This change implements a few different generic things which were brought
on by Google Voice SIP.
1. The concept of flow transports have been introduced. These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target. These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity). When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.
2. Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.
3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module. If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.
4. Configurable outbound extension support has been added to the outbound
registration module. When set the extension will be placed in the
Supported header.
5. Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.
6. Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.
All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.
ASTERISK-27971 #close
Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
These macros have been documented as legacy for a long time but are
still used in new code because they exist. Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc
These macro's are also removed. Only ao2_container_alloc remains due to
it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
__ast_mutex_logger used the variable `canlog` without accepting it as a
argument. Replace with internal macro `log_mutex_error` which takes
canlog as the first arguement. This will prevent confusion when working
with lock.c code, many of the function declare the canlog variable and
in some cases it previously appeared to be unused.
Change-Id: I83b372cb0654c5c18eadc512f65a57fa6c2e9853
* Update the post-answer documentation and example. The Dial example was
incorrect and misleading for the post-answer subroutine useage.
* Fix note and warning paragraphs in option descriptions. They don't show
up in the wiki.
Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14
Most were in comments. A couple were in warning messages.
Pointed out by Jonathan H on the Asterisk users mailing list.
Change-Id: I6286939dff5d0a27a2758140570106f1cb351855
* The dependency ensures that res_pjproject cannot be manually unloaded
before res_rtp_asterisk.
* The dependency allows startup loading errors to report that
res_rtp_asterisk depends upon res_pjproject.
Change-Id: Icf5e7581f4ddd6189929f6174c74dd951f887377
Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push. This will help
ensure we perform the necessary cleanup upon failure.
Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d
This patch is not in the upstream pjproject and does unsafe things with
the timer->_timer_id and timer->_grp_lock values in pj_timer_entry_reset()
outside of the timer heap lock. pj_timer_entry_reset() is also called for
timers that are not about to be rescheduled in a few places.
Change-Id: I4fe0b4bc648f7be5903cf4531b94fc87275713c1
This allows us to process AO2 statistics for total objects, memory
usage, memory overhead and lock usage.
* Install refstats.py and reflocks.py into the Asterisk scripts folder.
* Enable support for reflocks.py without DEBUG_THREADS.
Steal a bit from the ao2 magic to flag when an object lock is used.
Remove 'lockobj' from reflocks.py since we can now record 'used' or
'unused' for those objects.
Add comments to explain thread safety of the 'struct __priv_data'
bitfields.
Change-Id: I84e9d679cc86d772cc97c888d9d856a17e0d3a4a
This patch sets the callerid_tag to empty string by default.
If the callerid_tag is set to NULL then the tag does not
become part of a connected line update.
For example:
Alice's tag is "Alice".
Bob's tag is empty.
Charlie's tag is "Charlie".
Alice calls Bob and then does attended transfer to Charlie.
When Alice hangs up the CONNECTEDLINE(tag) is "Alice"
on the interception routine on the Charlie's channel, but should be empty.
Ths patch also fix memory leaks if there are more then one options
"callerid", "callerid_tag", "voicemail_extension" and "contact_user"
in the pjsip.conf endpoint definition.
Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4
thread_worker_pair, set_size_data and task_pushed_data structures are
allocated with AO2 objects, passed to a taskprocessor, then released.
They never have multiple owners or use locking so AO2 only adds
overhead.
Change-Id: I2204d2615d9d952670fcb48e0a9c0dd1a6ba5036
The field used to store call arguments was not large enough to hold the
arguments string that can be constructed for 'open'. Expand it to
prevent this warning/error.
Change-Id: I514927f256481bc84df10a51b19d5b5fb1bc387e
The return status when there was no change in statsd.conf was incorrect.
This resulted in the wrong status message on the CLI when reloading the
module.
* Fixed cleanup on initial load if initializing statsd failed.
Change-Id: Id24fae75f1a7ff584a444a5680e867d989792481
ast_sendtext_data() would create an incorrect T.140 text frame which
length include the null terminator byte. It causes ultimately RTP
packets to be send with this trailing 0. The proposed fix just set the
correct length to the text frame
ASTERISK-28089
Reported by: Emmanuel BUU
Tested by: Emmanuel BUU
Change-Id: I7ab1b9ed1e21683b2b667ea0a59d9aba3c77dd96
This has no effect on startup since AST_MODULE_LOAD_FAILURE aborts
startup, but it's possible for this code to be returned on manual load
of a module after startup.
It is an error for a module to not have a load callback but this is not
a fatal system error. In this case flag the module as declined, return
AST_MODULE_LOAD_FAILURE only if a required module is broken.
Expand doxygen documentation for AST_MODULE_LOAD_*.
Change-Id: I3c030bb917f6e5a0dfd9d91491a4661b348cabf8
* Make load_module() cleanup if it failed to setup the module.
* Make unload_module() always return 0. It is silly to fail unloading if
the hook function we try to unregister was not even registered.
Change-Id: I280fc6e8ba2a7ee2588ca01d870eebaf74b4ffe6
If codec_speex fails to register a translator it would cause Asterisk to
exit instead of continue as a DECLINED module.
* Make unload_module() always return 0. It is silly to fail unloading if
any translators we try to unregister were not even registered.
Change-Id: Ia262591f68333dad17673ba7104d11c88096f51a
Nick French
George Joseph
Sean Bright
Richard Mudgett
Corey Farrell
Jenkins2
Alexei Gradinari
neutrino88