Use the same header for RTP/RTCP packets in DSP and Asterisk

Fix a bug when flash hook is pressed on the second outgoing call during ring-back state

asterisk-chan-voicemngr_2.0.22: Display caller ID for the second incoming call during call waiting if CLIP is enabled

Fix for busy tone playing

Wenpeng Song authored 1 year ago and Yalu Zhang committed 1 year ago

* Play a beep before hangup when the remote hangs up.
* Stop playing the congestion tone from the extension, just wait for 5s that the
  voicemngr plays the tone.
* Re-arrange a bit with the cbbs sequence, so we have 5s busy tone played from
  voicemngr when cbbs is disabled.

Fix asterisk hang during call conference start

Fix a race condition caused by temporary 401/407 response

(cherry picked from commit bfa9f3e4)

... by calling pthread_cancel instead of pthread_kill

Fix a bug that RTP/RTCP ports are not closed for non-answered incoming calls

Fix toggling between calls for 'uk' flash_spec.

Add pointer validations to prevent some null pointer access

Grzegorz Sluja authored 1 year ago and Yalu Zhang committed 1 year ago

An example below.
REGISTER sip:192.168.8.144:5060 SIP/2.0
...
Contact: <sip:100@192.168.8.176:5060>;q=1

Implementation for some RTP stats needed for TR-104 CallLog objects

e83d75b Add support for getting jitter buffer overruns and underruns from the UBUS call
11ddfa3 Fix Source Identifier in RTCP Sender Report

asterisk-chan-voicemngr: Clear sub_peer call state on hangup to avoid "486 BUSY" for subsequent calls