Mon Mar 31 17:30:13 2014

Asterisk developer's documentation


Todo List

Global agentmonitoroutgoing_exec
XXX Needs to check option priorityjump etc etc

Class ao2_container
Linking and unlink objects is typically expensive, as it involves a malloc() of a small object which is very inefficient. To optimize this, we allocate larger arrays of bucket_list's when we run out of them, and then manage our own freelist. This will be more efficient as we can do the freelist management while we hold the lock (that we need anyways).

Global app_random
The Random() app should be removed from trunk following the release of 1.4

Global ast_audiohook_move_by_source

Currently only the first audiohook of a specific source found will be moved. We should add the capability to move multiple audiohooks from a single source as well.

Global ast_bridge_call
XXX how do we guarantee the latter ?

Global ast_bridge_call_thread
XXX for safety

Global ast_rtcp_calc_interval
XXX Do a more reasonable calculation on this one Look in RFC 3550 Section A.7 for an example

Global ast_write
XXX should return 0 maybe ?

Page Asterisk Language Syntaxes supported

Note that in future, we need to move to a model where we can differentiate further - e.g. between en_US & en_UK

Global authl
Move the sip_auth list to AST_LIST

Class bucket_list
this should be private to the container code

Global BUF_SIZE
Check this buf size estimate, it may be totally wrong for large frame video

File chan_dahdi.c
Deprecate the "musiconhold" configuration option post 1.4

File chan_sip.c

SIP over TCP

SIP over TLS

Better support of forking

VIA branch tag transaction checking

Transaction support

Global check_auth

need a better return code here

need a better return code here

Global d_descrip
XXX Remove this application after 1.4 is relased

Global dahdi_setoption
XXX This is an abuse of the stack!!

Global do_parking_thread

XXX Maybe we could do something with packets, like dial "0" for operator or something XXX

XXX Ick: jumping into an else statement??? XXX

File enum.c
Implement a caching mechanism for multile enum lookups

Global feature_exec_app
XXX should probably return res

File fskmodem.h
Translate Emiliano Zapata's spanish comments to english, please.

Global function_iaxpeer
: will be removed after the 1.4 relese

Global function_sippeer
Will be deprecated after 1.4

Global handle_show_settings
we could check musiconhold, voicemail, smdi, adsi, queues

Global load_config
XXX var_name or app_args ?

Global MAX_CHANLIST_LEN
Move definition of MAX_CHANLIST_LEN to a proper place.

Global mgcp_subchannel::cxident [80]
FIXME txident is replaced by rqnt_ident in endpoint. This should be obsoleted

Global park_exec

XXX we would like to wait on both!

XXX Play a message XXX

Global pbx_builtin_importvar
XXX should do !ast_strlen_zero(..) of the args ?

Global pbx_builtin_setglobalvar

XXX overwrites data ?

XXX watch out, leading whitespace ?

Global powerof
TODO: sample frames for each supported input format. We build this on the fly, by taking an SLIN frame and using the existing converter to play with it.

Global realtime_peer
Consider adding check of port address when matching here to follow the same algorithm as for static peers. Will we break anything by adding that?

Global reload_config
Remove 'port' option after 1.4

File res_adsi.c

Move app_getcpeid into this module

Create a core layer so that app_voicemail does not require res_adsi to load

File res_jabber.c

If you unload this module, chan_gtalk/jingle will be dead. How do we handle that?

If you have TLS, you can't unload this module. See bug #9738. This needs to be fixed, but the bug is in the unmantained Iksemel library

Global say_stub
XXX As the conversion from the old implementation of say.c to the new implementation will be completed, and the API suitably reworked by removing redundant functions and/or arguments, this mechanism may be reverted back to pure static functions, if needed.

Global sip_handle_t38_reinvite

Make sure we don't destroy the call if we can't handle the re-invite. Nothing should be changed until we have processed the SDP and know that we can handle it.

check if this is not set earlier when setting up the PVT. If not maybe it should move there.

Global sip_sipredirect
Fix this function so that we wait for reply to the REFER and react to errors, denials or other issues the other end might have.

Global SIP_TRANS_TIMEOUT
Use known T1 for timeout (peerpoke)

Global transmit_refer

Fix the transfer() dialplan function so that a transfer may fail

In theory, we should hang around and wait for a reply, before returning to the dial plan here. Don't know really how that would affect the transfer() app or the pbx, but, well, to make this useful we should have a STATUS code on transfer().


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