if asterisk offer T38 SDP with none error correction scheme and
the endpoint respond with redundancy EC scheme, asterisk switch
to that mode. Since we configure the endpoint as none EC mode
we should not switch to any other mode except none.
following logic implemented in code.
1. If asterisk offer none, and anything except none in answer
will be ignored.
2. If asterisk offer fec, answer with fec, redundancy and none will
be accepted.
3. If asterisk offer redundancy, answer with redundancy and none
will be accepted.
ASTERISK-28621
Change-Id: I343c62253ea4c8b7ee17abbfb377a4d484a14b19
If the name of a peer is known and a SIP request is sent using that
peer's name, the address of the peer will change even if the request
fails the authentication challenge. This means that an endpoint can
be altered and even rendered unusuable, even if it was in a working
state previously. This can only occur when the nat option is set to the
default, or auto_force_rport.
This change checks the result of authentication first to ensure it is
successful before setting the address and the nat option.
ASTERISK-28589 #close
Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df
If an AMI user without the "system" authorization calls the
Originate AMI command with the Originate application,
the second Originate could run the "System" command.
Action: Originate
Channel: Local/1111
Application: Originate
Data: Local/2222,app,System,touch /tmp/owned
If the "system" authorization isn't set, we now block the
Originate app as well as the System, Exec, etc. apps.
ASTERISK-28580
Reported by: Eliel Sardañons
Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa
The original clone depth of 10 was causing the need to rebase
changes whose parent was older than the 10 commits. The clone
depth has been increased to 100.
Workspace cleanup was only happening for successful builds which
wasn't enough to keep the 8G workspace in-memory drives on the
docker slaves from filling up. Now the workspaces are cleaned up
after every build regardless of success/failure. If you need to
preserve builds temporarily, you can log into Jenkins/Manage
Jenkins/Configure System and change the CLEANUP_WS_* environment
variable for the job type you're troubleshooting to "FALSE".
Change-Id: I0d7366e87cea714e5dbc9488caf718802fce75ca
Fixes: error: ‘domain_name’ may be used uninitialized in this function
Found with gcc (Ubuntu 9.2.1-9ubuntu2) 9.2.1 20191008
Change-Id: I44413b49ea1205aa25538142161deb73883c79e8
ASTERISK_28143 attempted to fix an issue where calls with no audio would never
timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its
calculations. Unfortunately these frames seem to show up at irregular time
intervals. This resulted in app_amd returning prematurely most of the time.
* Removed AST_FRAME_NULL from the calculations
* Added a check to see how much time has actually passed since app_amd began
ASTERISK-28608
Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42
Upon a short PRI disconnection, libpri may maintain Q.921 layer 'up' and
may thus not send PRI_EVENT_DCHAN_DOWN / PRI_EVENT_DCHAN_UP events.
If pri_event_alarm() clears DCHAN_UP status bit upon alarm detection
and no Q.921 reconnection sequence occurs, chan_dahdi will keep
seeing span status "Down" at the end of alarm.
This patch modifies pri_event_alarm() in order to keep DCHAN_UP bit
unchanged. libpri will send a PRI_EVENT_DCHAN_DOWN event if it detects
a disconnection of Q.921 layer and this will clear DCHAN_UP if required.
ASTERISK-28615
Change-Id: Ibe27df4971fd4c82cc6850020bce4a8b2692c996
The current code sets alert levels starting from index 1.
Need to set on whole pool starting from index 0.
Change-Id: I5decbb43160954fb9a512f04302637fc666b6f5d
OpenSSL can not tolerate if the packet sent out does not
match the length that it provided to the sender. This change
lies and says that each time the full packet was sent. If
a problem does occur then a retransmission will occur as
appropriate.
ASTERISK-28576
Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449
If the last line of a file does not have a terminating EOL sequence, we
potentially add garbage to the value returned from the FILE() function.
There is no overflow potential here as we are reading from a buffer of a
known size, we are just reading too much of it.
ASTERISK-26481 #close
Change-Id: I50dd4fcf416fb3c83150040a1a79a59d9eb1ae01
MOH continues to play to a channel if that channel was on hold prior to
entering a softmix bridge. MOH will not stop even if the original "holder"
attempts an unhold.
For the most part a softmix bridge ignores holds, so a participating channel
shouldn't join while on hold. This patch checks to see if the channel joining
the softmix bridge is currently on hold. If so then it indicates an unhold.
ASTERISK-28618
Change-Id: I66ccd4efc80f5b4c3dd68186b379eb442916392b
This patch fixes several issues reported by the lgtm code analysis tool:
https://lgtm.com/projects/g/asterisk/asterisk
Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:
* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards
Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25
When user wants to send json data, the default Content-Type header
is incorect (application/x-www-form-urlencoded). This patch allows
to set any custom headers so the Content-Type header can be
overriden. User can set multiple headers by multiple calls of
curlopt(). This approach is not consistent with other parameters,
but is more readable in dialplan than one call with multiple
headers.
ASTERISK-28613
Change-Id: I4dd68c3f4e25362ef941d73a3861f58348dcfbf9
resource_events:stasis_app_message_handler() was locking the session,
then attempting to determine if the app had debug enabled which
locked the app_registry container. res_stasis:__stasis_app_register
was locking the app_registry container then calling app_update
which caused app_handler (which locks the session) to run.
The result was a deadlock.
* Updated resource_events:stasis_app_message_handler() to determine
if debug was set (which locks the app_registry) before obtaining the
session lock.
* Updated res_stasis:__stasis_app_register to release the app_registry
container lock before calling app_update (which locks the sesison).
ASTERISK-28423
Reported by Ross Beer
Change-Id: I58c69d08cb372852a63933608e4d6c3e456247b4
There exists a scenario where a thread can hold a lock on the
channels container while trying to lock a bridge. At the same
time another thread can hold the lock for said bridge while
attempting to retrieve a channel. This causes a deadlock.
This change fixes this scenario by retrieving a channel snapshot
instead of a channel, as information present in the snapshot
is all that is needed.
ASTERISK-28616
Change-Id: I68ceb1d62c7378addcd286e21be08a660a7cecf2
Found during some testing, there is a race condition between selecting an
appropriate bridge type for a call versus the applying of media on the callee's
session. In some instances a native bridge type would have been chosen, but
due to the callee's media not yet being established at bridge compatibility
check time the simple bridge type is picked instead.
When using chan_pjsip this initiates a topology change event. The topologies
are then compared for the two sessions. However, when the topology was created
for the caller its streams are initialized to "inactive". This topology is then
used as a base when creating the callee's topology, and streams. Soon after
the caller's topology's stream(s) get updated based on the sdp (get set to
sendrecv in the failing scenario).
Now when the topology change event is raised, and the two topologies are
compared, the comparison fails due to a stream state mismatch (sendrecv vs
inactive). And since they differ a reinvite is sent out (to the caller in
this case).
This patch makes it such that when the caller's topology is initially created
it gets created based on its configured endpoint's media topology. When the
endpoint's topology is created its stream's state(s) are initialized to
sendrecv instead of inactive. Subsequently, now when the callee's topology is
created its topology streams are now initialized to sendrecv. Thus when the
topology change event occurs due to the mentioned scenario the stream states
match for the given sessions, and the reinvite is not sent unless due to some
other valid mismatch.
Note, this patch only changes one pending media state's creation point. It's
possible other places *could* be changed, however for now it was deemed best
to only alter what's here.
Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7
The following modules needed tweaks for API changes.
addons/cdr_mysql.c
addons/chan_ooh323.c
apps/app_meetme.c
ASTERISK-28604
Change-Id: Ib40e513ae55b5114be035cdc929abb6a8ce2d06d
If the "max_retries" option is set to 0 then upon failure no
further attemps are made, so explicitly document the behavior.
ASTERISK-28602
Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf
Calling ne_uri_parse allocates memory that needs to be freed with a
corresponding call to ne_uri_free.
ASTERISK-28572 #close
Change-Id: I8a6834da27000a6807d89cb7a157b2a88fcb5e61
This change ensures that the module isn't unloaded when a
WebSocket is open. Previously it was possible to unload the
module manually or during shutdown which could cause a crash
when any active WebSockets were terminated.
ASTERISK-28585
Change-Id: I85c71ab112f99875b586419a34c08c8b34c14c5c
When we created the External Media addition to ARI we created an
ExternalMedia object to be returned from the channels/externalMedia
REST endpoint. This object contained the channel object that was
created plus local_address and local_port attributes (which are
also in the Channel variables). At the time, we thought that
creating an ExternalMedia object would give us more flexibility
in the future but as we created the sample speech to text
application, we discovered that it doesn't work so well with ARI
client libraries that a) don't have the ExternalMedia object
defined and/or b) can't promote the embedded channel structure
to a first-class Channel object.
This change causes the channels/externalMedia REST endpoint to
return a Channel object (like channels/create and channels/originate)
instead of the ExternalMedia object.
Change-Id: If280094debd35102cf21e0a31a5e0846fec14af9
During execution "pjsip show channelstats" cli command by an
external module asterisk crashed. It seems this is a separate
thread running to fetch and print rtp stats. The crash happened on
the ao2_lock method, just before it going to read the rtp stats on
a rtp instance. According to gdb backtrace log, it seems the
session media was already cleaned up at that moment.
ASTERISK-28578
Change-Id: I3e05980dd4694577be6d39be2c21a5736bae3c6f
If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav"
and are using realtime ODBC backend, only the first format gets stored
in the database. So when you forward a message later on, there is a bug
generating the email, related to the stored format (GSM) being different
than the desired email format (WAV) specified for the user. Sox can
handle this, but Asterisk needs to tell sox exactly what to do.
ASTERISK-22192
Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd
Salah Ahmed
Benjamin Keith Ford
George Joseph
Friendly Automation
Kevin Harwell
Sean Bright
Michael Cargile
Frederic LE FOLL
lvl
Alexei Gradinari
Joshua Colp
Martin Tomec
Salah Ahmed
cmaj