With Asterisk 12 (commit 866d9681), the default of "icesupport" changed to
- "yes" in the module "res_rtp_asterisk" and
- "no" in the module "chan_sip".
The latter was reflected in the sample configuration file for "sip.conf". The
former did not make it into "rtp.conf.sample".

ASTERISK-20643

Change-Id: I2a2e0a900455d0767a99ea576e30adc6d7608a36

chan_skinny creates a new thread for each new session.  In trying
to be a good cleanup citizen, the threads are joinable and the
unload_module function does a pthread_cancel() and a pthread_join()
on any sessions that are active at that time.  This has an
unintended side effect though. Since you can call pthread_join on a
thread that's already terminated, pthreads keeps the thread's
storage around until you explicitly call pthread_join (or
pthread_detach()).   Since only the module_unload function was
calling pthread_join, and even then only on the ones active at the
tme, the storage for every thread/session ever created sticks
around until asterisk exits.

* A thread can detach itself so the session_destroy() function
  now calls pthread_detach() just before it frees the session
  memory allocation.  The module_unload function still takes care
  of the ones that are still active should the module be unloaded.

ASTERISK-27452
Reported by: Juan Sacco

Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd
(cherry picked from commit 8f5dff54)

When starting Asterisk in the foreground, there is a perceptible delay
when loading modules that use the ACO and sorcery config frameworks.
For example, a lightly configured res_pjsip took 853ms to load on my
VM.

I tracked down the slowness to the XPath queries used to associate the
relevant documentation with the config options. One improvement was
adding a call to xmlXPathOrderDocElems after loading an XML document.
From the libxml2 docs:

  Call this routine to speed up XPath computation on static documents.

The second change was to remove recursive descent and wildcard
operators from the XPath queries. After these changes, res_pjsip takes
85ms to load on my VM and there is no longer a perceptible delay when
starting Asterisk in the foreground.

Change-Id: I45d457f1580e26bf5a2b0dab16e8e9ae46dcbd82

This change makes the presence of the GMIME_MAJOR_VERSION
definition optional, as not all versions of gmime actually
define it.

ASTERISK-27454

Change-Id: I01d99590045971ed6787899147170a5954077238

Change-Id: I530c0a72f965437acef6a9a4fbfe5c487f078b65

The `pwd` parameter to AC_CONFIG_AUX_DIR is unnecessary, the default
value is $srcdir.

Additionally remove the AC_REVISION call.  It only added a comment and
is pointless without SVN tag replacements.

Change-Id: I99299a3217f095bddcb2edefb3b9af0ab147bc29

Previous commits maintained compatibility with older remote console
clients as well as maintaining all API's.

Remove the following compatibility code:
* ast_cli_generatornummatches.
* Remote command "_command nummatches".
* Sorting / duplicate removal by remote console.

Change-Id: I59e6ce94fa57ae564888442049695f7e46746437

When a format has no pre-recorded sound files, Asterisk has to transcode between
formats. For this, Asterisk has a fixed translation table. If the pre-recorded
sound files are not available in the same sample rate, Asterisk has not only to
transcode but also to resample.

Asterisk has pre-recorded files for SLN (8000 kHz) and SLN16 (16000 kHz).
However before this change, Asterisk did not take the sample rate into account,
because the translation paths to SLN and SLN16 got the same score/weight in the
table. Consequently, you might have got narrow-band audio with siren14, speex32,
silk24, and silk12 although those are (ultra) wide-band audio codecs.

With this change, the distance in sample-rates is taken into account. Now on the
Command-Line interface (CLI) 'core show channels', you should see:
(slin@16000)->(slin@32000)->(speex@32000).

ASTERISK-23735
Reported by: Richard Kenner

Change-Id: I9448295c1978be26f8633b6066395e7bbbe2e213

The patch for ASTERISK_24560 inverted a test checking if the bridge name
is being updated to a different name.

* Fix the test to return "Changing bridge name is not implemented" when
someone attempts to change the bridge name.

ASTERISK-27445

Change-Id: I4b70bf08b0e02e016108b077ff75b345dec12fc9

ASTERISK-24662

Change-Id: I3822956984292c99c48bca8e97807e498ccc0e88

Change-Id: Id2899331fe05d1909a862ea879742879d086bc64

Although the default value of defaultbranch is master I'm adding it
anyways.  This way when new major branches are being created the value
can be updated instead of having to remember the name of the key.

Change-Id: I3db009217c5ae399fb84bee95076f4dbb7fa52d2

ASTERISK-27353 #close

Reported by: Marco Giordani

Change-Id: I455096bd7da016b871afe09af86067c2c7c9f33f

If a 183 with sdp response is receive without a To tag the sdp is not
negotiated. According to RFC 3261 section 12.1.2 while a To tag is required,
the client needs to still be able to handle the missing tag case for
backwards compatibility.

This patch, accepted by and applied to pjproject, makes it so if an incoming
180/183 with SDP comes in without a To tag it gets appropriately handled.

ASTERISK-27442 #close

Change-Id: Ic9d6b01e05e8f4874eebbd7adfe05d932025d203

Previously, Asterisk sent srflx only when configured exclusively for IPv4. Now,
srflx is gathered and sent via SDP, even when Asterisk is enabled for
Dual Stack (IPv4+IPv6) and an IPv4 interface is available/used.

ASTERISK-27437

Change-Id: Ie07d8e2bfa7b6fe06fcdc73d390a7a9a4d8c0bc1