George Yang authored 1 year ago and Erik Karlsson committed 1 month ago

(cherry picked from commit f57639f7)

Suru Dissanaike authored 1 year ago and Erik Karlsson committed 1 month ago

(cherry picked from commit 45d9fc14)

UCI asterisk.sip_options.bindport was not applied, in case asterisk.sip_options.bindintf
also configured.

asterisk-chan-voicemngr_2.1.14: Fix a bug that call clearing tones are not started when two other call parties hangup

- Fixed a bug in CDR file rename
- Increase debug logging for RTP telephone event

asterisk-chan-voicemngr-2.1.12: Show more information of channels and sub-channels in CLI "telephony show status"

Set connection as -1 in chan_voicemngr_close_connection() if there is
other call on the line.

This config option has become obsolete afterv the below commit in asterisk.
voice/asterisk@bbc11efe

Queue DTMF frames to the channel if the call is terminated by Asterisk core or application

SIP provider like sip0, sip1 other than user (phone number) shall be used in extensions.conf
for incoming calls.

Fix a crash when an internal call between two DECT handsets is terminated

Forward inbound RTP telephone event packets to voicemngr transparently.

- Remove uci option asterisk.sipX.dtmf_mode and use asterisk.sip_options.dtmf_mode for all SIP
  clients
- Remove dtmfrelay from chan_telephony.conf since it is not used any longer. Now this config is
  got from the peer SIP channel

Reduce source jitters by chaning the scheduling policy and priority
of the RTP handling threads.

Fix a deadlock issue triggered by R1

Support dynamic DTMF payload type

No logic changes

Fix a regression that is introduced by RTP events generated by DSP

If multiple arguments are allowed to be passed to echo, it is possible
to inject newlines using something like '-e aaa\nbbb\nccc" in UCI.

Also fix missing quotation and missing escape_sed

And also with few improvements.

No logic changes

 - asterisk-chan-voicemngr 2.1.3: Send DTMF rtp-event packets generated by DSP to asterisk core for 2-way call  
 - asterisk: Pass DTMF rpt-event packets directly for 2-way calls

No logic changes

No logic changes

No logic changes

No logic changes