Wed Oct 28 11:53:24 2009

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 ast_callerid
Implement settings for transliteration between UTF8 caller ID names in to Ascii Caller ID's (DAHDI). Ísten ┼sklund might be transliterated into Osten Asklund or Oesten Aasklund depending upon language and person... We need automatic routines for incoming calls and static settings for our own accounts.

Class ast_generator
Add an explanation of an Asterisk generator

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

Class unistim_line
Convert to stringfields

File ael_structs.h
document this file (ael.h)

Global d_descrip
XXX Remove this application after 1.4 is relased

Global pickup_do
This application should return a result code, like PICKUPRESULT

Global eivr_comm
add deprecation debug message for X command here

File ast_expr.h
Explain this file!

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

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.

Make this optional. If this is only going to talk to 8 kHz endpoints, then it makes sense to use 8 kHz natively.

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

Move definition of MAX_CHANLIST_LEN to a proper place.

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

File chan_iax2.c
Implement musicclass settings for IAX2 devices

File chan_sip.c
Better support of forking

File chan_sip.c
VIA branch tag transaction checking

File chan_sip.c
Transaction support

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

File chan_sip.c
Fix TCP/TLS handling in dialplan, SRV records, transfers and much more

File chan_sip.c
Save TCP/TLS sessions in registry

File chan_sip.c
Add TCP/TLS information to function SIPPEER and SIPCHANINFO

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

Use known T1 for timeout (peerpoke)

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

Global authl
Move the sip_auth list to AST_LIST

Global _sip_tcp_helper_thread
XXX If there's no Content-Length or if the content-length and what we receive is not the same - we should generate an error

Global add_peer_mailboxes
document this function

Global check_auth
need a better return code here

Global check_rtp_timeout
Check video RTP keepalives

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

Global function_sippeer
Will be deprecated after 1.4

Global handle_request_info
Note: Doesn't read the duration of the DTMF. Should be fixed.

Global handle_request_invite
XXX This needs to be reviewed. YOu don't change the request URI really, you route the packet correctly instead...

Global handle_request_options
Fix handle_request_options device handling with optional authentication (this needs to be fixed in 1.4 as well)

Global initreqprep
Need to add back the VXML URL here at some point, possibly use build_string for all this junk

Global load_module
Fix this XXX This must be all wrong XXXX

Global sip_prepare_socket
document this function.

Global sip_prune_realtime
XXXX Propably needs an overhaul after removal of the devices

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 load_module
Leaking anything allocated by reload_config() ...

Global __ast_read
XXX It is possible to write a digit to the audiohook twice if the digit was originally read while the channel was in autoservice.

Global ast_write
XXX should return 0 maybe ?

File devicestate.h
Currently, when the state of a device changes, the device state provider calls one of the functions defined here to queue an object to say that the state of a device has changed. However, this does not include the new state. Another thread processes these device state change objects and calls the device state provider's callback to figure out what the new state is. It would make a lot more sense for the new state to be included in the original function call that says the state of a device has changed. However, it will take a lot of work to change this.

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

File enum.c
The service type selection needs to be redone.

Global ast_bridge_call
XXX how do we guarantee the latter ?

Global ast_feature_request_and_dial
XXX Check - this is very similar to the code in channel.c

Global builtin_blindtransfer
XXX Maybe we should have another message here instead of invalid extension XXX

Global feature_exec_app
XXX should probably return res

Global load_config
XXX var_name or app_args ?

Global park_exec_full
XXX we would like to wait on both!

Global park_exec_full
XXX Play a message XXX

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

File func_devstate.c
Delete the entry from AstDB when set to nothing like Set(DEVICE_STATE(Custom:lamp1)=)

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

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

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

File res_jabber.c
Dialplan applications need RETURN variable, like JABBERSENDSTATUS

This should really be renamed to xmpp.conf. For backwards compatibility, we need to read both files

File res_smdi.c
This module currently has its own mailbox monitoring thread. This should be converted to MWI subscriptions and just let the optional global voicemail polling thread handle it.

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

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.

File udptl.h
add doxygen documentation to this file!

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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