Merge requests · Voice / asterisk · GitLab

Fix sequence number used by asterisk for outgoing RTP packets

!133 · created Oct 31, 2023 by Grzegorz Sluja release-6.5
updated Nov 03, 2023
Draft: limit memory consumption after each call made

!132 · created Oct 31, 2023 by Lukasz Kotasa release-7.2
- Closed
- 1
updated Jan 18, 2024
Fixes for interarrivalJitter and lossRate rtp stats

!131 · created Oct 30, 2023 by Grzegorz Sluja release-6.5
- Merged
- 1
updated Oct 30, 2023
Fixup, SIPIPAddress, correct function type

!130 · created Oct 25, 2023 by Wenpeng Song
- Merged
- 2
updated Oct 25, 2023
fixup! Use the same header for RTP/RTCP packets in DSP and Asterisk

!129 · created Oct 23, 2023 by Grzegorz Sluja release-6.5
- Merged
- 3
updated Oct 23, 2023
Update SIPIPAddress for outgoing calls

!128 · created Oct 18, 2023 by Wenpeng Song
- Merged
- 4
- Approved
updated Oct 21, 2023
Use the same header for RTP packets in DSP and Asterisk

!127 · created Oct 12, 2023 by Grzegorz Sluja release-6.5
- Merged
- 1
updated Oct 12, 2023
SIPIPAddress correction

!126 · created Oct 10, 2023 by Wenpeng Song
- Merged
- 2
updated Oct 11, 2023
res_pjsip_session: Fix session reference leak.

!125 · created Oct 09, 2023 by Lukasz Kotasa
- Merged
- 3
updated Oct 12, 2023
Update the SIPIPAddress for CallLog on outgoing INVITE

!124 · created Oct 04, 2023 by Wenpeng Song
updated Oct 04, 2023
Merge asterisk '20.3.0' into devel

!123 · created Oct 03, 2023 by Grzegorz Sluja
updated Oct 03, 2023
Fix deadlock on 3-way closing and optimize a bit with codec sync

!122 · created Sep 28, 2023 by Wenpeng Song release-6.5
updated Sep 28, 2023
schedule a congestion tone playing instead of playing directly during outgoing INVITE to avoid temp status{401,407} drops the call from DECT

!121 · created Sep 25, 2023 by Wenpeng Song release-6.5
- Merged
- 11
- Approved
updated Sep 27, 2023
Draft: Hangup active channels if SIP registration is lost

!120 · created Sep 25, 2023 by Grzegorz Sluja release-6.5
- Closed
- 2
updated Sep 25, 2023
Draft: correction of ast_control_pvt_cause indication

!119 · created Sep 22, 2023 by Wenpeng Song release-6.5
- Closed
updated Nov 01, 2023
Draft: manager: AOC-S support for AOCMessage

!118 · created Sep 12, 2023 by Grzegorz Sluja
- Closed
updated Nov 03, 2023
Skip trans-coding for outgoing negotiation and ignore trans-path failure

!117 · created Sep 07, 2023 by Wenpeng Song
updated Sep 07, 2023
early media fix complement

!116 · created Sep 06, 2023 by Wenpeng Song release-6.5
updated Sep 07, 2023
Implementation for some rtp stats needed for tr104 objects

!115 · created Sep 06, 2023 by Grzegorz Sluja
- Merged
- 1
updated Sep 06, 2023
Clear sub_peer call state on hangup to avoid "486 BUSY" for subsequent calls

!114 · created Sep 06, 2023 by Lukasz Kotasa release-6.5
- Merged
updated Sep 06, 2023

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