Thu Apr 28 2011 17:14:04

Asterisk developer's documentation


Todo List

Global __ast_read (struct ast_channel *chan, int dropaudio)
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_reliable_xmit (struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod)
According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited

Global _sip_tcp_helper_thread (struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session)
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 (struct sip_peer *peer, const char *value)
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 if 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).

File app_originate.c
Make a way to be able to set variables (and functions) on the outbound channel, similar to the Variable headers for the AMI Originate, and the Set options for call files.

Global ast_audiohook_move_by_source (struct ast_channel *old_chan, struct ast_channel *new_chan, const char *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 (struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config)
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!

Class ast_generator
Add an explanation of an Asterisk generator

Global ast_rtcp_calc_interval (struct ast_rtp *rtp)
XXX Do a more reasonable calculation on this one Look in RFC 3550 Section A.7 for an example

Global ast_write (struct ast_channel *chan, struct ast_frame *frame)
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

File bridge_softmix.c
This bridge operates in 8 kHz mode unless a define is uncommented. This needs to be improved so the bridge moves between the dominant codec as needed depending on channels present in the bridge and transcoding capabilities.

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_iax2.c
Implement musicclass settings for IAX2 devices

File chan_sip.c

Better support of forking

VIA branch tag transaction checking

Transaction support

Global check_auth (struct sip_pvt *p, struct sip_request *req, const char *username, const char *secret, const char *md5secret, int sipmethod, char *uri, enum xmittype reliable, int ignore)

need a better return code here

need a better return code here

Global check_rtp_timeout (struct sip_pvt *dialog, time_t t)
Check video RTP keepalives

Global create_addr (struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin, int newdialog)
Fix this function. When we ask for SRV, we should check all transports In the future, we should first check NAPTR to find out transport preference

Global d_app
XXX Remove this application after 1.4 is relased

Global dahdi_setoption (struct ast_channel *chan, int option, void *data, int datalen)
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 dialog_needdestroy (void *dialogobj, void *arg, int flags)
Re-work this to improve efficiency. Currently, this function is called on _every_ dialog after processing _every_ incoming SIP/UDP packet, or potentially even more often when the scheduler has entries to run.

Global eivr_comm (struct ast_channel *chan, struct ivr_localuser *u, int *eivr_events_fd, int *eivr_commands_fd, int *eivr_errors_fd, const struct ast_str *args, const struct ast_flags flags)
add deprecation debug message for X command here

File enum.c

Implement a caching mechanism for multile enum lookups

The service type selection needs to be redone.

Global feature_exec_app (struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data)
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 (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
Will be deprecated after 1.4

Global get_rdnis (struct sip_pvt *p, struct sip_request *oreq)
This function does not take user-parameters into consideration. First look for @, then start looking for ; to find uri-parameters.

Global get_transport_pvt (struct sip_pvt *p)
Fix this function to work correctly

Global handle_redirect (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
It would be nice to be able to redirect 2 channels at the same time like you can with AMI redirect. However, it is not possible to acquire two channels without the potential for a deadlock with how ast_channel structs are managed today. Once ast_channel is a refcounted object, this command will be able to support that.

Global handle_request_info (struct sip_pvt *p, struct sip_request *req)
Note: Doesn't read the duration of the DTMF. Should be fixed.

Global handle_request_invite (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e, int *nounlock)
XXX This needs to be reviewed. YOu don't change the request URI really, you route the packet correctly instead...

Global handle_request_options (struct sip_pvt *p, struct sip_request *req)
Fix handle_request_options device handling with optional authentication (this needs to be fixed in 1.4 as well)

Global handle_show_settings (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
we could check musiconhold, voicemail, smdi, adsi, queues

Global initreqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod)
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 (void)
XXX var_name or app_args ?

Global load_module (void)

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 (struct ast_channel *chan, void *data, struct ast_parkinglot *parkinglot)

XXX we would like to wait on both!

XXX Play a message XXX

Global parse_moved_contact (struct sip_pvt *p, struct sip_request *req)
XXX Doesn't redirect over TLS on sips: uri's. If we get a redirect to a SIPS: uri, this needs to be going back to the dialplan (this is a request for a secure signalling path). Note that transport=tls is deprecated, but we need to support it on incoming requests.

Global parse_register_contact (struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
Check NAPTR/SRV if we have not got a port in the URI

Global pbx_builtin_importvar (struct ast_channel *, void *)
XXX should do !ast_strlen_zero(..) of the args ?

Global pickup_do (struct ast_channel *chan, struct ast_channel *target)
This application should return a result code, like PICKUPRESULT

Global powerof (unsigned int d)
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 register_verify (struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req, char *uri)

XXX here too we interpret a missing as a name-only URI, whereas the RFC says this is a domain-only uri.

OEJ Remove this - there's never RTP in a REGISTER dialog...

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_agi.c
Convert the rest of the AGI commands over to XML documentation

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_INIT (x)
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 set_address_from_contact (struct sip_pvt *pvt)
We need to save the TRANSPORT here too

Global set_destination (struct sip_pvt *p, char *uri)
XXX If we have use_dns on, then look for NAPTR/SRV, otherwise, just look for A records

Page SIP TCP and TLS support

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

Save TCP/TLS sessions in registry If someone registers a SIPS uri, this forces us to set up a TLS connection back.

Add TCP/TLS information to function SIPPEER and SIPCHANINFO

If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP. The tcpbindaddr config option should only be used to open ADDITIONAL ports So we should propably go back to bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP if tlsenable=yes, open TLS port (provided we also have cert) tcpbindaddr = extra address for additional TCP connections tlsbindaddr = extra address for additional TCP/TLS connections udpbindaddr = extra address for additional UDP connections These three options should take multiple IP/port pairs Note: Since opening additional listen sockets is a *new* feature we do not have today the XXXbindaddr options needs to be disabled until we have support for it

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

;transport=tls was deprecated in RFC3261 and should not be used at all. See section 26.2.2.

Since we have had multidomain support in Asterisk for quite a while, we need to support multiple domains in our TLS implementation, meaning one socket and one cert per domain

Selection of transport for a request needs to be done after we've parsed all route headers, also considering outbound proxy options. First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA) Intermediate requests: Outboundproxy(only when forced), routes, contact/uri DNS naptr support is crucial. A SIP uri might lead to a TLS connection. Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)

Default transports are set to UDP, which cause the wrong behaviour when contacting remote devices directly from the dialplan. UDP is only a fallback if no other method works, in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the MTU (like INIVTE with video, audio and RTT) TCP should be preferred.

re-evaluate the transport= setting in sip.conf. This is right now not well thought of. If a device in sip.conf contacts us via TCP, we should not switch transport, even if udp is the configured first transport.

Be prepared for one outbound and another incoming socket per pvt. This applies specially to communication with other peers (proxies).

If the message is smaller than the given Content-length, the request should get a 400 Bad request message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)

Global sip_prepare_socket (struct sip_pvt *p)

Get socket for dialog, prepare if needed, and return file handle

Check this... This might be wrong, depending on the proxy configuration If proxy is in "force" mode its correct.

Global sip_prune_realtime (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
XXXX Propably needs an overhaul after removal of the devices

Global sip_sipredirect (struct sip_pvt *p, const char *dest)
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 tdd_gen_holdtone (unsigned char *outbuf)
How big should this be???

Global terminate_uri (char *uri)
As this function does not support user-parameters, it's considered broken and needs fixing.

Global transmit_refer (struct sip_pvt *p, const char *dest)

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