When we fail over to a new target we create a new transaction
and it becomes the current INVITE transaction. This does not
prevent the previous transaction from raising state changes
and causing the session to be prematurely disconnected if a
transport error occurs immediately.
This change backports a fix from PJSIP that eliminates the
incorrect state change and reduces when they would be raised
in the first place.
ASTERISK-27408
Change-Id: Id22d087591782eee31311753d11e7eca4b95ef34
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
Currently, when the app_voicemail sending VoicemailUserEntry AMI event, there's
no OldMessageCount info for default.
To check the OldMessageCount info, it required IMAP_STORAGE define, but this is
not correct.
Added OldMessageCount item as a default.
ASTERISK-27456
Change-Id: I5c71521c2d1daf8b7b161e31c34d28cca6aea4c7
More complicated direct media reinvite negotiations can result in longer
delays before direct media flows. The strictrtp learning timeout time
was too short. One log showed that the first RTP packet came in just
after three seconds.
* Increase the strictrtp learning timeout time from 1.5 to 5 seconds.
ASTERISK-27453
Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c
With Asterisk 12 (commit 866d968), 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
The definition in config_site.h and the argument to the
configure script are not necessary to disable WebRTC
support. The correct argument, --disable-libwebrtc, is
already passed.
ASTERISK-26980
Change-Id: I27da2c894f87914956a72710222e17462d8a44bc
The m4_ifblank macro is not available on CentOS 6, reverse conditionals
to allow use of m4_ifval instead. ./bootstrap.sh was run but this patch
does not result in any difference to the generated configure script.
Change-Id: I280785deb872ed8d3339d99cce63a2b54d5f1438
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
ast_category_get() has an (undocumented) implementation detail where it
tries to match the category name first by an explicit pointer comparison
and if that fails falls back to a normal match.
When initially building an ast_config during ast_config_load, this
pointer comparison can never succeed, but we will end up iterating all
categories twice. As the number of categories using a template
increases, this dual looping becomes quite expensive. So we pass a flag
to category_get_sep() indicating if a pointer match is even possible
before trying to do so, saving us a full pass over the list of current
categories.
In my tests, loading a file with 3 template categories and 12000
additional categories that use those 3 templates (this file configures
4000 PJSIP endpoints with AOR & Auth) takes 1.2 seconds. After this
change, that drops to 22ms.
Change-Id: I59b95f288e11eb6bb34f31ce4cc772136b275e4a
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
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
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
