Since ASTERISK-26976, libSRTP 2.x can be used for sRTP. However, that change
added a private header which is not available on openSUSE for example. To
remain compatibility with very old libSRTP versions, the affected/missing
symbols AES_128_ICM and HMAC_SHA1 are defined manually.
ASTERISK-27733
Change-Id: I25c5cb8fa966043d1506ebef449e5a724412b4b6
The "ptime" SDP parameter received in a SIP response was not honoured.
Moreover, in the abscence of this "ptime" parameter, locally configured
framing was lost during response processing.
This patch systematically stores the framing information in the
ast_rtp_codecs structure, taking it from the response or from the
configuration as appropriate.
ASTERISK-27674
Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c
When HAVE_GETHOSTBYNAME_R_5 was set by the script ./configure, GCC 7.3.0 found
an unused variable. Actually, the variable was used (set to a dummy value) but
the compiler optimization might have removed that. Instead, this change ensures
that the variable 'res' is only used when it is really required.
Change-Id: Ic3ea23ccf84ac4bc2d501b514985b989030abab5
The pool cache gets in the way of finding use after free errors of memory
pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool
contents are used after free and who freed it.
To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.
Sample pjproject.conf setting:
[startup]
cache_pools=no
* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.
ASTERISK-27704
Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828
Between Asterisk 11 and Asterisk 13 there was a significant increase
in the number of AST_FRAME_NULL frames being processed by app_amd.c's
main loop. Each AST_FRAME_NULL frame was being counted as 100ms
towards the total time and silence. This may have been accurate
when app_amd.c was orginally added, but it is not in Asterisk 13.
As such the total analysis time and silence calculations were way
off effectively breaking app_amd.c
* Additional debug messages were added
* AST_FRAME_NULL are now ignored
ASTERISK-27610
Change-Id: I18aca01af98f87c1e168e6ae0d85c136d1df5ea9
* Fix --tarball-config so the option doesn't cause an error.
* Allow for missing /etc/os-release.
* Add a sleep between tarballing the coredump and removing the
output directory to allow the filesystem to settle.
Change-Id: I73e03b13087978bcc7f6bc9f45753990f82d9d77
When a line is the maximum length "\n" is found at sizeof(buf) - 2 since
the last character is actually the null terminator. In addition if a
line was exactly 8190 plus a multiple of 8192 characters long the config
parser would skip the following line.
Additionally fix comment in voicemail.conf sample config. It previously
stated that emailbody can only contain up to 512 characters which is
always wrong. The buffer is normally 8192 characters unless LOW_MEMORY
is enabled then it is 512 characters. The updated comment states that
the line can be up to 8190 or 510 characters since the line feed and
NULL terminator each use a character.
ASTERISK-26688 #close
Change-Id: I80864a0d40d2e2d8cd79d72af52a8f0a3a99c015
The transferrer's session channel was destroyed by the transferrer's
serializer thread in a race condition with the transfer target's
serializer thread during an attended transfer. The transfer target's
serializer was attempting to clean up a deferred end status on behalf of
the transferrer's channel when it should have passed the action to the
transferrer's serializer. When the transfer target's serializer lost the
race then both threads wind up trying to end the transferrer's session.
* Push the ast_sip_session_end_if_deferred() call onto the transferrer's
serializer to avoid a race condition that results in a crash. The
session_end() function that could be called by
ast_sip_session_end_if_deferred() really must be executed by the
transferrer's serializer to avoid this kind of crash.
ASTERISK-27568
Change-Id: Iacda724e7cb24d7520e49b2fd7e504aa398d7238
A few changes were made to AMI:
* Fixed "(null):" header in AMI AsyncAGIEnd event
* A mute header was added to the ConfbridgeJoin AMI event
* ConfbridgeList action's ConfbridgeList events now output all
the standard channel snapshot headers
Change-Id: I94a82a44b02c91becae08d254e9a56abba5697cf
pjproject's fmtp retrieval function failed to catch invalid fmtp attributes.
Because of this Asterisk would crash if given an SDP with an invalid fmtp
attribute.
When retrieving the format this patch now makes sure the fmtp attribute is
available. If not available it now returns an error status.
ASTERISK-27583 #close
Change-Id: I5cebe000ce2d846cae3af33b6d72c416e51caf2f
pjproject's media format parsing algorithm failed to catch invalid values.
Because of this Asterisk would crash if given an SDP with a invalid media
format description.
When parsing the media format description this patch now properly parses the
value and returns an error status if it can't successfully parse/convert the
value.
ASTERISK-27582 #close
Change-Id: I883b3a4ef85b6972397f7b56bf46c5779c55fdd6
