astobj2.c declares DEBUG_THREADS_LOOSE_ABI to avoid overhead of debug
threads tracking information in the internal structures of astobj2.
Unfortunately this means that ao2_global_obj contains the statically
allocated debug threads tracking fields which are used by initialization
and cleanup but main/astobj2.c believed those fields and associated
space did not exist.
Change-Id: Icef41ad97d88a8c1d1515e034ec8133cab3b1527
When a stale item was being updated the object was being retrieved, but its
reference was not being decremented after the update. This patch makes it so
the object is now appropriately de-referenced.
ASTERISK-28523
Change-Id: I9d8173d3a0416a242f4eba92fa0853279c500ec7
You can currently capture backtraces of memory allocations but they
only get displayed when you stop asterisk and the atexit hooks
are enabled. Now, if memory backtrace is on and you issue a
"memory show allocations" CLI command for a specific file, then
a backtrace will show for each allocation that occurred after
you turned "memory backtrace on". The backtrace display is shown
only when a specific file's allocations are displayed to prevent
a massive CLI dump of every file's allocations.
Change-Id: Ic657afc1fc6ec7205e16eb36a97a611d235a2b4f
ast_mwi_topic() returns a borrowed reference which should not be
unreferenced, doing so leads to a FRACK. This was hidden by the fact
that stasis_cache.c leaked the result of cache_remove in
caching_topic_exec.
Change-Id: I51101bf7d07b8dc8ce8fc46b6cb31fbbd213fbc7
When fax detection occurs on an outbound PJSIP channel the
redirect operation will result in a masquerade occurring and
the underlying channel on the session changing. The code
incorrectly relocked the new channel instead of the old
channel when returning. This resulted in the new channel
being locked indefinitely. The code now always acts on the
expected channel.
ASTERISK-28538
Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3
On FreeBSD using the clang/llvm compiler build fails to build due
to the switch statement argument being a non integer type expression.
Switch to an if/else if/else construct to sidestep the issue.
ASTERISK-28536 #close
Change-Id: Idf4a82cc1e94580a2d017fe9e351c226f23e20c8
When modifying an already defined variable in some channel drivers they
add a new variable with the same name to the list, but that value is
never used, only the first one found.
Introduce ast_variable_list_replace() and use it where appropriate.
ASTERISK-23756 #close
Patches:
setvar-multiplie.patch submitted by Michael Goryainov
Change-Id: Ie1897a96c82b8945e752733612ee963686f32839
This fix allows a realtime moh class to be unregistered from the command
line. This is useful when the contents of a directory referenced by a
realtime moh class have changed.
The realtime moh class is then reloaded on the next request and uses the
new directory contents.
ASTERISK-17808
Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce
ChanIsAvail() creates a temporary channel with ast_request() to test
resource availability. It should not generate a CDR when it hangs up
this temporary channel.
This patch disables CDR generation for the temporary channel with
ast_cdr_set_property().
ASTERISK-28527
Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1
When the remote ISDN party ends an ISDN call on a PRI link
(DISCONNECT), CHANNEL(hangupsource) information is not available.
chan_dahdi already contains an ast_set_hangupsource() in
__dahdi_exception() function but it seems that ISDN message processing
does not use this part of code.
Two other channel modules associate ast_queue_hangup() and
ast_set_hangupsource() functions calls:
- chan_pjsip in chan_pjsip_session_end() function,
- chan_sip in sip_queue_hangup_cause() function.
chan_iax2 separates them, in iax2_queue_hangup()/iax2_destroy() and
set_hangup_source_and_cause().
Thus, I propose to add ast_set_hangupsource() beside
ast_queue_hangup() in sig_pri_queue_hangup(), like chan_pjsip and
chan_sip already do.
ASTERISK-28525
Change-Id: I0f588a4bcf15ccd0648fd69830d1b801c3f21b7c
This change removes the assumption that a frame will always have
a src set on it. This assumption is incorrect.
Given a scenario where an RTP packet is received with no payload
the resulting audio frame will have no samples. If this frame goes
through a signed linear translation path an interpolated frame can
be created (if generic packet loss concealment is enabled) that has
minimal data on it, including no src. If this frame is given to a
translation path a crash will occur due to the lack of src.
ASTERISK-28499
Change-Id: I024d10dd98207eb8a6b35b59880bcdf1090538f8
On reading information about initial client packet unistim use dirty
implementation of destination ip address retrieval. This fix uses
CMSG_*(..) to get ip address and make clang compile without warning.
ASTERISK-25592 #close
Reported-by: Alexander Traud
Change-Id: Ic1fd34c2c2bcc951da65bf62e3f7a8adff8351b1
res_pjsip_mwi allows both solicited and unsolicited MWI subscription types.
While both can be set in the configuration for a given endpoint/aor, only
one is allowed. Precedence is given to unsolicited. Meaning if an endpoint/aor
is configured to allow both types then the solicited subscription is rejected
when it comes in. However, there is a configuration option to override that
behavior:
mwi_subscribe_replaces_unsolicited
When set to "yes" then when a solicited subscription comes in instead of
rejecting it Asterisk is suppose to replace the unsolicited one if it exists.
Prior to this patch there was a bug in Asterisk that allowed the solicted one
to be added, but did not remove the unsolicited. As a matter of fact a new
unsolicited subscription got added everytime a SIP register was received.
Over time this eventually could "flood" a phone with SIP notifies.
This patch fixes that behavior to now make it work as expected. If configured
to do so a solicited subscription now properly replaces the unsolicited one.
As well when an unsubscribe is received the unsolicited subscription is
restored. Logic was also put in to handle reloads, and any configuration changes
that might result from that. For instance, if a solicited subscription had
previously replaced an unsolicited one, but after reload it was configured to
not allow that then the solicited one needs to be shutdown, and the unsolicited
one added.
ASTERISK-28488
Change-Id: Iec2ec12d9431097e97ed5f37119963aee41af7b1
Current implementation of ast_channel_tech send_digit_begin hook uses
same function for tone playback as key press handler. This cause every
incoming dtmf send back to asterisk. In case of two unistim phones
connected to each other, it'll cause indefinite DTMF loop. Fix add
separate function for dtmf tone phone play.
Change-Id: I5795db468df552f0c89c7576b6b3858b26c4eab4
This patch fixes one-way oudio that users expirienced on
big-endian architechtires. RTP port number bytes was stored
in improper order and phone sent RTP to wrong RTP port.
Reported-by: Andrey Ionov
Change-Id: I9a9ca7f26e31a67bbbceff12923baa10dfb8a3be
NEC SIP Station interface with authenticated registration only supports cnonce
up to 32 characters. In Linux, PJSIP would generate 36 character cnonce
which included hyphens. Teluu developed this patch adding a compile time
setting to default to not include the hyphens. They felt it best to still
generate the UUID and strip the hyphens.
They have indicated it will be part of PJSIP 2.10.
ASTERISK-28509
Reported-by: Dan Cropp
Change-Id: Ibdfcf845d4f8c0a14df09fd983b11f2d72c5f470
In chan_sip, there was variable SIPFROMDOMAIN that allows to set
From header URI domain per channel. This patch introduces res_pjsip
variable SIPFROMDOMAIN for backward compatibility with chan_sip.
ASTERISK-28489
Change-Id: I715133e43172ce2a1e82093538dc39f9e99e5f2e
Given the following request path and 2 handler paths...
Request: /channels/externalMedia
Handler: /channels/{channelId} "wildcard"
Handler: /channels/externalmedia "non-wildcard"
...if /channels/externalMedia was registered as a handler after
/channels/{channelId} as shown above, the request would automatically
match the wildcard handler and attempt to parse "externalMedia" into
the channelId variable which isn't what was intended. It'd work
if the non-wildard entry was defined in rest-api/api-docs/channels.json
before the wildcard entry but that makes the json files
order-dependent which isn't a good thing.
To combat this issue, the search loop saves any wildcard match but
continues looking for exact matches at the same level. If it finds
one, it's used. If it hasn't found an exact match at the end of
the current level, the wildcard is used. Regardless, after
searching the current level, the wildcard is cleared so it won't
accidentally match for a different object or a higher level.
BTW, it's currently not possible for more than 1 wildcard entry
to be defined for a level. For instance, there couldn't be:
Handler: /channels/{channelId}
Handler: /channels/{channelName}
We wouldn't know which one to match.
Change-Id: I574aa3cbe4249c92c30f74b9b40e750e9002f925
There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:
1. There is no rx and no tx audio, so return nothing.
2. There is rx but no tx audio, so return rx.
3. There is tx but no rx audio, so return tx.
4. There is rx and tx audio, so mix them and return.
The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.
If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.
This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.
Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e
The function leave_voicemail checks if expungeonhangup is set,
but does not check if IMAP stream is closed,
so it could call imap function with NULL stream.
This leads to segfault.
ASTERISK-28505 #close
Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c
Because keypad() is exported by libtinfo, it needs to be explicitly
added to the linker options.
ASTERISK-28487 #close
Change-Id: I6c2ad5b95f422c263d078b5c0e84c111807dffc6
Somehow it's possible for the srtp session object to be NULL even though the
Asterisk srtp object itself is valid. When this happened it would cause a
crash down in the srtp code when attempting to protect or unprotect data.
After looking at the code there is at least one spot that makes this situation
possible. If Asterisk fails to unprotect the data, and after several retries
it still can't then the srtp->session gets freed, and set to NULL while still
leaving the Asterisk srtp object around. However, according to the original
issue reporter this does not appear to be their situation since they found
no errors logged stating the above happened (which Asterisk does for that
situation).
An issue was found however, where a possible race condition could occur between
the pjsip incoming negotiation, and the receiving of RTP packets. Both places
could attempt to create/setup srtp for the same rtp instance at the same time.
This potentially could be the cause of the problem as well.
Given the above this patch adds locking around srtp setup for a given rtp, or
rtcp instance. NULL checks for the session have also been added within the
protect and unprotect functions as a precaution. These checks should at least
stop Asterisk from crashing if it gets in this situation again.
This patch also fixes one other issue noticed during investigation. When doing
a replace the old object was freed before creating the replacement. If the new
replacement object failed to create then the rtp/rtcp instance would now point
to freed srtp data which could potentially cause a crash as well when the next
attempt to reference it was made. This is now fixed so the old srtp object is
kept upon replacement failure.
Lastly, more logging has been added to help diagnose future issues.
ASTERISK-28472
Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc
To make throttling by label fully active, the "throttle" option
has to be specified with a specific label.
You can now specify "skip_gate" in the Gerrit comments when you
do a +2 code review to tell Jenkins not to actually run the
gate. You'd do this if you plan to manually merge the change.
Also updated the "printenv" debug output to better sort multi-line
comments.
Change-Id: I4c0b1085acec4805f2ca207eebac50aad81f27e2
