Thu Oct 11 06:44:30 2012

Asterisk developer's documentation


Todo List

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).

Class bucket_list
this should be private to the container code

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

Global d_descrip
XXX Remove this application after 1.4 is relased

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 agentmonitoroutgoing_exec
XXX Needs to check option priorityjump etc etc

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

Global MAX_CHANLIST_LEN
Move definition of MAX_CHANLIST_LEN to a proper place.

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

Global function_iaxpeer
: will be removed after the 1.4 relese

File chan_sip.c
SIP over TCP

File chan_sip.c
SIP over TLS

File chan_sip.c
Better support of forking

File chan_sip.c
VIA branch tag transaction checking

File chan_sip.c
Transaction support

Global SIP_TRANS_TIMEOUT
Use known T1 for timeout (peerpoke)

Global authl
Move the sip_auth list to AST_LIST

Global check_auth
need a better return code here

Global function_sippeer
Will be deprecated after 1.4

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 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.

Global sip_handle_t38_reinvite
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 transmit_refer
Fix the transfer() dialplan function so that a transfer may fail

Global transmit_refer
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().

Global reload_config
Remove 'port' option after 1.4

Global ast_write
XXX should return 0 maybe ?

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

Global ast_bridge_call
XXX how do we guarantee the latter ?

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

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

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

Global pbx_builtin_setglobalvar
XXX overwrites data ?

Global pbx_builtin_setglobalvar
XXX watch out, leading whitespace ?

File res_adsi.c
Move app_getcpeid into this module

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

Global ast_bridge_call_thread
XXX for safety

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

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

Global feature_exec_app
XXX should probably return res

Global load_config
XXX var_name or app_args ?

Global park_exec
XXX we would like to wait on both!

Global park_exec
XXX Play a message XXX

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

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

Global SAY_INIT
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 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.

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


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