Global __ast_read (struct ast_channel *chan, int dropaudio, int dropnondefault)

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 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.

Page AMI - The Manager Interface

include missing manager txt

Class ao2_container

Linking and unlinking container objects is typically expensive, as it involves a malloc()/free() of a small object which is very inefficient. To optimize this, we can allocate larger arrays of container nodes 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 anyway).

File app_voicemail.c

This module requires res_adsi to load. This needs to be optional during compilation.

This file is now almost impossible to work with, due to all #ifdefs. Feels like the database code before realtime. Someone - please come up with a plan to clean this up.

Page ARA - The Asterisk Realtime Interface

include missing realtime txt

include missing extconfig txt

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_cel_fabricate_channel_from_event (const struct ast_event *event)

This function is very expensive, especially given that some CEL backends use it on every CEL event. This function really needs to go away at some point.

Global ast_channel_connected_line_macro (struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const void *connected_info, int is_caller, int frame)

Have multiple return codes based on the MACRO_RESULT

Make constants so that caller and frame can be more expressive than just '1' and '0'

Global ast_channel_connected_line_macro (struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const void *connected_info, int is_caller, int frame)

Have multiple return codes based on the MACRO_RESULT

Make constants so that caller and frame can be more expressive than just '1' and '0'

Global ast_channel_redirecting_macro (struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const void *redirecting_info, int is_caller, int is_frame)

Have multiple return codes based on the MACRO_RESULT

Make constants so that caller and frame can be more expressive than just '1' and '0'

File ast_expr.h

Explain this file!

Class ast_generator

Add an explanation of an Asterisk generator

Class ast_party_caller

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.

Global ast_party_name::char_set

Start using the party name character set value. Not currently used.

Global ast_pbx_outgoing_exten_predial (const char *type, struct ast_format_cap *cap, const char *addr, int timeout, const char *context, const char *exten, int priority, int *reason, int synchronous, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **locked_channel, int early_media, const struct ast_assigned_ids *assignedids, const char *predial_callee)

XXX Not good. The channel name is not unique if more than one originate fails at a time.

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_say_date_with_format_he (struct ast_channel *chan, time_t t, const char *ints, const char *lang, const char *format, const char *tzone)

A "ha" is missing in the standard date format, before the 'd'.
The numbers of 3000–19000 are not handled well

Global ast_security_event_severity

Do we need any more levels here?

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 bridge_agent_hold_push (struct ast_bridge *self, struct ast_bridge_channel *bridge_channel, struct ast_bridge_channel *swap)

XXX the login probation time should be only if it is needed.

Page CDR - Call Data Records and billing

include missing cdrdriver txt

File cel.h

TODO: There some event types that have been defined here, but are not yet used anywhere in the code. It would be really awesome if someone went through and had Asterisk generate these events where it is appropriate to do so. The defined, but unused events are: CONF_ENTER, CONF_EXIT, CONF_START, CONF_END, 3WAY_START, 3WAY_END, TRANSFER, and HOOKFLASH.

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, const char *uri, enum xmittype reliable)

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 ast_sockaddr *addr, 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 dahdi_answer (struct ast_channel *ast)

XXX this is redundantly set by the analog and PRI submodules!

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 do_forward (struct chanlist *o, struct cause_args *num, struct ast_flags64 *peerflags, int single, int caller_entertained, int *to, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)

eventually this function should be intergrated into and replaced by ast_call_forward()

File enum.c

Implement a caching mechanism for multile enum lookups

See https://issues.asterisk.org/view.php?id=6739

The service type selection needs to be redone.

Global FRAME_ENDED

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 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_invite (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *recount, const 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_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, const char *const explicit_uri)

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

Global load_module (void)

Leaking anything allocated by reload_config() ...

Don't error or warn on a default install. If the config is default we should not attempt to connect to a server. -lathama

Global main (int argc, char *argv[])

Document these options

Global mansession_session::addr

XXX need to document which fields it is protecting

Global MAX_CHANLIST_LEN

Move definition of MAX_CHANLIST_LEN to a proper place.

Global MAX_RECALC

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 mgcp_subchannel::cxident [80]

FIXME txident is replaced by rqnt_ident in endpoint. This should be obsoleted

Global MSG_MOREDATA

XXX MSG_MOREDATA should go to a header file.

Global OBJ_SEARCH_MASK

Eventually OBJ_SEARCH_MASK will shrink to a two bit field when the codebase is made to use the search field values as a field instead of independent bits.

Global parse_moved_contact (struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward)

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 *chan, const char *data)

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

Global pbx_dial_reason (enum ast_dial_result dial_result, int cause)

XXX The old originate reasons need to be trashed and replaced with normal disconnect cause codes if the call was not answered. The internal consumers of the reason values would also need to be updated: app_originate, call files, and AMI OriginateResponse.

Global peer_iphash_cb (const void *obj, const int flags)

Find a better hashing function

Global realtime_ldap_base_ap (unsigned int *entries_count_ptr, const char *basedn, const char *table_name, const struct ast_variable *fields)

XXX The interactions with base_var and append_var may cause a memory leak of base_var nodes. Also the append_var list and base_var list may get cross linked.

Global reload (void)

XXX here we should sort rules according to the same order we have in pbx.c so we have the same matching behaviour.

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_calendar.c

Support responding to a meeting invite

Support writing attendees

File res_security_log.c

Make informational security events optional

Escape quotes in string payload IE contents

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 scan_service (const char *fn, time_t now)

XXX There is some odd delayed duplicate servicing of call files going on. We need to suppress the error message if the file does not exist as a result.

Global SCHED_MULTITHREADED

XXX The IAX2 channel driver needs its native bridge code converted to the new bridge technology scheme.

Global set_address_from_contact (struct sip_pvt *pvt)

We need to save the TRANSPORT here too

Global set_destination (struct sip_pvt *p, const char *uri)

XXX If we have sip_cfg.srvlookup 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 CHANNEL function

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)

File streamplayer.c

Update this application to be able to connect to a stream via HTTP, since that is the #1 most requested feature, and it would be quite useful. A lot of people think that is what this is for and email me when it does not work. :)

Global tdd_gen_holdtone (unsigned char *outbuf)

How big should this be?

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.

File test_event.c

API Calls not yet touched by a test: XXX TODO

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

Page Video support in Asterisk

include missing video txt

Global wait_for_answer (struct queue_ent *qe, struct callattempt *outgoing, int *to, char *digit, int prebusies, int caller_disconnect, int forwardsallowed)

eventually all call forward logic should be intergerated into and replaced by ast_call_forward()