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Asterisk developer's documentation


app_dial.c

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00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 1999 - 2006, Digium, Inc.
00005  *
00006  * Mark Spencer <markster@digium.com>
00007  *
00008  * See http://www.asterisk.org for more information about
00009  * the Asterisk project. Please do not directly contact
00010  * any of the maintainers of this project for assistance;
00011  * the project provides a web site, mailing lists and IRC
00012  * channels for your use.
00013  *
00014  * This program is free software, distributed under the terms of
00015  * the GNU General Public License Version 2. See the LICENSE file
00016  * at the top of the source tree.
00017  */
00018 
00019 /*! \file
00020  *
00021  * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
00022  *
00023  * \author Mark Spencer <markster@digium.com>
00024  *
00025  * \ingroup applications
00026  */
00027 
00028 /*** MODULEINFO
00029         <depend>chan_local</depend>
00030  ***/
00031 
00032 
00033 #include "asterisk.h"
00034 
00035 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 198292 $")
00036 
00037 #include <sys/time.h>
00038 #include <sys/signal.h>
00039 #include <sys/stat.h>
00040 #include <netinet/in.h>
00041 
00042 #include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
00043 #include "asterisk/lock.h"
00044 #include "asterisk/file.h"
00045 #include "asterisk/channel.h"
00046 #include "asterisk/pbx.h"
00047 #include "asterisk/module.h"
00048 #include "asterisk/translate.h"
00049 #include "asterisk/say.h"
00050 #include "asterisk/config.h"
00051 #include "asterisk/features.h"
00052 #include "asterisk/musiconhold.h"
00053 #include "asterisk/callerid.h"
00054 #include "asterisk/utils.h"
00055 #include "asterisk/app.h"
00056 #include "asterisk/causes.h"
00057 #include "asterisk/rtp.h"
00058 #include "asterisk/cdr.h"
00059 #include "asterisk/manager.h"
00060 #include "asterisk/privacy.h"
00061 #include "asterisk/stringfields.h"
00062 #include "asterisk/global_datastores.h"
00063 
00064 static char *app = "Dial";
00065 
00066 static char *synopsis = "Place a call and connect to the current channel";
00067 
00068 static char *descrip =
00069 "  Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]):\n"
00070 "This application will place calls to one or more specified channels. As soon\n"
00071 "as one of the requested channels answers, the originating channel will be\n"
00072 "answered, if it has not already been answered. These two channels will then\n"
00073 "be active in a bridged call. All other channels that were requested will then\n"
00074 "be hung up.\n"
00075 "  Unless there is a timeout specified, the Dial application will wait\n"
00076 "indefinitely until one of the called channels answers, the user hangs up, or\n"
00077 "if all of the called channels are busy or unavailable. Dialplan executing will\n"
00078 "continue if no requested channels can be called, or if the timeout expires.\n\n"
00079 "  This application sets the following channel variables upon completion:\n"
00080 "    DIALEDTIME   - This is the time from dialing a channel until when it\n"
00081 "                   is disconnected.\n"
00082 "    ANSWEREDTIME - This is the amount of time for actual call.\n"
00083 "    DIALSTATUS   - This is the status of the call:\n"
00084 "                   CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n"
00085 "                   DONTCALL | TORTURE | INVALIDARGS\n"
00086 "  For the Privacy and Screening Modes, the DIALSTATUS variable will be set to\n"
00087 "DONTCALL if the called party chooses to send the calling party to the 'Go Away'\n"
00088 "script. The DIALSTATUS variable will be set to TORTURE if the called party\n"
00089 "wants to send the caller to the 'torture' script.\n"
00090 "  This application will report normal termination if the originating channel\n"
00091 "hangs up, or if the call is bridged and either of the parties in the bridge\n"
00092 "ends the call.\n"
00093 "  The optional URL will be sent to the called party if the channel supports it.\n"
00094 "  If the OUTBOUND_GROUP variable is set, all peer channels created by this\n"
00095 "application will be put into that group (as in Set(GROUP()=...).\n"
00096 "  If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this\n"
00097 "application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,\n"
00098 "however, the variable will be unset after use.\n\n"
00099 "  Options:\n"
00100 "    A(x) - Play an announcement to the called party, using 'x' as the file.\n"
00101 "    C    - Reset the CDR for this call.\n"
00102 "    c    - If DIAL cancels this call, always set the flag to tell the channel\n"
00103 "           driver that the call is answered elsewhere.\n"
00104 "    d    - Allow the calling user to dial a 1 digit extension while waiting for\n"
00105 "           a call to be answered. Exit to that extension if it exists in the\n"
00106 "           current context, or the context defined in the EXITCONTEXT variable,\n"
00107 "           if it exists.\n"
00108 "    D([called][:calling]) - Send the specified DTMF strings *after* the called\n"
00109 "           party has answered, but before the call gets bridged. The 'called'\n"
00110 "           DTMF string is sent to the called party, and the 'calling' DTMF\n"
00111 "           string is sent to the calling party. Both parameters can be used\n"
00112 "           alone.\n"
00113 "    e    - execute the 'h' extension for peer after the call ends\n"
00114 "    f    - Force the callerid of the *calling* channel to be set as the\n"
00115 "           extension associated with the channel using a dialplan 'hint'.\n"
00116 "           For example, some PSTNs do not allow CallerID to be set to anything\n"
00117 "           other than the number assigned to the caller.\n"
00118 "    g    - Proceed with dialplan execution at the current extension if the\n"
00119 "           destination channel hangs up.\n"
00120 "    G(context^exten^pri) - If the call is answered, transfer the calling party to\n"
00121 "           the specified priority and the called party to the specified priority+1.\n"
00122 "           Optionally, an extension, or extension and context may be specified. \n"
00123 "           Otherwise, the current extension is used. You cannot use any additional\n"
00124 "           action post answer options in conjunction with this option.\n"
00125 "    h    - Allow the called party to hang up by sending the '*' DTMF digit, or\n"
00126 "           whatever sequence was defined in the featuremap section for\n"
00127 "           'disconnect' in features.conf\n"
00128 "    H    - Allow the calling party to hang up by hitting the '*' DTMF digit, or\n"
00129 "           whatever sequence was defined in the featuremap section for\n"
00130 "           'disconnect' in features.conf\n"
00131 "    i    - Asterisk will ignore any forwarding requests it may receive on this\n"
00132 "           dial attempt.\n"
00133 "    k    - Allow the called party to enable parking of the call by sending\n"
00134 "           the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
00135 "    K    - Allow the calling party to enable parking of the call by sending\n"
00136 "           the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
00137 "    L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are\n"
00138 "           left. Repeat the warning every 'z' ms. The following special\n"
00139 "           variables can be used with this option:\n"
00140 "           * LIMIT_PLAYAUDIO_CALLER   yes|no (default yes)\n"
00141 "                                      Play sounds to the caller.\n"
00142 "           * LIMIT_PLAYAUDIO_CALLEE   yes|no\n"
00143 "                                      Play sounds to the callee.\n"
00144 "           * LIMIT_TIMEOUT_FILE       File to play when time is up.\n"
00145 "           * LIMIT_CONNECT_FILE       File to play when call begins.\n"
00146 "           * LIMIT_WARNING_FILE       File to play as warning if 'y' is defined.\n"
00147 "                                      The default is to say the time remaining.\n"
00148 "    m([class]) - Provide hold music to the calling party until a requested\n"
00149 "           channel answers. A specific MusicOnHold class can be\n"
00150 "           specified.\n"
00151 "    M(x[^arg]) - Execute the Macro for the *called* channel before connecting\n"
00152 "           to the calling channel. Arguments can be specified to the Macro\n"
00153 "           using '^' as a delimeter. The Macro can set the variable\n"
00154 "           MACRO_RESULT to specify the following actions after the Macro is\n"
00155 "           finished executing.\n"
00156 "           * ABORT        Hangup both legs of the call.\n"
00157 "           * CONGESTION   Behave as if line congestion was encountered.\n"
00158 "           * BUSY         Behave as if a busy signal was encountered.\n"
00159 "           * CONTINUE     Hangup the called party and allow the calling party\n"
00160 "                          to continue dialplan execution at the next priority.\n"
00161 "           * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
00162 "                          specified priority. Optionally, an extension, or\n"
00163 "                          extension and priority can be specified.\n"
00164 "           You cannot use any additional action post answer options in conjunction\n"
00165 "           with this option. Also, pbx services are not run on the peer (called) channel,\n"
00166 "           so you will not be able to set timeouts via the TIMEOUT() function in this macro.\n"
00167 "           Be aware of the limitations that macros have, specifically with regards to use of\n"
00168 "           the WaitExten application. For more information, see the documentation for Macro()\n"
00169 "    n    - This option is a modifier for the screen/privacy mode. It specifies\n"
00170 "           that no introductions are to be saved in the priv-callerintros\n"
00171 "           directory.\n"
00172 "    N    - This option is a modifier for the screen/privacy mode. It specifies\n"
00173 "           that if callerID is present, do not screen the call.\n"
00174 "    o    - Specify that the CallerID that was present on the *calling* channel\n"
00175 "           be set as the CallerID on the *called* channel. This was the\n"
00176 "           behavior of Asterisk 1.0 and earlier.\n"
00177 "    O([x]) - \"Operator Services\" mode (DAHDI channel to DAHDI channel\n"
00178 "             only, if specified on non-DAHDI interface, it will be ignored).\n"
00179 "             When the destination answers (presumably an operator services\n"
00180 "             station), the originator no longer has control of their line.\n"
00181 "             They may hang up, but the switch will not release their line\n"
00182 "             until the destination party hangs up (the operator). Specified\n"
00183 "             without an arg, or with 1 as an arg, the originator hanging up\n"
00184 "             will cause the phone to ring back immediately. With a 2 specified,\n"
00185 "             when the \"operator\" flashes the trunk, it will ring their phone\n"
00186 "             back.\n"
00187 "    p    - This option enables screening mode. This is basically Privacy mode\n"
00188 "           without memory.\n"
00189 "    P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if\n"
00190 "           it is provided. The current extension is used if a database\n"
00191 "           family/key is not specified.\n"
00192 "    r    - Indicate ringing to the calling party. Pass no audio to the calling\n"
00193 "           party until the called channel has answered.\n"
00194 "    S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
00195 "           answered the call.\n"
00196 "    t    - Allow the called party to transfer the calling party by sending the\n"
00197 "           DTMF sequence defined in the blindxfer setting in the featuremap section\n"
00198 "           of features.conf.\n"
00199 "    T    - Allow the calling party to transfer the called party by sending the\n"
00200 "           DTMF sequence defined in the blindxfer setting in the featuremap section\n"
00201 "           of features.conf.\n"
00202 "    U(x[^arg]) - Execute via Gosub the routine 'x' for the *called* channel before connecting\n"
00203 "           to the calling channel. Arguments can be specified to the Gosub\n"
00204 "           using '^' as a delimeter. The Gosub routine can set the variable\n"
00205 "           GOSUB_RESULT to specify the following actions after the Gosub returns.\n"
00206 "           * ABORT        Hangup both legs of the call.\n"
00207 "           * CONGESTION   Behave as if line congestion was encountered.\n"
00208 "           * BUSY         Behave as if a busy signal was encountered.\n"
00209 "           * CONTINUE     Hangup the called party and allow the calling party\n"
00210 "                          to continue dialplan execution at the next priority.\n"
00211 "           * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
00212 "                          specified priority. Optionally, an extension, or\n"
00213 "                          extension and priority can be specified.\n"
00214 "           You cannot use any additional action post answer options in conjunction\n"
00215 "           with this option. Also, pbx services are not run on the peer (called) channel,\n"
00216 "           so you will not be able to set timeouts via the TIMEOUT() function in this routine.\n"
00217 "    w    - Allow the called party to enable recording of the call by sending\n"
00218 "           the DTMF sequence defined in the automon setting in the featuremap section\n"
00219 "           of features.conf.\n"
00220 "    W    - Allow the calling party to enable recording of the call by sending\n"
00221 "           the DTMF sequence defined in the automon setting in the featuremap section\n"
00222 "           of features.conf.\n"
00223 "    x    - Allow the called party to enable recording of the call by sending\n"
00224 "           the DTMF sequence defined in the automixmon setting in the featuremap section\n"
00225 "           of features.conf.\n"
00226 "    X    - Allow the calling party to enable recording of the call by sending\n"
00227 "           the DTMF sequence defined in the automixmon setting in the featuremap section\n"
00228 "           of features.conf.\n";
00229 
00230 /* RetryDial App by Anthony Minessale II <anthmct@yahoo.com> Jan/2005 */
00231 static char *rapp = "RetryDial";
00232 static char *rsynopsis = "Place a call, retrying on failure allowing optional exit extension.";
00233 static char *rdescrip =
00234 "  RetryDial(announce,sleep,retries,dialargs): This application will attempt to\n"
00235 "place a call using the normal Dial application. If no channel can be reached,\n"
00236 "the 'announce' file will be played. Then, it will wait 'sleep' number of\n"
00237 "seconds before retrying the call. After 'retries' number of attempts, the\n"
00238 "calling channel will continue at the next priority in the dialplan. If the\n"
00239 "'retries' setting is set to 0, this application will retry endlessly.\n"
00240 "  While waiting to retry a call, a 1 digit extension may be dialed. If that\n"
00241 "extension exists in either the context defined in ${EXITCONTEXT} or the current\n"
00242 "one, The call will jump to that extension immediately.\n"
00243 "  The 'dialargs' are specified in the same format that arguments are provided\n"
00244 "to the Dial application.\n";
00245 
00246 enum {
00247    OPT_ANNOUNCE =          (1 << 0),
00248    OPT_RESETCDR =          (1 << 1),
00249    OPT_DTMF_EXIT =         (1 << 2),
00250    OPT_SENDDTMF =          (1 << 3),
00251    OPT_FORCECLID =         (1 << 4),
00252    OPT_GO_ON =             (1 << 5),
00253    OPT_CALLEE_HANGUP =     (1 << 6),
00254    OPT_CALLER_HANGUP =     (1 << 7),
00255    OPT_DURATION_LIMIT =    (1 << 9),
00256    OPT_MUSICBACK =         (1 << 10),
00257    OPT_CALLEE_MACRO =      (1 << 11),
00258    OPT_SCREEN_NOINTRO =    (1 << 12),
00259    OPT_SCREEN_NOCLID =     (1 << 13),
00260    OPT_ORIGINAL_CLID =     (1 << 14),
00261    OPT_SCREENING =         (1 << 15),
00262    OPT_PRIVACY =           (1 << 16),
00263    OPT_RINGBACK =          (1 << 17),
00264    OPT_DURATION_STOP =     (1 << 18),
00265    OPT_CALLEE_TRANSFER =   (1 << 19),
00266    OPT_CALLER_TRANSFER =   (1 << 20),
00267    OPT_CALLEE_MONITOR =    (1 << 21),
00268    OPT_CALLER_MONITOR =    (1 << 22),
00269    OPT_GOTO =              (1 << 23),
00270    OPT_OPERMODE =          (1 << 24),
00271    OPT_CALLEE_PARK =       (1 << 25),
00272    OPT_CALLER_PARK =       (1 << 26),
00273    OPT_IGNORE_FORWARDING = (1 << 27),
00274    OPT_CALLEE_GOSUB =      (1 << 28),
00275    OPT_CALLEE_MIXMONITOR = (1 << 29),
00276    OPT_CALLER_MIXMONITOR = (1 << 30),
00277 };
00278 
00279 #define DIAL_STILLGOING      (1 << 31)
00280 #define DIAL_NOFORWARDHTML   ((uint64_t)1 << 32) /* flags are now 64 bits, so keep it up! */
00281 #define OPT_CANCEL_ELSEWHERE ((uint64_t)1 << 33)
00282 #define OPT_PEER_H           ((uint64_t)1 << 34)
00283 
00284 enum {
00285    OPT_ARG_ANNOUNCE = 0,
00286    OPT_ARG_SENDDTMF,
00287    OPT_ARG_GOTO,
00288    OPT_ARG_DURATION_LIMIT,
00289    OPT_ARG_MUSICBACK,
00290    OPT_ARG_CALLEE_MACRO,
00291    OPT_ARG_CALLEE_GOSUB,
00292    OPT_ARG_PRIVACY,
00293    OPT_ARG_DURATION_STOP,
00294    OPT_ARG_OPERMODE,
00295    /* note: this entry _MUST_ be the last one in the enum */
00296    OPT_ARG_ARRAY_SIZE,
00297 };
00298 
00299 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
00300    AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
00301    AST_APP_OPTION('C', OPT_RESETCDR),
00302    AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
00303    AST_APP_OPTION('d', OPT_DTMF_EXIT),
00304    AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
00305    AST_APP_OPTION('e', OPT_PEER_H),
00306    AST_APP_OPTION('f', OPT_FORCECLID),
00307    AST_APP_OPTION('g', OPT_GO_ON),
00308    AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
00309    AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
00310    AST_APP_OPTION('H', OPT_CALLER_HANGUP),
00311    AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
00312    AST_APP_OPTION('k', OPT_CALLEE_PARK),
00313    AST_APP_OPTION('K', OPT_CALLER_PARK),
00314    AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
00315    AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
00316    AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
00317    AST_APP_OPTION('n', OPT_SCREEN_NOINTRO),
00318    AST_APP_OPTION('N', OPT_SCREEN_NOCLID),
00319    AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
00320    AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
00321    AST_APP_OPTION('p', OPT_SCREENING),
00322    AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
00323    AST_APP_OPTION('r', OPT_RINGBACK),
00324    AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
00325    AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
00326    AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
00327    AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
00328    AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
00329    AST_APP_OPTION('W', OPT_CALLER_MONITOR),
00330    AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
00331    AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
00332 END_OPTIONS );
00333 
00334 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
00335    OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
00336    OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK) && \
00337    !chan->audiohooks && !peer->audiohooks)
00338 
00339 /*
00340  * The list of active channels
00341  */
00342 struct chanlist {
00343    struct chanlist *next;
00344    struct ast_channel *chan;
00345    uint64_t flags;
00346 };
00347 
00348 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode);
00349 
00350 static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
00351 {
00352    /* Hang up a tree of stuff */
00353    struct chanlist *oo;
00354    while (outgoing) {
00355       /* Hangup any existing lines we have open */
00356       if (outgoing->chan && (outgoing->chan != exception)) {
00357          if (answered_elsewhere)
00358             ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
00359          ast_hangup(outgoing->chan);
00360       }
00361       oo = outgoing;
00362       outgoing = outgoing->next;
00363       ast_free(oo);
00364    }
00365 }
00366 
00367 #define AST_MAX_WATCHERS 256
00368 
00369 /*
00370  * argument to handle_cause() and other functions.
00371  */
00372 struct cause_args {
00373    struct ast_channel *chan;
00374    int busy;
00375    int congestion;
00376    int nochan;
00377 };
00378 
00379 static void handle_cause(int cause, struct cause_args *num)
00380 {
00381    struct ast_cdr *cdr = num->chan->cdr;
00382 
00383    switch(cause) {
00384    case AST_CAUSE_BUSY:
00385       if (cdr)
00386          ast_cdr_busy(cdr);
00387       num->busy++;
00388       break;
00389 
00390    case AST_CAUSE_CONGESTION:
00391       if (cdr)
00392          ast_cdr_failed(cdr);
00393       num->congestion++;
00394       break;
00395 
00396    case AST_CAUSE_NO_ROUTE_DESTINATION:
00397    case AST_CAUSE_UNREGISTERED:
00398       if (cdr)
00399          ast_cdr_failed(cdr);
00400       num->nochan++;
00401       break;
00402 
00403    case AST_CAUSE_NO_ANSWER:
00404       if (cdr) {
00405          ast_cdr_noanswer(cdr);
00406       }
00407       break;
00408    case AST_CAUSE_NORMAL_CLEARING:
00409       break;
00410 
00411    default:
00412       num->nochan++;
00413       break;
00414    }
00415 }
00416 
00417 /* free the buffer if allocated, and set the pointer to the second arg */
00418 #define S_REPLACE(s, new_val)    \
00419    do {           \
00420       if (s)         \
00421          ast_free(s);   \
00422       s = (new_val);    \
00423    } while (0)
00424 
00425 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
00426 {
00427    char rexten[2] = { exten, '\0' };
00428 
00429    if (context) {
00430       if (!ast_goto_if_exists(chan, context, rexten, pri))
00431          return 1;
00432    } else {
00433       if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
00434          return 1;
00435       else if (!ast_strlen_zero(chan->macrocontext)) {
00436          if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
00437             return 1;
00438       }
00439    }
00440    return 0;
00441 }
00442 
00443 
00444 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
00445 {
00446    const char *context = S_OR(chan->macrocontext, chan->context);
00447    const char *exten = S_OR(chan->macroexten, chan->exten);
00448 
00449    return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
00450 }
00451 
00452 static void senddialevent(struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
00453 {
00454    manager_event(EVENT_FLAG_CALL, "Dial",
00455       "SubEvent: Begin\r\n"
00456       "Channel: %s\r\n"
00457       "Destination: %s\r\n"
00458       "CallerIDNum: %s\r\n"
00459       "CallerIDName: %s\r\n"
00460       "UniqueID: %s\r\n"
00461       "DestUniqueID: %s\r\n"
00462       "Dialstring: %s\r\n",
00463       src->name, dst->name, S_OR(src->cid.cid_num, "<unknown>"),
00464       S_OR(src->cid.cid_name, "<unknown>"), src->uniqueid,
00465       dst->uniqueid, dialstring ? dialstring : "");
00466 }
00467 
00468 static void senddialendevent(const struct ast_channel *src, const char *dialstatus)
00469 {
00470    manager_event(EVENT_FLAG_CALL, "Dial",
00471       "SubEvent: End\r\n"
00472       "Channel: %s\r\n"
00473       "UniqueID: %s\r\n"
00474       "DialStatus: %s\r\n",
00475       src->name, src->uniqueid, dialstatus);
00476 }
00477 
00478 /*!
00479  * helper function for wait_for_answer()
00480  *
00481  * XXX this code is highly suspicious, as it essentially overwrites
00482  * the outgoing channel without properly deleting it.
00483  */
00484 static void do_forward(struct chanlist *o,
00485    struct cause_args *num, struct ast_flags64 *peerflags, int single)
00486 {
00487    char tmpchan[256];
00488    struct ast_channel *original = o->chan;
00489    struct ast_channel *c = o->chan; /* the winner */
00490    struct ast_channel *in = num->chan; /* the input channel */
00491    char *stuff;
00492    char *tech;
00493    int cause;
00494 
00495    ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
00496    if ((stuff = strchr(tmpchan, '/'))) {
00497       *stuff++ = '\0';
00498       tech = tmpchan;
00499    } else {
00500       const char *forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
00501       if (ast_strlen_zero(forward_context)) {
00502          forward_context = NULL;
00503       }
00504       snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
00505       stuff = tmpchan;
00506       tech = "Local";
00507    }
00508    /* Before processing channel, go ahead and check for forwarding */
00509    ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, c->name);
00510    /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
00511    if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
00512       ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
00513       c = o->chan = NULL;
00514       cause = AST_CAUSE_BUSY;
00515    } else {
00516       /* Setup parameters */
00517       c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
00518       if (c) {
00519          if (single)
00520             ast_channel_make_compatible(o->chan, in);
00521          ast_channel_inherit_variables(in, o->chan);
00522          ast_channel_datastore_inherit(in, o->chan);
00523       } else
00524          ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
00525    }
00526    if (!c) {
00527       ast_clear_flag64(o, DIAL_STILLGOING);
00528       handle_cause(cause, num);
00529       ast_hangup(original);
00530    } else {
00531       char *new_cid_num, *new_cid_name;
00532       struct ast_channel *src;
00533 
00534       ast_rtp_make_compatible(c, in, single);
00535       if (ast_test_flag64(o, OPT_FORCECLID)) {
00536          new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
00537          new_cid_name = NULL; /* XXX no name ? */
00538          src = c; /* XXX possible bug in previous code, which used 'winner' ? it may have changed */
00539       } else {
00540          new_cid_num = ast_strdup(in->cid.cid_num);
00541          new_cid_name = ast_strdup(in->cid.cid_name);
00542          src = in;
00543       }
00544       ast_string_field_set(c, accountcode, src->accountcode);
00545       c->cdrflags = src->cdrflags;
00546       S_REPLACE(c->cid.cid_num, new_cid_num);
00547       S_REPLACE(c->cid.cid_name, new_cid_name);
00548 
00549       if (in->cid.cid_ani) { /* XXX or maybe unconditional ? */
00550          S_REPLACE(c->cid.cid_ani, ast_strdup(in->cid.cid_ani));
00551       }
00552       S_REPLACE(c->cid.cid_rdnis, ast_strdup(S_OR(in->macroexten, in->exten)));
00553       if (ast_call(c, tmpchan, 0)) {
00554          ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
00555          ast_clear_flag64(o, DIAL_STILLGOING);
00556          ast_hangup(original);
00557          ast_hangup(c);
00558          c = o->chan = NULL;
00559          num->nochan++;
00560       } else {
00561          senddialevent(in, c, stuff);
00562          /* After calling, set callerid to extension */
00563          if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
00564             char cidname[AST_MAX_EXTENSION] = "";
00565             ast_set_callerid(c, S_OR(in->macroexten, in->exten), get_cid_name(cidname, sizeof(cidname), in), NULL);
00566          }
00567          /* Hangup the original channel now, in case we needed it */
00568          ast_hangup(original);
00569       }
00570       if (single) {
00571          ast_indicate(in, -1);
00572       }
00573    }
00574 }
00575 
00576 /* argument used for some functions. */
00577 struct privacy_args {
00578    int sentringing;
00579    int privdb_val;
00580    char privcid[256];
00581    char privintro[1024];
00582    char status[256];
00583 };
00584 
00585 static struct ast_channel *wait_for_answer(struct ast_channel *in,
00586    struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags,
00587    struct privacy_args *pa,
00588    const struct cause_args *num_in, int *result)
00589 {
00590    struct cause_args num = *num_in;
00591    int prestart = num.busy + num.congestion + num.nochan;
00592    int orig = *to;
00593    struct ast_channel *peer = NULL;
00594    /* single is set if only one destination is enabled */
00595    int single = outgoing && !outgoing->next && !ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
00596 #ifdef HAVE_EPOLL
00597    struct chanlist *epollo;
00598 #endif
00599    struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
00600    if (single) {
00601       /* Turn off hold music, etc */
00602       ast_deactivate_generator(in);
00603       /* If we are calling a single channel, make them compatible for in-band tone purpose */
00604       ast_channel_make_compatible(outgoing->chan, in);
00605    }
00606 
00607 #ifdef HAVE_EPOLL
00608    for (epollo = outgoing; epollo; epollo = epollo->next)
00609       ast_poll_channel_add(in, epollo->chan);
00610 #endif
00611 
00612    while (*to && !peer) {
00613       struct chanlist *o;
00614       int pos = 0; /* how many channels do we handle */
00615       int numlines = prestart;
00616       struct ast_channel *winner;
00617       struct ast_channel *watchers[AST_MAX_WATCHERS];
00618 
00619       watchers[pos++] = in;
00620       for (o = outgoing; o; o = o->next) {
00621          /* Keep track of important channels */
00622          if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
00623             watchers[pos++] = o->chan;
00624          numlines++;
00625       }
00626       if (pos == 1) { /* only the input channel is available */
00627          if (numlines == (num.busy + num.congestion + num.nochan)) {
00628             ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
00629             if (num.busy)
00630                strcpy(pa->status, "BUSY");
00631             else if (num.congestion)
00632                strcpy(pa->status, "CONGESTION");
00633             else if (num.nochan)
00634                strcpy(pa->status, "CHANUNAVAIL");
00635          } else {
00636             ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
00637          }
00638          *to = 0;
00639          return NULL;
00640       }
00641       winner = ast_waitfor_n(watchers, pos, to);
00642       for (o = outgoing; o; o = o->next) {
00643          struct ast_frame *f;
00644          struct ast_channel *c = o->chan;
00645 
00646          if (c == NULL)
00647             continue;
00648          if (ast_test_flag64(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
00649             if (!peer) {
00650                ast_verb(3, "%s answered %s\n", c->name, in->name);
00651                peer = c;
00652                ast_copy_flags64(peerflags, o,
00653                   OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
00654                   OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
00655                   OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
00656                   OPT_CALLEE_PARK | OPT_CALLER_PARK |
00657                   OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
00658                   DIAL_NOFORWARDHTML);
00659                ast_copy_string(c->dialcontext, "", sizeof(c->dialcontext));
00660                ast_copy_string(c->exten, "", sizeof(c->exten));
00661             }
00662             continue;
00663          }
00664          if (c != winner)
00665             continue;
00666          /* here, o->chan == c == winner */
00667          if (!ast_strlen_zero(c->call_forward)) {
00668             do_forward(o, &num, peerflags, single);
00669             continue;
00670          }
00671          f = ast_read(winner);
00672          if (!f) {
00673             in->hangupcause = c->hangupcause;
00674 #ifdef HAVE_EPOLL
00675             ast_poll_channel_del(in, c);
00676 #endif
00677             ast_hangup(c);
00678             c = o->chan = NULL;
00679             ast_clear_flag64(o, DIAL_STILLGOING);
00680             handle_cause(in->hangupcause, &num);
00681             continue;
00682          }
00683          if (f->frametype == AST_FRAME_CONTROL) {
00684             switch(f->subclass) {
00685             case AST_CONTROL_ANSWER:
00686                /* This is our guy if someone answered. */
00687                if (!peer) {
00688                   ast_verb(3, "%s answered %s\n", c->name, in->name);
00689                   peer = c;
00690                   if (peer->cdr) {
00691                      peer->cdr->answer = ast_tvnow();
00692                      peer->cdr->disposition = AST_CDR_ANSWERED;
00693                   }
00694                   ast_copy_flags64(peerflags, o,
00695                      OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
00696                      OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
00697                      OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
00698                      OPT_CALLEE_PARK | OPT_CALLER_PARK |
00699                      OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
00700                      DIAL_NOFORWARDHTML);
00701                   ast_copy_string(c->dialcontext, "", sizeof(c->dialcontext));
00702                   ast_copy_string(c->exten, "", sizeof(c->exten));
00703                   if (CAN_EARLY_BRIDGE(peerflags, in, peer))
00704                      /* Setup early bridge if appropriate */
00705                      ast_channel_early_bridge(in, peer);
00706                }
00707                /* If call has been answered, then the eventual hangup is likely to be normal hangup */
00708                in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
00709                c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
00710                break;
00711             case AST_CONTROL_BUSY:
00712                ast_verb(3, "%s is busy\n", c->name);
00713                in->hangupcause = c->hangupcause;
00714                ast_hangup(c);
00715                c = o->chan = NULL;
00716                ast_clear_flag64(o, DIAL_STILLGOING);
00717                handle_cause(AST_CAUSE_BUSY, &num);
00718                break;
00719             case AST_CONTROL_CONGESTION:
00720                ast_verb(3, "%s is circuit-busy\n", c->name);
00721                in->hangupcause = c->hangupcause;
00722                ast_hangup(c);
00723                c = o->chan = NULL;
00724                ast_clear_flag64(o, DIAL_STILLGOING);
00725                handle_cause(AST_CAUSE_CONGESTION, &num);
00726                break;
00727             case AST_CONTROL_RINGING:
00728                ast_verb(3, "%s is ringing\n", c->name);
00729                /* Setup early media if appropriate */
00730                if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
00731                   ast_channel_early_bridge(in, c);
00732                if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
00733                   ast_indicate(in, AST_CONTROL_RINGING);
00734                   pa->sentringing++;
00735                }
00736                break;
00737             case AST_CONTROL_PROGRESS:
00738                ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
00739                /* Setup early media if appropriate */
00740                if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
00741                   ast_channel_early_bridge(in, c);
00742                if (!ast_test_flag64(outgoing, OPT_RINGBACK))
00743                   ast_indicate(in, AST_CONTROL_PROGRESS);
00744                break;
00745             case AST_CONTROL_VIDUPDATE:
00746                ast_verb(3, "%s requested a video update, passing it to %s\n", c->name, in->name);
00747                ast_indicate(in, AST_CONTROL_VIDUPDATE);
00748                break;
00749             case AST_CONTROL_SRCUPDATE:
00750                ast_verb(3, "%s requested a source update, passing it to %s\n", c->name, in->name);
00751                ast_indicate(in, AST_CONTROL_SRCUPDATE);
00752                break;
00753             case AST_CONTROL_PROCEEDING:
00754                ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
00755                if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
00756                   ast_channel_early_bridge(in, c);
00757                if (!ast_test_flag64(outgoing, OPT_RINGBACK))
00758                   ast_indicate(in, AST_CONTROL_PROCEEDING);
00759                break;
00760             case AST_CONTROL_HOLD:
00761                ast_verb(3, "Call on %s placed on hold\n", c->name);
00762                ast_indicate(in, AST_CONTROL_HOLD);
00763                break;
00764             case AST_CONTROL_UNHOLD:
00765                ast_verb(3, "Call on %s left from hold\n", c->name);
00766                ast_indicate(in, AST_CONTROL_UNHOLD);
00767                break;
00768             case AST_CONTROL_OFFHOOK:
00769             case AST_CONTROL_FLASH:
00770                /* Ignore going off hook and flash */
00771                break;
00772             case -1:
00773                if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
00774                   ast_verb(3, "%s stopped sounds\n", c->name);
00775                   ast_indicate(in, -1);
00776                   pa->sentringing = 0;
00777                }
00778                break;
00779             default:
00780                ast_debug(1, "Dunno what to do with control type %d\n", f->subclass);
00781             }
00782          } else if (single) {
00783             /* XXX are we sure the logic is correct ? or we should just switch on f->frametype ? */
00784             if (f->frametype == AST_FRAME_VOICE && !ast_test_flag64(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
00785                if (ast_write(in, f))
00786                   ast_log(LOG_WARNING, "Unable to forward voice frame\n");
00787             } else if (f->frametype == AST_FRAME_IMAGE && !ast_test_flag64(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
00788                if (ast_write(in, f))
00789                   ast_log(LOG_WARNING, "Unable to forward image\n");
00790             } else if (f->frametype == AST_FRAME_TEXT && !ast_test_flag64(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
00791                if (ast_write(in, f))
00792                   ast_log(LOG_WARNING, "Unable to send text\n");
00793             } else if (f->frametype == AST_FRAME_HTML && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)) {
00794                if (ast_channel_sendhtml(in, f->subclass, f->data, f->datalen) == -1)
00795                   ast_log(LOG_WARNING, "Unable to send URL\n");
00796             }
00797          }
00798          ast_frfree(f);
00799       } /* end for */
00800       if (winner == in) {
00801          struct ast_frame *f = ast_read(in);
00802 #if 0
00803          if (f && (f->frametype != AST_FRAME_VOICE))
00804             printf("Frame type: %d, %d\n", f->frametype, f->subclass);
00805          else if (!f || (f->frametype != AST_FRAME_VOICE))
00806             printf("Hangup received on %s\n", in->name);
00807 #endif
00808          if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
00809             /* Got hung up */
00810             *to = -1;
00811             strcpy(pa->status, "CANCEL");
00812             ast_cdr_noanswer(in->cdr);
00813             if (f)
00814                ast_frfree(f);
00815             return NULL;
00816          }
00817 
00818          /* now f is guaranteed non-NULL */
00819          if (f->frametype == AST_FRAME_DTMF) {
00820             if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
00821                const char *context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
00822                if (onedigit_goto(in, context, (char) f->subclass, 1)) {
00823                   ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
00824                   *to = 0;
00825                   ast_cdr_noanswer(in->cdr);
00826                   *result = f->subclass;
00827                   strcpy(pa->status, "CANCEL");
00828                   ast_frfree(f);
00829                   return NULL;
00830                }
00831             }
00832 
00833             if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
00834                detect_disconnect(in, f->subclass, featurecode)) {
00835                ast_verb(3, "User requested call disconnect.\n");
00836                *to = 0;
00837                strcpy(pa->status, "CANCEL");
00838                ast_cdr_noanswer(in->cdr);
00839                ast_frfree(f);
00840                return NULL;
00841             }
00842          }
00843 
00844          /* Forward HTML stuff */
00845          if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML))
00846             if (ast_channel_sendhtml(outgoing->chan, f->subclass, f->data, f->datalen) == -1)
00847                ast_log(LOG_WARNING, "Unable to send URL\n");
00848 
00849          if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF_BEGIN) || (f->frametype == AST_FRAME_DTMF_END)))  {
00850             if (ast_write(outgoing->chan, f))
00851                ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
00852          }
00853          if (single && (f->frametype == AST_FRAME_CONTROL) &&
00854             ((f->subclass == AST_CONTROL_HOLD) ||
00855             (f->subclass == AST_CONTROL_UNHOLD) ||
00856             (f->subclass == AST_CONTROL_VIDUPDATE) ||
00857              (f->subclass == AST_CONTROL_SRCUPDATE))) {
00858             ast_verb(3, "%s requested special control %d, passing it to %s\n", in->name, f->subclass, outgoing->chan->name);
00859             ast_indicate_data(outgoing->chan, f->subclass, f->data, f->datalen);
00860          }
00861          ast_frfree(f);
00862       }
00863       if (!*to)
00864          ast_verb(3, "Nobody picked up in %d ms\n", orig);
00865       if (!*to || ast_check_hangup(in))
00866          ast_cdr_noanswer(in->cdr);
00867    }
00868 
00869 #ifdef HAVE_EPOLL
00870    for (epollo = outgoing; epollo; epollo = epollo->next) {
00871       if (epollo->chan)
00872          ast_poll_channel_del(in, epollo->chan);
00873    }
00874 #endif
00875 
00876    return peer;
00877 }
00878 
00879 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode)
00880 {
00881    struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
00882    struct ast_call_feature feature = { 0, };
00883    int res;
00884 
00885    ast_str_append(&featurecode, 1, "%c", code);
00886 
00887    res = ast_feature_detect(chan, &features, ast_str_buffer(featurecode), &feature);
00888 
00889    if (res != AST_FEATURE_RETURN_STOREDIGITS) {
00890       ast_str_reset(featurecode);
00891    }
00892    if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
00893       return 1;
00894    }
00895 
00896    return 0;
00897 }
00898 
00899 static void replace_macro_delimiter(char *s)
00900 {
00901    for (; *s; s++)
00902       if (*s == '^')
00903          *s = ',';
00904 }
00905 
00906 /* returns true if there is a valid privacy reply */
00907 static int valid_priv_reply(struct ast_flags64 *opts, int res)
00908 {
00909    if (res < '1')
00910       return 0;
00911    if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
00912       return 1;
00913    if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
00914       return 1;
00915    return 0;
00916 }
00917 
00918 static int do_timelimit(struct ast_channel *chan, struct ast_bridge_config *config,
00919    char *parse, unsigned int *calldurationlimit)
00920 {
00921    char *stringp = ast_strdupa(parse);
00922    char *limit_str, *warning_str, *warnfreq_str;
00923    const char *var;
00924    int play_to_caller = 0, play_to_callee = 0;
00925    int delta;
00926 
00927    limit_str = strsep(&stringp, ":");
00928    warning_str = strsep(&stringp, ":");
00929    warnfreq_str = strsep(&stringp, ":");
00930 
00931    config->timelimit = atol(limit_str);
00932    if (warning_str)
00933       config->play_warning = atol(warning_str);
00934    if (warnfreq_str)
00935       config->warning_freq = atol(warnfreq_str);
00936 
00937    if (!config->timelimit) {
00938       ast_log(LOG_WARNING, "Dial does not accept L(%s), hanging up.\n", limit_str);
00939       config->timelimit = config->play_warning = config->warning_freq = 0;
00940       config->warning_sound = NULL;
00941       return -1; /* error */
00942    } else if ( (delta = config->play_warning - config->timelimit) > 0) {
00943       int w = config->warning_freq;
00944 
00945       /* If the first warning is requested _after_ the entire call would end,
00946          and no warning frequency is requested, then turn off the warning. If
00947          a warning frequency is requested, reduce the 'first warning' time by
00948          that frequency until it falls within the call's total time limit.
00949          Graphically:
00950               timelim->|    delta        |<-playwarning
00951          0__________________|_________________|
00952                 | w  |    |    |    |
00953 
00954          so the number of intervals to cut is 1+(delta-1)/w
00955       */
00956 
00957       if (w == 0) {
00958          config->play_warning = 0;
00959       } else {
00960          config->play_warning -= w * ( 1 + (delta-1)/w );
00961          if (config->play_warning < 1)
00962             config->play_warning = config->warning_freq = 0;
00963       }
00964    }
00965 
00966    var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLER");
00967    play_to_caller = var ? ast_true(var) : 1;
00968 
00969    var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLEE");
00970    play_to_callee = var ? ast_true(var) : 0;
00971 
00972    if (!play_to_caller && !play_to_callee)
00973       play_to_caller = 1;
00974 
00975    var = pbx_builtin_getvar_helper(chan, "LIMIT_WARNING_FILE");
00976    config->warning_sound = S_OR(var, "timeleft");
00977 
00978    /* The code looking at config wants a NULL, not just "", to decide
00979     * that the message should not be played, so we replace "" with NULL.
00980     * Note, pbx_builtin_getvar_helper _can_ return NULL if the variable is
00981     * not found.
00982     */
00983    var = pbx_builtin_getvar_helper(chan, "LIMIT_TIMEOUT_FILE");
00984    config->end_sound = S_OR(var, NULL);
00985    var = pbx_builtin_getvar_helper(chan, "LIMIT_CONNECT_FILE");
00986    config->start_sound = S_OR(var, NULL);
00987 
00988    /* undo effect of S(x) in case they are both used */
00989    *calldurationlimit = 0;
00990    /* more efficient to do it like S(x) does since no advanced opts */
00991    if (!config->play_warning && !config->start_sound && !config->end_sound && config->timelimit) {
00992       if (config->timelimit > 1000) {
00993          *calldurationlimit = config->timelimit / 1000;
00994       } else if (config->timelimit > 0) {
00995          /* Not enough granularity to make it less, but we can't use the special value 0 */
00996          *calldurationlimit = 1;
00997       }
00998       ast_verb(3, "Setting call duration limit to %d seconds.\n",
00999          *calldurationlimit);
01000       config->timelimit = play_to_caller = play_to_callee =
01001       config->play_warning = config->warning_freq = 0;
01002    } else {
01003       ast_verb(3, "Limit Data for this call:\n");
01004       ast_verb(4, "timelimit      = %ld\n", config->timelimit);
01005       ast_verb(4, "play_warning   = %ld\n", config->play_warning);
01006       ast_verb(4, "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
01007       ast_verb(4, "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
01008       ast_verb(4, "warning_freq   = %ld\n", config->warning_freq);
01009       ast_verb(4, "start_sound    = %s\n", S_OR(config->start_sound, ""));
01010       ast_verb(4, "warning_sound  = %s\n", config->warning_sound);
01011       ast_verb(4, "end_sound      = %s\n", S_OR(config->end_sound, ""));
01012    }
01013    if (play_to_caller)
01014       ast_set_flag(&(config->features_caller), AST_FEATURE_PLAY_WARNING);
01015    if (play_to_callee)
01016       ast_set_flag(&(config->features_callee), AST_FEATURE_PLAY_WARNING);
01017    return 0;
01018 }
01019 
01020 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
01021    struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
01022 {
01023 
01024    int res2;
01025    int loopcount = 0;
01026 
01027    /* Get the user's intro, store it in priv-callerintros/$CID,
01028       unless it is already there-- this should be done before the
01029       call is actually dialed  */
01030 
01031    /* all ring indications and moh for the caller has been halted as soon as the
01032       target extension was picked up. We are going to have to kill some
01033       time and make the caller believe the peer hasn't picked up yet */
01034 
01035    if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
01036       char *original_moh = ast_strdupa(chan->musicclass);
01037       ast_indicate(chan, -1);
01038       ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
01039       ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
01040       ast_string_field_set(chan, musicclass, original_moh);
01041    } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
01042       ast_indicate(chan, AST_CONTROL_RINGING);
01043       pa->sentringing++;
01044    }
01045 
01046    /* Start autoservice on the other chan ?? */
01047    res2 = ast_autoservice_start(chan);
01048    /* Now Stream the File */
01049    for (loopcount = 0; loopcount < 3; loopcount++) {
01050       if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
01051          break;
01052       if (!res2) /* on timeout, play the message again */
01053          res2 = ast_play_and_wait(peer, "priv-callpending");
01054       if (!valid_priv_reply(opts, res2))
01055          res2 = 0;
01056       /* priv-callpending script:
01057          "I have a caller waiting, who introduces themselves as:"
01058       */
01059       if (!res2)
01060          res2 = ast_play_and_wait(peer, pa->privintro);
01061       if (!valid_priv_reply(opts, res2))
01062          res2 = 0;
01063       /* now get input from the called party, as to their choice */
01064       if (!res2) {
01065          /* XXX can we have both, or they are mutually exclusive ? */
01066          if (ast_test_flag64(opts, OPT_PRIVACY))
01067             res2 = ast_play_and_wait(peer, "priv-callee-options");
01068          if (ast_test_flag64(opts, OPT_SCREENING))
01069             res2 = ast_play_and_wait(peer, "screen-callee-options");
01070       }
01071       /*! \page DialPrivacy Dial Privacy scripts
01072       \par priv-callee-options script:
01073          "Dial 1 if you wish this caller to reach you directly in the future,
01074             and immediately connect to their incoming call
01075           Dial 2 if you wish to send this caller to voicemail now and
01076             forevermore.
01077           Dial 3 to send this caller to the torture menus, now and forevermore.
01078           Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
01079           Dial 5 to allow this caller to come straight thru to you in the future,
01080             but right now, just this once, send them to voicemail."
01081       \par screen-callee-options script:
01082          "Dial 1 if you wish to immediately connect to the incoming call
01083           Dial 2 if you wish to send this caller to voicemail.
01084           Dial 3 to send this caller to the torture menus.
01085           Dial 4 to send this caller to a simple "go away" menu.
01086       */
01087       if (valid_priv_reply(opts, res2))
01088          break;
01089       /* invalid option */
01090       res2 = ast_play_and_wait(peer, "vm-sorry");
01091    }
01092 
01093    if (ast_test_flag64(opts, OPT_MUSICBACK)) {
01094       ast_moh_stop(chan);
01095    } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
01096       ast_indicate(chan, -1);
01097       pa->sentringing = 0;
01098    }
01099    ast_autoservice_stop(chan);
01100    if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
01101       /* map keypresses to various things, the index is res2 - '1' */
01102       static const char *_val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
01103       static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
01104       int i = res2 - '1';
01105       ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
01106          opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
01107       ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
01108    }
01109    switch (res2) {
01110    case '1':
01111       break;
01112    case '2':
01113       ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
01114       break;
01115    case '3':
01116       ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
01117       break;
01118    case '4':
01119       ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
01120       break;
01121    case '5':
01122       /* XXX should we set status to DENY ? */
01123       if (ast_test_flag64(opts, OPT_PRIVACY))
01124          break;
01125       /* if not privacy, then 5 is the same as "default" case */
01126    default: /* bad input or -1 if failure to start autoservice */
01127       /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do?  */
01128       /* well, there seems basically two choices. Just patch the caller thru immediately,
01129            or,... put 'em thru to voicemail. */
01130       /* since the callee may have hung up, let's do the voicemail thing, no database decision */
01131       ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
01132       /* XXX should we set status to DENY ? */
01133       /* XXX what about the privacy flags ? */
01134       break;
01135    }
01136 
01137    if (res2 == '1') { /* the only case where we actually connect */
01138       /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
01139          just clog things up, and it's not useful information, not being tied to a CID */
01140       if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
01141          ast_filedelete(pa->privintro, NULL);
01142          if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
01143             ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
01144          else
01145             ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
01146       }
01147       return 0; /* the good exit path */
01148    } else {
01149       ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
01150       return -1;
01151    }
01152 }
01153 
01154 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
01155 static int setup_privacy_args(struct privacy_args *pa,
01156    struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
01157 {
01158    char callerid[60];
01159    int res;
01160    char *l;
01161 
01162    if (!ast_strlen_zero(chan->cid.cid_num)) {
01163       l = ast_strdupa(chan->cid.cid_num);
01164       ast_shrink_phone_number(l);
01165       if (ast_test_flag64(opts, OPT_PRIVACY) ) {
01166          ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
01167          pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
01168       } else {
01169          ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
01170          pa->privdb_val = AST_PRIVACY_UNKNOWN;
01171       }
01172    } else {
01173       char *tnam, *tn2;
01174 
01175       tnam = ast_strdupa(chan->name);
01176       /* clean the channel name so slashes don't try to end up in disk file name */
01177       for (tn2 = tnam; *tn2; tn2++) {
01178          if (*tn2 == '/')  /* any other chars to be afraid of? */
01179             *tn2 = '=';
01180       }
01181       ast_verb(3, "Privacy-- callerid is empty\n");
01182 
01183       snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
01184       l = callerid;
01185       pa->privdb_val = AST_PRIVACY_UNKNOWN;
01186    }
01187 
01188    ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
01189 
01190    if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCLID)) {
01191       /* if callerid is set and OPT_SCREEN_NOCLID is set also */
01192       ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
01193       pa->privdb_val = AST_PRIVACY_ALLOW;
01194    } else if (ast_test_flag64(opts, OPT_SCREEN_NOCLID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
01195       ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
01196    }
01197    
01198    if (pa->privdb_val == AST_PRIVACY_DENY) {
01199       ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
01200       ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
01201       return 0;
01202    } else if (pa->privdb_val == AST_PRIVACY_KILL) {
01203       ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
01204       return 0; /* Is this right? */
01205    } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
01206       ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
01207       return 0; /* is this right??? */
01208    } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
01209       /* Get the user's intro, store it in priv-callerintros/$CID,
01210          unless it is already there-- this should be done before the
01211          call is actually dialed  */
01212 
01213       /* make sure the priv-callerintros dir actually exists */
01214       snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
01215       if ((res = ast_mkdir(pa->privintro, 0755))) {
01216          ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
01217          return -1;
01218       }
01219 
01220       snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
01221       if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
01222          /* the DELUX version of this code would allow this caller the
01223             option to hear and retape their previously recorded intro.
01224          */
01225       } else {
01226          int duration; /* for feedback from play_and_wait */
01227          /* the file doesn't exist yet. Let the caller submit his
01228             vocal intro for posterity */
01229          /* priv-recordintro script:
01230 
01231             "At the tone, please say your name:"
01232 
01233          */
01234          ast_answer(chan);
01235          res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "gsm", &duration, 128, 2000, 0);  /* NOTE: I've reduced the total time to 4 sec */
01236                            /* don't think we'll need a lock removed, we took care of
01237                               conflicts by naming the pa.privintro file */
01238          if (res == -1) {
01239             /* Delete the file regardless since they hung up during recording */
01240             ast_filedelete(pa->privintro, NULL);
01241             if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
01242                ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
01243             else
01244                ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
01245             return -1;
01246          }
01247          if (!ast_streamfile(chan, "vm-dialout", chan->language) )
01248             ast_waitstream(chan, "");
01249       }
01250    }
01251    return 1; /* success */
01252 }
01253 
01254 static void end_bridge_callback(void *data)
01255 {
01256    char buf[80];
01257    time_t end;
01258    struct ast_channel *chan = data;
01259 
01260    if (!chan->cdr) {
01261       return;
01262    }
01263 
01264    time(&end);
01265 
01266    ast_channel_lock(chan);
01267    if (chan->cdr->answer.tv_sec) {
01268       snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
01269       pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
01270    }
01271 
01272    if (chan->cdr->start.tv_sec) {
01273       snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
01274       pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
01275    }
01276    ast_channel_unlock(chan);
01277 }
01278 
01279 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
01280    bconfig->end_bridge_callback_data = originator;
01281 }
01282 
01283 static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags64 *peerflags, int *continue_exec)
01284 {
01285    int res = -1; /* default: error */
01286    char *rest, *cur; /* scan the list of destinations */
01287    struct chanlist *outgoing = NULL; /* list of destinations */
01288    struct ast_channel *peer;
01289    int to; /* timeout */
01290    struct cause_args num = { chan, 0, 0, 0 };
01291    int cause;
01292    char numsubst[256];
01293    char cidname[AST_MAX_EXTENSION] = "";
01294 
01295    struct ast_bridge_config config = { { 0, } };
01296    unsigned int calldurationlimit = 0;
01297    char *dtmfcalled = NULL, *dtmfcalling = NULL;
01298    struct privacy_args pa = {
01299       .sentringing = 0,
01300       .privdb_val = 0,
01301       .status = "INVALIDARGS",
01302    };
01303    int sentringing = 0, moh = 0;
01304    const char *outbound_group = NULL;
01305    int result = 0;
01306    char *parse;
01307    int opermode = 0;
01308    AST_DECLARE_APP_ARGS(args,
01309       AST_APP_ARG(peers);
01310       AST_APP_ARG(timeout);
01311       AST_APP_ARG(options);
01312       AST_APP_ARG(url);
01313    );
01314    struct ast_flags64 opts = { 0, };
01315    char *opt_args[OPT_ARG_ARRAY_SIZE];
01316    struct ast_datastore *datastore = NULL;
01317    int fulldial = 0, num_dialed = 0;
01318 
01319    /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
01320    pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
01321    pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
01322    pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
01323    pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
01324    pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
01325 
01326    if (ast_strlen_zero(data)) {
01327       ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
01328       pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01329       return -1;
01330    }
01331 
01332    parse = ast_strdupa(data);
01333 
01334    AST_STANDARD_APP_ARGS(args, parse);
01335 
01336    if (!ast_strlen_zero(args.options) &&
01337       ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
01338       pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01339       goto done;
01340    }
01341 
01342    if (ast_strlen_zero(args.peers)) {
01343       ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
01344       pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01345       goto done;
01346    }
01347 
01348    if (ast_test_flag64(&opts, OPT_OPERMODE)) {
01349       opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
01350       ast_verb(3, "Setting operator services mode to %d.\n", opermode);
01351    }
01352    
01353    if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
01354       calldurationlimit = atoi(opt_args[OPT_ARG_DURATION_STOP]);
01355       if (!calldurationlimit) {
01356          ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
01357          pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01358          goto done;
01359       }
01360       ast_verb(3, "Setting call duration limit to %d seconds.\n", calldurationlimit);
01361    }
01362 
01363    if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
01364       dtmfcalling = opt_args[OPT_ARG_SENDDTMF];
01365       dtmfcalled = strsep(&dtmfcalling, ":");
01366    }
01367 
01368    if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
01369       if (do_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
01370          goto done;
01371    }
01372 
01373    if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
01374       ast_cdr_reset(chan->cdr, NULL);
01375    if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
01376       opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
01377 
01378    if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
01379       res = setup_privacy_args(&pa, &opts, opt_args, chan);
01380       if (res <= 0)
01381          goto out;
01382       res = -1; /* reset default */
01383    }
01384 
01385    if (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
01386       __ast_answer(chan, 0, 0);
01387    }
01388 
01389    if (continue_exec)
01390       *continue_exec = 0;
01391 
01392    /* If a channel group has been specified, get it for use when we create peer channels */
01393    if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
01394       outbound_group = ast_strdupa(outbound_group);
01395       pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
01396    } else {
01397       outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP");
01398    }
01399 
01400    ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING);
01401 
01402    /* loop through the list of dial destinations */
01403    rest = args.peers;
01404    while ((cur = strsep(&rest, "&")) ) {
01405       struct chanlist *tmp;
01406       struct ast_channel *tc; /* channel for this destination */
01407       /* Get a technology/[device:]number pair */
01408       char *number = cur;
01409       char *interface = ast_strdupa(number);
01410       char *tech = strsep(&number, "/");
01411       /* find if we already dialed this interface */
01412       struct ast_dialed_interface *di;
01413       AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
01414       num_dialed++;
01415       if (!number) {
01416          ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
01417          goto out;
01418       }
01419       if (!(tmp = ast_calloc(1, sizeof(*tmp))))
01420          goto out;
01421       if (opts.flags) {
01422          ast_copy_flags64(tmp, &opts,
01423             OPT_CANCEL_ELSEWHERE |
01424             OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
01425             OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
01426             OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
01427             OPT_CALLEE_PARK | OPT_CALLER_PARK |
01428             OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
01429             OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
01430          ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
01431       }
01432       ast_copy_string(numsubst, number, sizeof(numsubst));
01433       /* Request the peer */
01434 
01435       ast_channel_lock(chan);
01436       datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
01437       ast_channel_unlock(chan);
01438 
01439       if (datastore)
01440          dialed_interfaces = datastore->data;
01441       else {
01442          if (!(datastore = ast_channel_datastore_alloc(&dialed_interface_info, NULL))) {
01443             ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
01444             ast_free(tmp);
01445             goto out;
01446          }
01447 
01448          datastore->inheritance = DATASTORE_INHERIT_FOREVER;
01449 
01450          if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
01451             ast_free(tmp);
01452             goto out;
01453          }
01454 
01455          datastore->data = dialed_interfaces;
01456          AST_LIST_HEAD_INIT(dialed_interfaces);
01457 
01458          ast_channel_lock(chan);
01459          ast_channel_datastore_add(chan, datastore);
01460          ast_channel_unlock(chan);
01461       }
01462 
01463       AST_LIST_LOCK(dialed_interfaces);
01464       AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
01465          if (!strcasecmp(di->interface, interface)) {
01466             ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
01467                di->interface);
01468             break;
01469          }
01470       }
01471       AST_LIST_UNLOCK(dialed_interfaces);
01472 
01473       if (di) {
01474          fulldial++;
01475          ast_free(tmp);
01476          continue;
01477       }
01478 
01479       /* It is always ok to dial a Local interface.  We only keep track of
01480        * which "real" interfaces have been dialed.  The Local channel will
01481        * inherit this list so that if it ends up dialing a real interface,
01482        * it won't call one that has already been called. */
01483       if (strcasecmp(tech, "Local")) {
01484          if (!(di = ast_calloc(1, sizeof(*di) + strlen(interface)))) {
01485             AST_LIST_UNLOCK(dialed_interfaces);
01486             ast_free(tmp);
01487             goto out;
01488          }
01489          strcpy(di->interface, interface);
01490 
01491          AST_LIST_LOCK(dialed_interfaces);
01492          AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
01493          AST_LIST_UNLOCK(dialed_interfaces);
01494       }
01495 
01496       tc = ast_request(tech, chan->nativeformats, numsubst, &cause);
01497       if (!tc) {
01498          /* If we can't, just go on to the next call */
01499          ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
01500             tech, cause, ast_cause2str(cause));
01501          handle_cause(cause, &num);
01502          if (!rest) /* we are on the last destination */
01503             chan->hangupcause = cause;
01504          ast_free(tmp);
01505          continue;
01506       }
01507       pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
01508 
01509       /* Setup outgoing SDP to match incoming one */
01510       ast_rtp_make_compatible(tc, chan, !outgoing && !rest);
01511       
01512       /* Inherit specially named variables from parent channel */
01513       ast_channel_inherit_variables(chan, tc);
01514       ast_channel_datastore_inherit(chan, tc);
01515 
01516       tc->appl = "AppDial";
01517       tc->data = "(Outgoing Line)";
01518       tc->whentohangup = 0;
01519 
01520       S_REPLACE(tc->cid.cid_num, ast_strdup(chan->cid.cid_num));
01521       S_REPLACE(tc->cid.cid_name, ast_strdup(chan->cid.cid_name));
01522       S_REPLACE(tc->cid.cid_ani, ast_strdup(chan->cid.cid_ani));
01523       S_REPLACE(tc->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
01524       
01525       /* Copy language from incoming to outgoing */
01526       ast_string_field_set(tc, language, chan->language);
01527       ast_string_field_set(tc, accountcode, chan->accountcode);
01528       tc->cdrflags = chan->cdrflags;
01529       if (ast_strlen_zero(tc->musicclass))
01530          ast_string_field_set(tc, musicclass, chan->musicclass);
01531       /* Pass callingpres, type of number, tns, ADSI CPE, transfer capability */
01532       tc->cid.cid_pres = chan->cid.cid_pres;
01533       tc->cid.cid_ton = chan->cid.cid_ton;
01534       tc->cid.cid_tns = chan->cid.cid_tns;
01535       tc->cid.cid_ani2 = chan->cid.cid_ani2;
01536       tc->adsicpe = chan->adsicpe;
01537       tc->transfercapability = chan->transfercapability;
01538 
01539       /* If we have an outbound group, set this peer channel to it */
01540       if (outbound_group)
01541          ast_app_group_set_channel(tc, outbound_group);
01542 
01543       /* Inherit context and extension */
01544       if (!ast_strlen_zero(chan->macrocontext))
01545          ast_copy_string(tc->dialcontext, chan->macrocontext, sizeof(tc->dialcontext));
01546       else
01547          ast_copy_string(tc->dialcontext, chan->context, sizeof(tc->dialcontext));
01548       if (!ast_strlen_zero(chan->macroexten))
01549          ast_copy_string(tc->exten, chan->macroexten, sizeof(tc->exten));
01550       else
01551          ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));
01552 
01553       res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
01554 
01555       /* Save the info in cdr's that we called them */
01556       if (chan->cdr)
01557          ast_cdr_setdestchan(chan->cdr, tc->name);
01558 
01559       /* check the results of ast_call */
01560       if (res) {
01561          /* Again, keep going even if there's an error */
01562          ast_debug(1, "ast call on peer returned %d\n", res);
01563          ast_verb(3, "Couldn't call %s\n", numsubst);
01564          if (tc->hangupcause) {
01565             chan->hangupcause = tc->hangupcause;
01566          }
01567          ast_hangup(tc);
01568          tc = NULL;
01569          ast_free(tmp);
01570          continue;
01571       } else {
01572          senddialevent(chan, tc, numsubst);
01573          ast_verb(3, "Called %s\n", numsubst);
01574          if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID))
01575             ast_set_callerid(tc, S_OR(chan->macroexten, chan->exten), get_cid_name(cidname, sizeof(cidname), chan), NULL);
01576       }
01577       /* Put them in the list of outgoing thingies...  We're ready now.
01578          XXX If we're forcibly removed, these outgoing calls won't get
01579          hung up XXX */
01580       ast_set_flag64(tmp, DIAL_STILLGOING);
01581       tmp->chan = tc;
01582       tmp->next = outgoing;
01583       outgoing = tmp;
01584       /* If this line is up, don't try anybody else */
01585       if (outgoing->chan->_state == AST_STATE_UP)
01586          break;
01587    }
01588    
01589    if (ast_strlen_zero(args.timeout)) {
01590       to = -1;
01591    } else {
01592       to = atoi(args.timeout);
01593       if (to > 0)
01594          to *= 1000;
01595       else
01596          ast_log(LOG_WARNING, "Invalid timeout specified: '%s'\n", args.timeout);
01597    }
01598 
01599    if (!outgoing) {
01600       strcpy(pa.status, "CHANUNAVAIL");
01601       if (fulldial == num_dialed) {
01602          res = -1;
01603          goto out;
01604       }
01605    } else {
01606       /* Our status will at least be NOANSWER */
01607       strcpy(pa.status, "NOANSWER");
01608       if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
01609          moh = 1;
01610          if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
01611             char *original_moh = ast_strdupa(chan->musicclass);
01612             ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
01613             ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
01614             ast_string_field_set(chan, musicclass, original_moh);
01615          } else {
01616             ast_moh_start(chan, NULL, NULL);
01617          }
01618          ast_indicate(chan, AST_CONTROL_PROGRESS);
01619       } else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
01620          ast_indicate(chan, AST_CONTROL_RINGING);
01621          sentringing++;
01622       }
01623    }
01624 
01625    peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &result);
01626 
01627    /* The ast_channel_datastore_remove() function could fail here if the
01628     * datastore was moved to another channel during a masquerade. If this is
01629     * the case, don't free the datastore here because later, when the channel
01630     * to which the datastore was moved hangs up, it will attempt to free this
01631     * datastore again, causing a crash
01632     */
01633    if (!ast_channel_datastore_remove(chan, datastore))
01634       ast_channel_datastore_free(datastore);
01635    if (!peer) {
01636       if (result) {
01637          res = result;
01638       } else if (to) { /* Musta gotten hung up */
01639          res = -1;
01640       } else { /* Nobody answered, next please? */
01641          res = 0;
01642       }
01643       /* almost done, although the 'else' block is 400 lines */
01644    } else {
01645       const char *number;
01646 
01647       strcpy(pa.status, "ANSWER");
01648       pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01649       /* Ah ha!  Someone answered within the desired timeframe.  Of course after this
01650          we will always return with -1 so that it is hung up properly after the
01651          conversation.  */
01652       hanguptree(outgoing, peer, 1);
01653       outgoing = NULL;
01654       /* If appropriate, log that we have a destination channel */
01655       if (chan->cdr)
01656          ast_cdr_setdestchan(chan->cdr, peer->name);
01657       if (peer->name)
01658          pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
01659 
01660       number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
01661       if (!number)
01662          number = numsubst;
01663       pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
01664       if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
01665          ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
01666          ast_channel_sendurl( peer, args.url );
01667       }
01668       if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
01669          if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
01670             res = 0;
01671             goto out;
01672          }
01673       }
01674       if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
01675          res = 0;
01676       } else {
01677          int digit = 0;
01678          /* Start autoservice on the other chan */
01679          res = ast_autoservice_start(chan);
01680          /* Now Stream the File */
01681          if (!res)
01682             res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
01683          if (!res) {
01684             digit = ast_waitstream(peer, AST_DIGIT_ANY);
01685          }
01686          /* Ok, done. stop autoservice */
01687          res = ast_autoservice_stop(chan);
01688          if (digit > 0 && !res)
01689             res = ast_senddigit(chan, digit, 0);
01690          else
01691             res = digit;
01692 
01693       }
01694 
01695       if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
01696          replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
01697          ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
01698          /* peer goes to the same context and extension as chan, so just copy info from chan*/
01699          ast_copy_string(peer->context, chan->context, sizeof(peer->context));
01700          ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
01701          peer->priority = chan->priority + 2;
01702          ast_pbx_start(peer);
01703          hanguptree(outgoing, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
01704          if (continue_exec)
01705             *continue_exec = 1;
01706          res = 0;
01707          goto done;
01708       }
01709 
01710       if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
01711          struct ast_app *theapp;
01712          const char *macro_result;
01713 
01714          res = ast_autoservice_start(chan);
01715          if (res) {
01716             ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
01717             res = -1;
01718          }
01719 
01720          theapp = pbx_findapp("Macro");
01721 
01722          if (theapp && !res) { /* XXX why check res here ? */
01723             /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
01724             ast_copy_string(peer->context, chan->context, sizeof(peer->context));
01725             ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
01726 
01727             replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
01728             res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
01729             ast_debug(1, "Macro exited with status %d\n", res);
01730             res = 0;
01731          } else {
01732             ast_log(LOG_ERROR, "Could not find application Macro\n");
01733             res = -1;
01734          }
01735 
01736          if (ast_autoservice_stop(chan) < 0) {
01737             ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
01738             res = -1;
01739          }
01740 
01741          if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
01742             char *macro_transfer_dest;
01743 
01744             if (!strcasecmp(macro_result, "BUSY")) {
01745                ast_copy_string(pa.status, macro_result, sizeof(pa.status));
01746                ast_set_flag64(peerflags, OPT_GO_ON);
01747                res = -1;
01748             } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
01749                ast_copy_string(pa.status, macro_result, sizeof(pa.status));
01750                ast_set_flag64(peerflags, OPT_GO_ON);
01751                res = -1;
01752             } else if (!strcasecmp(macro_result, "CONTINUE")) {
01753                /* hangup peer and keep chan alive assuming the macro has changed
01754                   the context / exten / priority or perhaps
01755                   the next priority in the current exten is desired.
01756                */
01757                ast_set_flag64(peerflags, OPT_GO_ON);
01758                res = -1;
01759             } else if (!strcasecmp(macro_result, "ABORT")) {
01760                /* Hangup both ends unless the caller has the g flag */
01761                res = -1;
01762             } else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
01763                res = -1;
01764                /* perform a transfer to a new extension */
01765                if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
01766                   replace_macro_delimiter(macro_transfer_dest);
01767                   if (!ast_parseable_goto(chan, macro_transfer_dest))
01768                      ast_set_flag64(peerflags, OPT_GO_ON);
01769                }
01770             }
01771          }
01772       }
01773 
01774       if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
01775          struct ast_app *theapp;
01776          const char *gosub_result;
01777          char *gosub_args, *gosub_argstart;
01778 
01779          res = ast_autoservice_start(chan);
01780          if (res) {
01781             ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
01782             res = -1;
01783          }
01784 
01785          theapp = pbx_findapp("Gosub");
01786 
01787          if (theapp && !res) { /* XXX why check res here ? */
01788             replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
01789 
01790             /* Set where we came from */
01791             ast_copy_string(peer->context, "app_dial_gosub_virtual_context", sizeof(peer->context));
01792             ast_copy_string(peer->exten, "s", sizeof(peer->exten));
01793             peer->priority = 0;
01794 
01795             gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
01796             if (gosub_argstart) {
01797                *gosub_argstart = 0;
01798                if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
01799                   ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
01800                   gosub_args = NULL;
01801                }
01802                *gosub_argstart = ',';
01803             } else {
01804                if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
01805                   ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
01806                   gosub_args = NULL;
01807                }
01808             }
01809 
01810             if (gosub_args) {
01811                res = pbx_exec(peer, theapp, gosub_args);
01812                if (!res) {
01813                   struct ast_pbx_args args;
01814                   /* A struct initializer fails to compile for this case ... */
01815                   memset(&args, 0, sizeof(args));
01816                   args.no_hangup_chan = 1;
01817                   ast_pbx_run_args(peer, &args);
01818                }
01819                ast_free(gosub_args);
01820                ast_debug(1, "Gosub exited with status %d\n", res);
01821             } else {
01822                ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
01823             }
01824 
01825             res = 0;
01826          } else {
01827             ast_log(LOG_ERROR, "Could not find application Gosub\n");
01828             res = -1;
01829          }
01830 
01831          if (ast_autoservice_stop(chan) < 0) {
01832             ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
01833             res = -1;
01834          }
01835 
01836          if (!res && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
01837             char *gosub_transfer_dest;
01838 
01839             if (!strcasecmp(gosub_result, "BUSY")) {
01840                ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
01841                ast_set_flag64(peerflags, OPT_GO_ON);
01842                res = -1;
01843             } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
01844                ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
01845                ast_set_flag64(peerflags, OPT_GO_ON);
01846                res = -1;
01847             } else if (!strcasecmp(gosub_result, "CONTINUE")) {
01848                /* hangup peer and keep chan alive assuming the macro has changed
01849                   the context / exten / priority or perhaps
01850                   the next priority in the current exten is desired.
01851                */
01852                ast_set_flag64(peerflags, OPT_GO_ON);
01853                res = -1;
01854             } else if (!strcasecmp(gosub_result, "ABORT")) {
01855                /* Hangup both ends unless the caller has the g flag */
01856                res = -1;
01857             } else if (!strncasecmp(gosub_result, "GOTO:", 5) && (gosub_transfer_dest = ast_strdupa(gosub_result + 5))) {
01858                res = -1;
01859                /* perform a transfer to a new extension */
01860                if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
01861                   replace_macro_delimiter(gosub_transfer_dest);
01862                   if (!ast_parseable_goto(chan, gosub_transfer_dest))
01863                      ast_set_flag64(peerflags, OPT_GO_ON);
01864                }
01865             }
01866          }
01867       }
01868 
01869       if (!res) {
01870          if (calldurationlimit > 0) {
01871             peer->whentohangup = time(NULL) + calldurationlimit;
01872          }
01873          if (!ast_strlen_zero(dtmfcalled)) {
01874             ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
01875             res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
01876          }
01877          if (!ast_strlen_zero(dtmfcalling)) {
01878             ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
01879             res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
01880          }
01881       }
01882 
01883       if (res) { /* some error */
01884          res = -1;
01885       } else {
01886          if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
01887             ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
01888          if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
01889             ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
01890          if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
01891             ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
01892          if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
01893             ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
01894          if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
01895             ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
01896          if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
01897             ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
01898          if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
01899             ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
01900          if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
01901             ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
01902          if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
01903             ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
01904          if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
01905             ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
01906          if (ast_test_flag64(peerflags, OPT_GO_ON))
01907             ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
01908 
01909          config.end_bridge_callback = end_bridge_callback;
01910          config.end_bridge_callback_data = chan;
01911          config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
01912          
01913          if (moh) {
01914             moh = 0;
01915             ast_moh_stop(chan);
01916          } else if (sentringing) {
01917             sentringing = 0;
01918             ast_indicate(chan, -1);
01919          }
01920          /* Be sure no generators are left on it */
01921          ast_deactivate_generator(chan);
01922          /* Make sure channels are compatible */
01923          res = ast_channel_make_compatible(chan, peer);
01924          if (res < 0) {
01925             ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
01926             ast_hangup(peer);
01927             res = -1;
01928             goto done;
01929          }
01930          if (opermode && !strncmp(chan->tech->type, "DAHDI", 5) && !strncmp(peer->name, "DAHDI", 5)) {
01931             /* what's this special handling for dahdi <-> dahdi ?
01932              * A: dahdi to dahdi calls are natively bridged at the kernel driver
01933              * level, so we need to ensure that this mode gets propagated
01934              * all the way down. */
01935             struct oprmode oprmode;
01936 
01937             oprmode.peer = peer;
01938             oprmode.mode = opermode;
01939 
01940             ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
01941          }
01942          res = ast_bridge_call(chan, peer, &config);
01943       }
01944 
01945       strcpy(peer->context, chan->context);
01946 
01947       if (ast_test_flag64(&opts, OPT_PEER_H) && ast_exists_extension(peer, peer->context, "h", 1, peer->cid.cid_num)) {
01948          int autoloopflag;
01949          int found;
01950          strcpy(peer->exten, "h");
01951          peer->priority = 1;
01952          autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
01953          ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);
01954 
01955          while ((res = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
01956             peer->priority++;
01957 
01958          if (found && res) {
01959             /* Something bad happened, or a hangup has been requested. */
01960             ast_debug(1, "Spawn extension (%s,%s,%d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
01961             ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
01962          }
01963          ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP);  /* set it back the way it was */
01964       }
01965       if (!ast_check_hangup(chan))
01966          chan->hangupcause = peer->hangupcause;
01967       ast_hangup(peer);
01968    }
01969 out:
01970    if (moh) {
01971       moh = 0;
01972       ast_moh_stop(chan);
01973    } else if (sentringing) {
01974       sentringing = 0;
01975       ast_indicate(chan, -1);
01976    }
01977    ast_channel_early_bridge(chan, NULL);
01978    hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
01979    pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01980    senddialendevent(chan, pa.status);
01981    ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
01982    
01983         if (ast_test_flag64(peerflags, OPT_GO_ON) && !ast_check_hangup(chan)) {
01984                 if (calldurationlimit)
01985                         chan->whentohangup = 0;
01986                 res = 0;
01987         }
01988 done:
01989    return res;
01990 }
01991 
01992 static int dial_exec(struct ast_channel *chan, void *data)
01993 {
01994    struct ast_flags64 peerflags;
01995 
01996    memset(&peerflags, 0, sizeof(peerflags));
01997 
01998    return dial_exec_full(chan, data, &peerflags, NULL);
01999 }
02000 
02001 static int retrydial_exec(struct ast_channel *chan, void *data)
02002 {
02003    char *parse;
02004    const char *context = NULL;
02005    int sleep = 0, loops = 0, res = -1;
02006    struct ast_flags64 peerflags = { 0, };
02007    AST_DECLARE_APP_ARGS(args,
02008       AST_APP_ARG(announce);
02009       AST_APP_ARG(sleep);
02010       AST_APP_ARG(retries);
02011       AST_APP_ARG(dialdata);
02012    );
02013 
02014    if (ast_strlen_zero(data)) {
02015       ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
02016       return -1;
02017    }
02018 
02019    parse = ast_strdupa(data);
02020    AST_STANDARD_APP_ARGS(args, parse);
02021 
02022    if (!ast_strlen_zero(args.sleep) && (sleep = atoi(args.sleep)))
02023       sleep *= 1000;
02024 
02025    if (!ast_strlen_zero(args.retries)) {
02026       loops = atoi(args.retries);
02027    }
02028 
02029    if (!args.dialdata) {
02030       ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
02031       goto done;
02032    }
02033 
02034    if (sleep < 1000)
02035       sleep = 10000;
02036 
02037    if (!loops)
02038       loops = -1; /* run forever */
02039 
02040    context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
02041 
02042    res = 0;
02043    while (loops) {
02044       int continue_exec;
02045 
02046       chan->data = "Retrying";
02047       if (ast_test_flag(chan, AST_FLAG_MOH))
02048          ast_moh_stop(chan);
02049 
02050       res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
02051       if (continue_exec)
02052          break;
02053 
02054       if (res == 0) {
02055          if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
02056             if (!ast_strlen_zero(args.announce)) {
02057                if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
02058                   if (!(res = ast_streamfile(chan, args.announce, chan->language)))
02059                      ast_waitstream(chan, AST_DIGIT_ANY);
02060                } else
02061                   ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
02062             }
02063             if (!res && sleep) {
02064                if (!ast_test_flag(chan, AST_FLAG_MOH))
02065                   ast_moh_start(chan, NULL, NULL);
02066                res = ast_waitfordigit(chan, sleep);
02067             }
02068          } else {
02069             if (!ast_strlen_zero(args.announce)) {
02070                if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
02071                   if (!(res = ast_streamfile(chan, args.announce, chan->language)))
02072                      res = ast_waitstream(chan, "");
02073                } else
02074                   ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
02075             }
02076             if (sleep) {
02077                if (!ast_test_flag(chan, AST_FLAG_MOH))
02078                   ast_moh_start(chan, NULL, NULL);
02079                if (!res)
02080                   res = ast_waitfordigit(chan, sleep);
02081             }
02082          }
02083       }
02084 
02085       if (res < 0)
02086          break;
02087       else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
02088          if (onedigit_goto(chan, context, (char) res, 1)) {
02089             res = 0;
02090             break;
02091          }
02092       }
02093       loops--;
02094    }
02095    if (loops == 0)
02096       res = 0;
02097    else if (res == 1)
02098       res = 0;
02099 
02100    if (ast_test_flag(chan, AST_FLAG_MOH))
02101       ast_moh_stop(chan);
02102  done:
02103    return res;
02104 }
02105 
02106 static int unload_module(void)
02107 {
02108    int res;
02109    struct ast_context *con;
02110 
02111    res = ast_unregister_application(app);
02112    res |= ast_unregister_application(rapp);
02113 
02114    if ((con = ast_context_find("app_dial_gosub_virtual_context"))) {
02115       ast_context_remove_extension2(con, "s", 1, NULL, 0);
02116       ast_context_destroy(con, "app_dial"); /* leave nothing behind */
02117    }
02118 
02119    return res;
02120 }
02121 
02122 static int load_module(void)
02123 {
02124    int res;
02125    struct ast_context *con;
02126 
02127    con = ast_context_find_or_create(NULL, NULL, "app_dial_gosub_virtual_context", "app_dial");
02128    if (!con)
02129       ast_log(LOG_ERROR, "Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create\n");
02130    else
02131       ast_add_extension2(con, 1, "s", 1, NULL, NULL, "NoOp", ast_strdup(""), ast_free_ptr, "app_dial");
02132 
02133    res = ast_register_application(app, dial_exec, synopsis, descrip);
02134    res |= ast_register_application(rapp, retrydial_exec, rsynopsis, rdescrip);
02135 
02136    return res;
02137 }
02138 
02139 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialing Application");

Generated on Thu Jul 9 13:40:17 2009 for Asterisk - the Open Source PBX by  doxygen 1.4.7