ast_module_info* ast_module_info = &__mod_info [static]

Definition at line 23599 of file chan_sip.c.

struct sip_auth* authl = NULL [static]

Authentication list for realm authentication.

Todo:

Move the sip_auth list to AST_LIST

Definition at line 1772 of file chan_sip.c.

Referenced by build_reply_digest(), reload_config(), sip_show_settings(), and unload_module().

int autocreatepeer [static]

Auto creation of peers at registration? Default off.

Definition at line 760 of file chan_sip.c.

struct sockaddr_in bindaddr [static]

UDP: The address we bind to

Definition at line 1793 of file chan_sip.c.

struct ast_threadstorage check_auth_buf = { .once = PTHREAD_ONCE_INIT, .key_init = __init_check_auth_buf , .custom_init = NULL , } [static]

Definition at line 10882 of file chan_sip.c.

Referenced by check_auth(), and transmit_fake_auth_response().

struct ast_custom_function checksipdomain_function [static]

Definition at line 15517 of file chan_sip.c.

Referenced by load_module(), and unload_module().

struct ast_cli_entry cli_sip[] [static]

SIP Cli commands definition.

Definition at line 23360 of file chan_sip.c.

Referenced by load_module(), and unload_module().

struct ast_cli_entry cli_sip_do_history_deprecated = { .handler = sip_do_history_deprecated , .summary = "Enable/Disable SIP history" ,__VA_ARGS__ } [static]

Definition at line 23358 of file chan_sip.c.

int compactheaders [static]

send compact sip headers

Definition at line 782 of file chan_sip.c.

const char config[] = "sip.conf" [static]

Main configuration file

Definition at line 287 of file chan_sip.c.

struct sockaddr_in debugaddr [static]

Definition at line 1838 of file chan_sip.c.

char default_callerid[AST_MAX_EXTENSION] [static]

Definition at line 726 of file chan_sip.c.

char default_context[AST_MAX_CONTEXT] [static]

Definition at line 723 of file chan_sip.c.

int default_expiry = DEFAULT_DEFAULT_EXPIRY [static]

Definition at line 244 of file chan_sip.c.

char default_fromdomain[AST_MAX_EXTENSION] [static]

Definition at line 727 of file chan_sip.c.

struct ast_jb_conf default_jbconf [static]

Global jitterbuffer configuration - by default, jb is disabled.

Definition at line 278 of file chan_sip.c.

char default_language[MAX_LANGUAGE] [static]

Definition at line 725 of file chan_sip.c.

int default_maxcallbitrate [static]

Maximum bitrate for call

Definition at line 735 of file chan_sip.c.

char default_mohinterpret[MAX_MUSICCLASS] [static]

Global setting for moh class to use when put on hold

Definition at line 731 of file chan_sip.c.

char default_mohsuggest[MAX_MUSICCLASS] [static]

Global setting for moh class to suggest when putting a bridged channel on hold

Definition at line 732 of file chan_sip.c.

char default_notifymime[AST_MAX_EXTENSION] [static]

Definition at line 728 of file chan_sip.c.

char default_parkinglot[AST_MAX_CONTEXT] [static]

Parkinglot

Definition at line 734 of file chan_sip.c.

struct ast_codec_pref default_prefs [static]

Default codec prefs

Definition at line 736 of file chan_sip.c.

Referenced by build_device(), reload_config(), set_peer_defaults(), sip_alloc(), sip_show_settings(), and temp_peer().

int default_qualify [static]

Default Qualify= setting

Definition at line 729 of file chan_sip.c.

char default_subscribecontext[AST_MAX_CONTEXT] [static]

Definition at line 724 of file chan_sip.c.

struct ast_tls_config default_tls_cfg [static]

Default TLS connection configuration.

Definition at line 2294 of file chan_sip.c.

Referenced by reload_config(), sip_prepare_socket(), sip_show_settings(), and unload_module().

char default_vmexten[AST_MAX_EXTENSION] [static]

Definition at line 730 of file chan_sip.c.

char* descrip_dtmfmode = " SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call\n" [static]

Definition at line 23064 of file chan_sip.c.

char* descrip_sipaddheader [static]

Definition at line 23070 of file chan_sip.c.

struct ao2_container* dialogs

Here we implement the container for dialogs (sip_pvt), defining generic wrapper functions to ease the transition from the current implementation (a single linked list) to a different container. In addition to a reference to the container, we need functions to lock/unlock the container and individual items, and functions to add/remove references to the individual items.

Definition at line 1419 of file chan_sip.c.

Referenced by complete_sipch(), create_addr_from_peer(), dialog_needdestroy(), dialog_unlink_all(), do_monitor(), find_call(), get_sip_pvt_byid_locked(), handle_request_subscribe(), load_module(), manager_sipnotify(), sip_alloc(), sip_cli_notify(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), sip_show_channel(), sip_show_channels(), sip_show_channelstats(), sip_show_history(), sip_show_objects(), and unload_module().

struct _map_x_s dtmfstr[] [static]

mapping between dtmf flags and strings

Definition at line 13171 of file chan_sip.c.

Referenced by conf_run(), dtmfmode2str(), handle_link_data(), handle_remote_data(), send_dtmf(), and str2dtmfmode().

int dumphistory [static]

Dump history to verbose before destroying SIP dialog

Definition at line 784 of file chan_sip.c.

time_t externexpire [static]

Expiration counter for re-resolving external host name in dynamic DNS

Definition at line 1824 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor().

char externhost[MAXHOSTNAMELEN] [static]

External host name

Definition at line 1823 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor(), and sip_show_settings().

struct sockaddr_in externip [static]

our external IP address/port for SIP sessions. externip.sin_addr is only set when we know we might be behind a NAT, and this is done using a variety of (mutually exclusive) ways from the config file:

+ with "externip = host[:port]" we specify the address/port explicitly. The address is looked up only once when (re)loading the config file;

+ with "externhost = host[:port]" we do a similar thing, but the hostname is stored in externhost, and the hostname->IP mapping is refreshed every 'externrefresh' seconds;

+ with "stunaddr = host[:port]" we run queries every externrefresh seconds to the specified server, and store the result in externip.

Other variables (externhost, externexpire, externrefresh) are used to support the above functions. External IP address if we are behind NAT

Definition at line 1821 of file chan_sip.c.

int externrefresh = 10 [static]

Definition at line 1825 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor(), and sip_show_settings().

int global_allowguest [static]

allow unauthenticated peers to connect?

Definition at line 768 of file chan_sip.c.

int global_allowsubscribe [static]

Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE the global setting is in globals_flags[1]

Definition at line 772 of file chan_sip.c.

enum transfermodes global_allowtransfer [static]

SIP Refer restriction scheme

Definition at line 798 of file chan_sip.c.

int global_alwaysauthreject [static]

Send 401 Unauthorized for all failing requests

Definition at line 757 of file chan_sip.c.

int global_authfailureevents [static]

Whether we send authentication failure manager events or not. Default no.

Definition at line 792 of file chan_sip.c.

int global_autoframing [static]

Turn autoframing on or off.

Definition at line 797 of file chan_sip.c.

int global_callcounter [static]

Enable call counters for all devices. This is currently enabled by setting the peer call-limit to 999. When we remove the call-limit from the code, we can make it with just a boolean flag in the device structure

Definition at line 769 of file chan_sip.c.

int global_callevents [static]

Whether we send manager events or not

Definition at line 791 of file chan_sip.c.

int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263 [static]

Codecs that we support by default:.

Definition at line 805 of file chan_sip.c.

struct ast_ha* global_contact_ha = NULL [static]

Global list of addresses dynamic peers are not allowed to use.

Definition at line 815 of file chan_sip.c.

Referenced by build_peer(), parse_register_contact(), and reload_config().

unsigned int global_cos_audio [static]

802.1p class of service for audio RTP packets

Definition at line 779 of file chan_sip.c.

unsigned int global_cos_sip [static]

802.1p class of service for SIP packets

Definition at line 778 of file chan_sip.c.

unsigned int global_cos_text [static]

802.1p class of service for text RTP packets

Definition at line 781 of file chan_sip.c.

unsigned int global_cos_video [static]

802.1p class of service for video RTP packets

Definition at line 780 of file chan_sip.c.

int global_directrtpsetup [static]

Enable support for Direct RTP setup (no re-invites)

Definition at line 753 of file chan_sip.c.

int global_dynamic_exclude_static = 0 [static]

Definition at line 816 of file chan_sip.c.

struct ast_flags global_flags[2] = {{0}} [static]

global SIP_ flags

Definition at line 827 of file chan_sip.c.

Referenced by build_peer(), build_radius_record(), get_destination(), load_module(), load_moh_classes(), local_ast_moh_start(), set_peer_defaults(), sip_alloc(), sip_show_settings(), and transmit_response_using_temp().

struct ast_jb_conf global_jbconf [static]

Global jitterbuffer configuration

Definition at line 285 of file chan_sip.c.

int global_match_auth_username [static]

Match auth username if available instead of From: Default off.

Definition at line 761 of file chan_sip.c.

int global_matchexterniplocally [static]

Match externip/externhost setting against localnet setting

Definition at line 800 of file chan_sip.c.

int global_max_se [static]

Highest threshold for session refresh interval

Definition at line 810 of file chan_sip.c.

int global_min_se [static]

Lowest threshold for session refresh interval

Definition at line 809 of file chan_sip.c.

int global_notifyhold [static]

Send notifications on hold

Definition at line 756 of file chan_sip.c.

int global_notifyringing [static]

Send notifications on ringing

Definition at line 755 of file chan_sip.c.

struct sip_proxy global_outboundproxy [static]

Outbound proxy

Definition at line 799 of file chan_sip.c.

Referenced by obproxy_get(), ref_proxy(), and sip_show_settings().

int global_qualifyfreq [static]

Qualify frequency

Definition at line 801 of file chan_sip.c.

char global_realm[MAXHOSTNAMELEN] [static]

Default realm

Definition at line 785 of file chan_sip.c.

int global_reg_timeout [static]

Definition at line 766 of file chan_sip.c.

int global_regattempts_max [static]

Registration attempts before giving up

Definition at line 767 of file chan_sip.c.

char global_regcontext[AST_MAX_CONTEXT] [static]

Context for auto-extensions

Definition at line 786 of file chan_sip.c.

int global_regextenonqualify [static]

Whether to add/remove regexten when qualifying peers

Definition at line 796 of file chan_sip.c.

int global_relaxdtmf [static]

Relax DTMF

Definition at line 762 of file chan_sip.c.

int global_rtautoclear [static]

Realtime ??

Definition at line 754 of file chan_sip.c.

int global_rtpholdtimeout [static]

Time out call if no RTP during hold

Definition at line 764 of file chan_sip.c.

int global_rtpkeepalive [static]

Send RTP keepalives

Definition at line 765 of file chan_sip.c.

int global_rtptimeout [static]

Time out call if no RTP

Definition at line 763 of file chan_sip.c.

char global_sdpowner[AST_MAX_EXTENSION] [static]

SDP owner name for the SIP channel

Definition at line 789 of file chan_sip.c.

char global_sdpsession[AST_MAX_EXTENSION] [static]

SDP session name for the SIP channel

Definition at line 788 of file chan_sip.c.

int global_srvlookup [static]

SRV Lookup on or off. Default is on

Definition at line 758 of file chan_sip.c.

enum st_mode global_st_mode [static]

Mode of operation for Session-Timers

Definition at line 807 of file chan_sip.c.

enum st_refresher global_st_refresher [static]

Session-Timer refresher

Definition at line 808 of file chan_sip.c.

int global_t1 [static]

T1 time

Definition at line 793 of file chan_sip.c.

int global_t1min [static]

T1 roundtrip time minimum

Definition at line 794 of file chan_sip.c.

int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600 [static]

Definition at line 1120 of file chan_sip.c.

int global_timer_b [static]

Timer B - RFC 3261 Section 17.1.1.2

Definition at line 795 of file chan_sip.c.

unsigned int global_tos_audio [static]

IP type of service for audio RTP packets

Definition at line 775 of file chan_sip.c.

unsigned int global_tos_sip [static]

IP type of service for SIP packets

Definition at line 774 of file chan_sip.c.

unsigned int global_tos_text [static]

IP type of service for text RTP packets

Definition at line 777 of file chan_sip.c.

unsigned int global_tos_video [static]

IP type of service for video RTP packets

Definition at line 776 of file chan_sip.c.

char global_useragent[AST_MAX_EXTENSION] [static]

Useragent for the SIP channel

Definition at line 787 of file chan_sip.c.

int hash_dialog_size = 563 [static]

Definition at line 1660 of file chan_sip.c.

Referenced by load_module().

int hash_peer_size = 563 [static]

Size of peer hash table, prime number preferred!

Definition at line 1659 of file chan_sip.c.

Referenced by load_module().

int hash_user_size = 563 [static]

Definition at line 1661 of file chan_sip.c.

struct _map_x_s insecurestr[] [static]

Definition at line 13192 of file chan_sip.c.

Referenced by insecure2str().

struct sockaddr_in internip [static]

our (internal) default address/port to put in SIP/SDP messages internip is initialized picking a suitable address from one of the interfaces, and the same port number we bind to. It is used as the default address/port in SIP messages, and as the default address (but not port) in SDP messages.

Definition at line 1801 of file chan_sip.c.

struct invstate2stringtable invitestate2string[] [static]

Readable descriptions of device states.

Note:

Should be aligned to above table as index

Referenced by show_chanstats_cb().

struct io_context* io [static]

The IO context

Definition at line 847 of file chan_sip.c.

struct ast_ha* localaddr [static]

List of local networks We store "localnet" addresses from the config file into an access list, marked as 'DENY', so the call to ast_apply_ha() will return AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local' (i.e. presumably public) addresses.

List of local networks, on the same side of NAT as this Asterisk

Definition at line 1834 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor(), sip_show_settings(), and unload_module().

char mandescr_show_peer[] [static]

Initial value:

 
"Description: Show one SIP peer with details on current status.\n"
"Variables: \n"
"  Peer: <name>           The peer name you want to check.\n"
"  ActionID: <id>   Optional action ID for this AMI transaction.\n"

Definition at line 13503 of file chan_sip.c.

Referenced by load_module().

char mandescr_show_peers[] [static]

Definition at line 12882 of file chan_sip.c.

Referenced by load_module().

char mandescr_show_registry[] [static]

Manager Action SIPShowRegistry description.

Definition at line 12837 of file chan_sip.c.

Referenced by load_module().

char mandescr_sipnotify[] [static]

Definition at line 9836 of file chan_sip.c.

Referenced by load_module().

int max_expiry = DEFAULT_MAX_EXPIRY [static]

Maximum accepted registration time

Definition at line 243 of file chan_sip.c.

int min_expiry = DEFAULT_MIN_EXPIRY [static]

Minimum accepted registration time

Definition at line 242 of file chan_sip.c.

pthread_t monitor_thread = AST_PTHREADT_NULL [static]

This is the thread for the monitor which checks for input on the channels which are not currently in use.

Definition at line 841 of file chan_sip.c.

ast_mutex_t monlock = ((ast_mutex_t) PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP ) [static]

Protect the monitoring thread, so only one process can kill or start it, and not when it's doing something critical.

Definition at line 835 of file chan_sip.c.

struct _map_x_s natmodes[] [static]

Definition at line 12620 of file chan_sip.c.

Referenced by nat2str().

ast_mutex_t netlock = ((ast_mutex_t) PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP ) [static]

Definition at line 831 of file chan_sip.c.

const char notify_config[] = "sip_notify.conf" [static]

Configuration file for sending Notify with CLI commands to reconfigure or reboot phones

Definition at line 288 of file chan_sip.c.

struct ast_config* notify_types [static]

The list of manual NOTIFY types we know how to send

Definition at line 1840 of file chan_sip.c.

Referenced by complete_sipnotify(), and sip_cli_notify().

int ourport_tcp [static]

The port used for TCP connections

Definition at line 1836 of file chan_sip.c.

Referenced by _sip_tcp_helper_thread().

int ourport_tls [static]

The port used for TCP/TLS connections

Definition at line 1837 of file chan_sip.c.

Referenced by _sip_tcp_helper_thread().

int pedanticsipchecking [static]

Extra checking ? Default off

Definition at line 759 of file chan_sip.c.

struct ao2_container* peers

The peer list: Users, Peers and Friends.

Definition at line 1668 of file chan_sip.c.

struct ao2_container* peers_by_ip

Definition at line 1669 of file chan_sip.c.

Referenced by expire_register(), find_peer(), load_module(), parse_register_contact(), register_verify(), sip_prune_realtime(), and unload_module().

int recordhistory [static]

Record SIP history. Off by default

Definition at line 783 of file chan_sip.c.

struct _map_x_s referstatusstrings[] [static]

Definition at line 1185 of file chan_sip.c.

Referenced by referstatus2str().

struct ast_register_list regl [static]

The register list: Other SIP proxies we register with and place calls to.

Referenced by load_module(), manager_show_registry(), reload_config(), sip_register(), sip_send_all_registers(), sip_show_objects(), sip_show_registry(), and unload_module().

int regobjs = 0 [static]

Registry objects

Definition at line 824 of file chan_sip.c.

struct _map_x_s regstatestrings[] [static]

Definition at line 9845 of file chan_sip.c.

Referenced by regstate2str().

int rpeerobjs = 0 [static]

Realtime peers

Definition at line 822 of file chan_sip.c.

struct sched_context* sched [static]

The scheduling context

Definition at line 846 of file chan_sip.c.

struct sip_settings sip_cfg [static]

Definition at line 745 of file chan_sip.c.

Referenced by build_peer(), destroy_association(), handle_response_peerpoke(), realtime_update_peer(), sip_poke_noanswer(), sip_show_settings(), and update_peer().

struct ast_custom_function sip_header_function [static]

Definition at line 15493 of file chan_sip.c.

Referenced by load_module(), and unload_module().

struct cfsip_methods sip_methods[] [static]

The core structure to setup dialogs. We parse incoming messages by using structure and then route the messages according to the type.

Note:

Note that sip_methods[i].id == i must hold or the code breaks

Referenced by __sip_autodestruct(), __sip_destroy(), __sip_pretend_ack(), __sip_semi_ack(), build_reply_digest(), check_auth(), dialog_needdestroy(), do_proxy_auth(), find_call(), find_call_cb(), find_sip_method(), get_destination(), handle_incoming(), handle_request_bye(), handle_request_invite(), handle_request_register(), handle_request_subscribe(), handle_response(), init_req(), initialize_initreq(), initreqprep(), method_match(), reqprep(), retrans_pkt(), send_request(), send_response(), sip_alloc(), sip_scheddestroy(), and transmit_register().

struct cfsip_options sip_options[] [static]

List of well-known SIP options. If we get this in a require, we should check the list and answer accordingly.

Referenced by _sip_show_peer(), parse_sip_options(), and sip_show_channel().

ast_mutex_t sip_reload_lock = ((ast_mutex_t) PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP ) [static]

Definition at line 837 of file chan_sip.c.

Referenced by do_monitor(), and sip_reload().

int sip_reloading = FALSE [static]

Flag for avoiding multiple reloads at the same time

Definition at line 843 of file chan_sip.c.

enum channelreloadreason sip_reloadreason [static]

Reason for last reload/load of configuration

Definition at line 844 of file chan_sip.c.

struct ast_rtp_protocol sip_rtp [static]

Interface structure with callbacks used to connect to RTP module.

Definition at line 2349 of file chan_sip.c.

Referenced by load_module(), and unload_module().

struct ast_tcptls_session_args sip_tcp_desc [static]

The TCP server definition.

Definition at line 2297 of file chan_sip.c.

Referenced by reload_config(), sip_show_settings(), transmit_register(), and unload_module().

struct ast_channel_tech sip_tech [static]

Definition of this channel for PBX channel registration.

Definition at line 2253 of file chan_sip.c.

Referenced by load_module(), sip_new(), and unload_module().

struct ast_channel_tech sip_tech_info [static]

This version of the sip channel tech has no send_digit_begin callback so that the core knows that the channel does not want DTMF BEGIN frames. The struct is initialized just before registering the channel driver, and is for use with channels using SIP INFO DTMF.

Definition at line 2287 of file chan_sip.c.

Referenced by load_module(), and sip_new().

struct ast_tls_config sip_tls_cfg [static]

Working TLS connection configuration.

Definition at line 2291 of file chan_sip.c.

struct ast_tcptls_session_args sip_tls_desc [static]

The TCP/TLS server definition.

Definition at line 2308 of file chan_sip.c.

Referenced by reload_config(), sip_show_settings(), and unload_module().

struct ast_udptl_protocol sip_udptl [static]

Initial value:

 {
   type: "SIP",
   get_udptl_info: sip_get_udptl_peer,
   set_udptl_peer: sip_set_udptl_peer,
}

Interface structure with callbacks used to connect to UDPTL module.

Definition at line 2610 of file chan_sip.c.

Referenced by load_module(), and unload_module().

struct ast_custom_function sipchaninfo_function [static]

Structure to declare a dialplan function: SIPCHANINFO.

Definition at line 15707 of file chan_sip.c.

Referenced by load_module(), and unload_module().

enum sip_debug_e sipdebug [static]

Definition at line 1135 of file chan_sip.c.

int sipdebug_text [static]

extra debugging for 'text' related events. At the moment this is set together with sip_debug_console.

Note:

It should either go away or be implemented properly.

Definition at line 1141 of file chan_sip.c.

struct ast_custom_function sippeer_function

Structure to declare a dialplan function: SIPPEER.

Definition at line 15616 of file chan_sip.c.

Referenced by load_module(), and unload_module().

int sipsock = -1 [static]

Main socket for SIP communication.

sipsock is shared between the SIP manager thread (which handles reload requests), the io handler (sipsock_read()) and the user routines that issue writes (using __sip_xmit()). The socket is -1 only when opening fails (this is a permanent condition), or when we are handling a reload() that changes its address (this is a transient situation during which we might have a harmless race, see below). Because the conditions for the race to be possible are extremely rare, we don't want to pay the cost of locking on every I/O. Rather, we remember that when the race may occur, communication is bound to fail anyways, so we just live with this event and let the protocol handle this above us.

Definition at line 1791 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor(), do_monitor(), reg_source_db(), sip_prepare_socket(), sipsock_read(), and unload_module().

int* sipsock_read_id [static]

ID of IO entry for sipsock FD

Definition at line 848 of file chan_sip.c.

int speerobjs = 0 [static]

Static peers

Definition at line 821 of file chan_sip.c.

struct _map_x_s stmodes[] [static]

Report Peer status in character string.

Returns:

0 if peer is unreachable, 1 if peer is online, -1 if unmonitored

Definition at line 12660 of file chan_sip.c.

Referenced by stmode2str(), and str2stmode().

struct _map_x_s strefreshers[] [static]

Definition at line 12678 of file chan_sip.c.

Referenced by str2strefresher(), and strefresher2str().

struct sockaddr_in stunaddr [static]

stun server address

Definition at line 1826 of file chan_sip.c.

struct cfsubscription_types subscription_types[] [static]

Subscription types that we support. We support

• dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
• SIMPLE presence used for device status
• Voicemail notification subscriptions.

Referenced by find_subscription_type(), and subscription_type2str().

char* synopsis_dtmfmode = "Change the dtmfmode for a SIP call" [static]

Definition at line 23063 of file chan_sip.c.

char* synopsis_sipaddheader = "Add a SIP header to the outbound call" [static]

Definition at line 23068 of file chan_sip.c.

struct ast_threadstorage ts_temp_pvt = { .once = PTHREAD_ONCE_INIT, .key_init = __init_ts_temp_pvt , .custom_init = temp_pvt_init , } [static]

Definition at line 1760 of file chan_sip.c.

Referenced by transmit_response_using_temp().

char used_context[AST_MAX_CONTEXT] [static]

name of automatically created context for unloading

Definition at line 828 of file chan_sip.c.

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