Check if initreq data string exists before using it when processing a
CANCEL request.
ASTERISK-27666
Change-Id: Id1d0f0fa4ec94e81b332b2973d93e5a14bb4cc97
Jansson is thread safe for all read-only functions and reference
counting starting v2.11. This allows simplification of our code and
removal of locking around reference counting and dumping.
Change-Id: Id985cb3ffa6681f9ac765642e20fcd187bd4aeee
When <http://github.com/BelledonneCommunications/bcg729> is installed, PJProject
tries to link that. Support for this bcg729 was added with PJProject 2.7. The
issue happens, because Teluu enabled that new feature on default.
ASTERISK-27584
Reported by: Stuart Henderson
Change-Id: I88b6b18ad777bcfe2d8201187b4b90eec0a172a6
When the latest version of that library was installed, Asterisk did not build.
ASTERISK-27669
Reported by: Николай Михо
Change-Id: I27e09bb875fdd56423bd9fae1be85fddb428eb96
Need to remove all CDR's listed by a CDR object from the active_cdrs_all
container including the root/master record.
ASTERISK-27656
Change-Id: I48b4970663fea98baa262593d2204ef304aaf80e
It seems that the ALSA backend of PortAudio doesn't know how to both
read and write at the same time by adding a per-device mutex.
FIXME: currently only a draft version. Need to either auto-detect
we work with the ALSA backend or add an extra configuration option
to use this mutex.
ASTERISK-27426 #close
Change-Id: I635eacee45f5413faa18f5a3b606af03b926dacb
* Made the AMI ConfbridgeList action's ConfbridgeList events output all
the standard channel snapshot headers instead of a few hand-coded channel
snapshot headers. The benefit is that the CallerIDName gets disruptive
characters like CR, LF, Tab, and a few others escaped. However, an empty
CallerIDName is now output as "<unknown>" instead of "<no name>".
ASTERISK-27651
Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977
res_pjsip_endpoint_identifier_user.c:
* Fix copy/paste error in find_endpoint(). We were using a constant
"anonymous" string instead of the passed in endpoint_name when checking
the transport domain for an endpoint match.
* Eliminate RAII_VAR in find_endpoint().
* Remove always true check in find_transport_state_in_use().
* Remove useless CMD_STOP in find_transport_state_in_use().
res_pjsip_endpoint_identifier_anonymous.c:
* Eliminate RAII_VAR in anonymous_identify().
* Remove always true check in find_transport_state_in_use().
* Remove useless CMD_STOP in find_transport_state_in_use().
Change-Id: I86924c31db5bd225ca0c1219c761b668c6f91189
* Changed to create ami_event string only when the given blob is not
json_null().
* Fixed bad expression.
ASTERISK-27621
Change-Id: Ice58c16361f9d9e8648261c9ed5d6c8245fb0d8f
ast_str_append_event_header() could potentially leak and corrupt memory if
the ast_str needed to expand to add the AMI event header.
* Fixed to return error if the ast_str_append() failed.
Change-Id: I92f36b855540743b208d76e274152ee2d758176d
* Made not allocate memory if the channel snapshot is an internal channel.
* Free memory earlier when no longer needed.
Change-Id: Ia06e0c065f1bd095781aa3f4a626d58fa4d28b38
In an earlier release, inbound registrations on a reliable transport
were pruned on Asterisk restart since the TCP connection would have
been torn down and become unusable when Asterisk stopped. This same
process is now also applied to inbound subscriptions.
Also fixed issues in res_pjsip_registrar where it wasn't handling the
monitoring correctly when multiple registrations came in over the same
transport.
To accomplish this, the pjsip_transport_event feature needed to
be refactored to allow multiple monitors (multiple subcriptions or
registrations from the same endpoint) to exist on the same transport.
Since this changed the API, any external modules that may have used the
transport monitor feature (highly unlikey) will need to be changed.
ASTERISK-27612
Reported by: Ross Beer
Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36
The functionality of this module was already moved to
res_pjsip_registrar, mark it deprecated and add message to CHANGES.
Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813
Currently in app_confbridge if someone mutes a channel while that channel
is talking, the talk detection code is suspended while the channel is
muted. As far an an external observer is concerned, the muted channel's
talk status is still "talking" even though the channel is not contributing
audio to the conference bridge. When the channel is later unmuted, it
takes the usual 'dsp_silence_threshold' option time to clear the talking
status even though the channel may have stopped talking while the channel
was muted.
* In bridge_softmix.c, clear the talking status and report talking stopped
if the channel was talking when the channel is muted. When the channel is
unmuted and the channel is still talking then report the channel as
talking since it is contributing audio to the bridge again.
ASTERISK-27647
Change-Id: Ie4fdbc05a0bc7343c2972bab012e2567917b3d4e
The dsp_talking_threshold does not represent time in milliseconds. It
represents the average magnitude per sample in the audio packets. This is
what the DSP uses to determine if a packet is silence or talking/noise.
Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443
Jenkins2
Joshua Colp
Corey Farrell
Richard Mudgett
Alexander Traud
Oron Peled
Sungtae Kim
Joshua Elson
Sean Bright
George Joseph