Skip to content
GitLab
Explore
Sign in
Register
Voice
asterisk
Merge requests
Open
1
Merged
142
Closed
21
All
164
Actions
Subscribe to RSS feed
Recent searches
{{formattedKey}}
{{ title }}
{{ help }}
{{name}}
@{{username}}
None
Any
{{name}}
@{{username}}
None
Any
{{name}}
@{{username}}
None
Any
{{name}}
@{{username}}
None
Any
{{name}}
@{{username}}
None
Any
Upcoming
Started
{{title}}
None
Any
{{title}}
None
Any
{{title}}
None
Any
{{name}}
Yes
No
Yes
No
{{title}}
{{title}}
{{title}}
Created date
Cleanup ast_channel references for both hold/unhold SIP channels
!144
· created
Jan 23, 2024
by
George Yang
Merged
0
updated
Jan 23, 2024
DNS SRV resolver: stick to the latest server address
!143
· created
Jan 18, 2024
by
George Yang
Closed
3
updated
Jan 23, 2024
DNS SRV resolver: stick to the latest server address
!142
· created
Jan 18, 2024
by
George Yang
Closed
0
updated
Jan 18, 2024
DNS SRV resolver and cleanup ast_channel references
!141
· created
Jan 17, 2024
by
George Yang
Closed
5
updated
Jan 23, 2024
Use the real timestamp value for the audio frame->ts
!140
· created
Jan 04, 2024
by
Grzegorz Sluja
Merged
Approved
2
updated
Jan 10, 2024
Relace the calls to ubus_free() with ubus_free_context()
!139
· created
Dec 07, 2023
by
Yalu Zhang
Merged
0
updated
Dec 07, 2023
Merge branch asterisk-20.3.0 into devel properly
!138
· created
Nov 09, 2023
by
Andreas Gnau
Merged
1
updated
Nov 10, 2023
Fix the issue that asterisk crash during some auto-test
!137
· created
Nov 08, 2023
by
Wenpeng Song
Merged
5
updated
Nov 10, 2023
Fix sequence number used by asterisk for outgoing RTP packets
!136
· created
Nov 02, 2023
by
Grzegorz Sluja
release-7.2
Merged
0
updated
Nov 02, 2023
Fix sequence number used by asterisk for outgoing RTP packets
!135
· created
Nov 02, 2023
by
Grzegorz Sluja
Merged
0
updated
Nov 02, 2023
Correction for some error message during asterisk restart
!134
· created
Oct 31, 2023
by
Wenpeng Song
Merged
3
updated
Nov 01, 2023
Fix sequence number used by asterisk for outgoing RTP packets
!133
· created
Oct 31, 2023
by
Grzegorz Sluja
release-6.5
Merged
0
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
Approved
4
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
Prev
1
2
3
4
5
6
…
9
Next