Todo List

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 _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

File ael_structs.h

document this file (ael.h)

Global ALLOWED_METHODS

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

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 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 ?

Class ast_callerid

Implement settings for transliteration between UTF8 caller ID names in to Ascii Caller ID's (DAHDI). Östen Åsklund 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.

File ast_expr.h

Explain this file!

Global ast_feature_request_and_dial

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

Class ast_generator

Add an explanation of an Asterisk generator

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

Global builtin_blindtransfer

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

File chan_dahdi.c

Deprecate the "musiconhold" configuration option post 1.4

File chan_iax2.c

Implement musicclass settings for IAX2 devices

File chan_sip.c

Better support of forking

VIA branch tag transaction checking

Transaction support

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

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

Save TCP/TLS sessions in registry

Add TCP/TLS information to function SIPPEER and SIPCHANINFO

Global check_auth

need a better return code here

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 d_descrip

XXX Remove this application after 1.4 is relased

Global dahdi_setoption

XXX This is an abuse of the stack!!

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.

Global eivr_comm

add deprecation debug message for X command here

File enum.c

Implement a caching mechanism for multile enum lookups

• See http://bugs.digium.com/view.php?id=6739

The service type selection needs to be redone.

Global feature_exec_app

XXX should probably return res

File func_devstate.c

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

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 handle_show_settings

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

Global initreqprep

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

Global JABBER_CONFIG

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

Global load_config

XXX var_name or app_args ?

Global load_module

Fix this XXX This must be all wrong XXXX

Leaking anything allocated by reload_config() ...

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_full

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 pickup_do

This application should return a result code, like PICKUPRESULT

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 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?

Dialplan applications need RETURN variable, like JABBERSENDSTATUS

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 SAMPLE_RATE

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

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_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 SIP_TRANS_TIMEOUT

Use known T1 for timeout (peerpoke)

Global SUPPORTED_EXTENSIONS

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

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().

File udptl.h

add doxygen documentation to this file!

Class unistim_line

Convert to stringfields