When a Contact is updated the only material change that qualify
support cares about is the underlying configuration for the AOR.
In this case we will update things with the new AOR information but
otherwise the callback to indicate the Contact has changed can be
ignored.
This is because it is only when a Contact is added or deleted that
material changes occur within the qualify support. An update can't
change the URI since it would result in a new Contact so it can be
ignored.
Change-Id: I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d
We were still getting crashes after the first fix. Somehow we receive a
non-2xx final response before we get a 200 final response. With the
failure response we had already cleaned up and destroyed some data
structures. When the unexpected 200 response comes in we crash.
* Add protection code to prevent processing another final T.38 reINVITE
response.
ASTERISK-27944
Change-Id: I8b5baba8d07fe4d63f0d7d05d3eb9a3d27d40a74
The allow_unqualified_fetch option for the sorcery realtime backend
blocked actually fetching all rows when the option is set to warn.
* Made issue a warning and actually do the request when
allow_unqualified_fetch=warn is set.
Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312
* In destroy_peer(), fixed memory leaks of lookup history strings and
qualify transactions when destroying peers.
* In destroy_peer(), fixed leaving the registerexpire scheduled callback
active when a peer is destroyed on a reload. The reload marks and sweeps
peers so any peers not explicitly configured get destroyed. Peers created
dynamically from the '*' peer will not exist until they re-register after
the reload. These destroyed peers caused memory corruption when the
registerexpire timer expired.
* Made build_peer() not schedule any callbacks on the '*' peer
(empty_eid). It is a special peer that is cloned to dynamically created
peers so it doesn't actually get involved in any message transactions.
* Made do_register_expire() remove the dundi/dpeers AstDB entry when a
peer registration expires.
* Fix deep_copy_peer() to not copy some things that cannot be copied to
the cloned peer structure. Timers, message transactions, and lookup
history are specific to a peer instance.
* Made set_config() lock around processing the mappings configuration.
* Reordered unload_module() to handle load_module() declining the load due
to error.
Change-Id: Ib846b2b60d027f3a2c2b3b563d9a83a357dce1d6
* Fixed a typo in the name of the REGREQ frame decode string array.
* Fixed off by one range check indexing into the frame decode string
array.
* Removed some unneeded casts associated with the decode string array.
Change-Id: I77435e81cd284bab6209d545919bf236ad7933c2
Compiling without SRTP support installed resulted in some unused variable
warnings. These warnings also showed that the srtp variable was obtained
and passed around some functions but not really used even when a system
has SRTP installed.
Change-Id: I6daad34be3e89b19adef6e2fbe738018975155fc
If the review to be tested is in a project with restricted access,
we need to use the jenkins user's gerrit https credentials when we
do the checkout or the checkout will fail.
Change-Id: I9dc9994763c5ebfeb9f1cff60fb53f6902b7fd5f
Fix not show the error
"/bin/sh: /etc/os-release: No such file or directory" when the command
'make config' is run in a System without systemv.
The instruction 'make config' pre execute the syntax
"$(shell . /etc/os-release && echo $$ID)" to identified if system is a
Slackware and Opensuse.
This change prevent show the message and is send to the /dev/null
Change-Id: I7f43e281a8d9405b2519fc653de82d9b8b645fdf
If in the initial sdp the caller doesn't include the line
a=rtcp-mux
Then asterisk shoud not include rtcp-mux in the response regardless
of rtcp-mux being enabled on the endpoint
ASTERISK-28007 #close
Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7
sip_options_get_endpoint_state_compositor_state leaked a reference to
the first available endpoint state compositor that was found.
Change-Id: Idb6be19f7219b6eed1dfb19c1e740dd40cb3fdc7
This change adds the "party" parameter to the Remote-Party-ID header
which indicates which party the header information is applicable
to. In Asterisk this is determined on whether we are the calling
or called party. This is added to improve interoperability with some
implementations.
ASTERISK-28006
Change-Id: I1eec3e377ffff8633b5c1dd59a05e9533122cfca
When a conference contained a mixture of audio/video and audio-only
users, a NOTICE message would pop up stating there are no joint
capabilities between streams. This happens because streams can never be
removed, but they can be in a REMOVED state. If we have the scenario
where user A joins with audio/video, user B joins with audio-only, and
user C joins with audio/video, then user A leaves, the message would
be triggered. That removed stream is still in the SDP, but Asterisk
would pass it through, causing it to be seen as a ulaw stream. A check
has been added for removed streams, setting their status to REMOVED when
handling negotiated SDPs.
Also addressed an issue where user A joins, then user B joins but does
not receive video until much later. Full frames were not being sent,
causing some PLI from the browser. Because the video was flowing in one
direction, the browser sets the SSRC to 1, but Asterisk was dropping the
PLI because of that. Added a check to see if the SSRC is 1 or not, which
sends full frames and allows video to flow between user A and user B.
This should only happen when dealing with PSFB or FUR, and in the case
of PSFB, only for PLI.
ASTERISK-27398
Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e
When a call leaves a queue on leaveempty condition, QUEUESTATUS
must be set to LEAVEEMPTY, no matter whether Queue was executed with or
without the "c" (continue) option.
The regression was introduced in the fix for ASTERISK_25665.
The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was
overwritten in case when "c" is set, regardless of what was the cause
for leaving the queue.
ASTERISK-27973 #close
Reported-by: Valentin Safonov
Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c
It is valid for a config file to be empty or contain only comments, but
not valid for a config value to be set when no uncommented context
exists. This caused an error to be loged numerous times during start
when loading the default pjsip.conf.
Change-Id: Icf3b0d69b4ecb6e935eecd43c99ed8b32a5a1cf6
Enable coverage with `./tests/CI/buildAsterisk.sh --coverage`. This
will cause Asterisk to be compiled with coverage support. It also
initializes 'before' coverage data for all sources. Accept
--tested-only to disable modules which are not run by any test.
Enabling coverage also sets tested-only true by default. To build
everything with coverage enabled use `--coverage --tested-only=0`.
./tests/CI/processCoverage.sh is used to process the coverage and
generate HTML reports.
Fix utils/check_expr2 which failed to compiled with coverage enabled.
Add status output 5 times per stage of astobj2_test_perf to ensure
remote CLI does not timeout when compiled with coverage. Remote CLI
disconnects if no output is received for 60 seconds. When coverage is
enabled it takes about 70 seconds for my laptop to run the stages of
this test, so with the change a message is printed every 14 seconds.
Change-Id: I890f7d5665087426ad7d3e363187691b9afc2222
When the stasis cache is used a hash is calculated for
retrieving or inserting messages. This change calculates
a hash when the message type is initialized that is then
used each time needed. This ensures that the hash is
calculated only once for the message type.
Change-Id: I4fe6bfdafb55bf5c322dd313fbd8c32cce73ef37
The authors of PJProject undef s_addr because of some issue in Microsoft
Windows. However in Oracle Solaris, s_addr is not a structure member, but
defined to map to the real structure member.
Updates the patch from ASTERISK_20366
ASTERISK-27997
Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11
* Don't include pjlib.h twice in res_pjsip.h
* Consistently use #include <> form for pjproject includes.
(pjsip.h and pjlib.h)
Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7
If asterisk offer an endpoint with SRTP and that endpoint respond
with non srtp, in that case channel(rtp,secure,audio) reply wrong
status.
Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key:
Currently this flag has being set redundantly. In either case identical
or different remote_key this flag has being set. So if we
don't set it while we receive identical remote_key or non SRTP SDP
response then we can take decision of srtp use by using that flag.
ASTERISK-27999
Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7
The script configure from Teluu expects shared libraries (.so) in a subfolder
called 'lib', when --with-xyz=PATH is specified. However for OpenSSL, the
default location is the root of the source folder = PATH. Furthermore, Asterisk
supports both, 'lib' and root. For consistency and because Asterisk is using
(only) OpenSSL in PJProject, it is enhanced to support both locations, just
like Asterisk.
ASTERISK-27995
Change-Id: I8eb916a88b6b8c22e29bb40bee8faaca6c73406f
George Joseph
Joshua Colp
Richard Mudgett
Kirsty Tyerman
Kevin Harwell
Rodrigo Ramírez Norambuena
Torrey Searle
Corey Farrell
Jenkins2
Ben Ford
Ivan Poddubny
Alexander Traud
Salah Ahmed