- Feb 05, 2018
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Richard Mudgett authored
ASTERISK-27651 Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34
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Jenkins2 authored
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Jenkins2 authored
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- Feb 03, 2018
- Feb 02, 2018
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Richard Mudgett authored
Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068
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Richard Mudgett authored
Change-Id: I3c7106ff77009754725cee790eadf5da44154ab6
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Richard Mudgett authored
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Joshua Colp authored
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- Feb 01, 2018
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Sungtae Kim authored
* Changed to create ami_event string only when the given blob is not json_null(). * Fixed bad expression. ASTERISK-27621 Change-Id: Ice58c16361f9d9e8648261c9ed5d6c8245fb0d8f
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Joshua Elson authored
ASTERISK-27652 #close Change-Id: I78a0d38bfd8d0d82830f3d53da04872d6b67284d
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Sean Bright authored
Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8
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Richard Mudgett authored
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
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Richard Mudgett authored
* Made not allocate memory if the channel snapshot is an internal channel. * Free memory earlier when no longer needed. Change-Id: Ia06e0c065f1bd095781aa3f4a626d58fa4d28b38
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Jenkins2 authored
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George Joseph authored
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Jenkins2 authored
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Jenkins2 authored
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- Jan 31, 2018
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Jenkins2 authored
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Jenkins2 authored
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Jenkins2 authored
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Corey Farrell authored
Verified nothing in the testsuite lists this module as a dependency. Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813
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Richard Mudgett authored
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
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Richard Mudgett authored
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
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Richard Mudgett authored
Need to include signal.h to define pthread_kill() and SIGURG. Change-Id: I10ae3aa4bf8e7386ac29ade78c0f2caed8e674fa
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Jenkins2 authored
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Jenkins2 authored
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Jenkins2 authored
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Jenkins2 authored
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Corey Farrell authored
pjproject does not have a function to reverse pjsip_inv_usage_init. This means we need to ignore any calls to the functions once shutdown is final. ASTERISK-27571 #close Change-Id: Ia550fcba563e2328f03162d79fb185f16b7c9b9d
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- Jan 30, 2018
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Corey Farrell authored
Create ast_atomic macro's to provide a consistent interface to the common functionality of __atomic and __sync built-in functions. ASTERISK-27619 Change-Id: Ieba3f81832a0e25c5725ea067e5d6f742d33eb5b
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George Joseph authored
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
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Jenkins2 authored
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Jenkins2 authored
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Jenkins2 authored
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- Jan 29, 2018
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George Joseph authored
res_pjsip_registrar_expire remains as an empty module for now. Change-Id: Ib93698938bae548d2199cb542f3692d1a171239f
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Corey Farrell authored
The sample modules.conf explicitly loaded res_musiconhold.so. This is redundent as autoload=yes is already set. It causes warnings if res_musiconhold.so was not installed and results in an unexpected load if the admin disables autoload without remembering to remove the res_musiconhold load statement. Also remove reference to unknown module pbx_gtkconsole. Change-Id: Ib01888994d9f1364b14d3c9fb6ff96774a6e580a
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Jenkins2 authored
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