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Updated date
Implementation of speed dialing callList, REF 14345
!90
· created
Aug 29, 2024
by
Grzegorz Sluja
asterisk_chan_voicemngr_rdkb
Merged
35
updated
Sep 03, 2024
Extend 'call_status' ubus method with "call_waiting_status" and "conference_status", REF 14694
!86
· created
Jun 28, 2024
by
Iryna Antsyferova
asterisk_chan_voicemngr_rdkb
Merged
13
updated
Aug 13, 2024
Fix crash at local calls between pbx<->fxs, REF 14916
!89
· created
Aug 01, 2024
by
Iryna Antsyferova
asterisk_chan_voicemngr_rdkb
Merged
2
updated
Aug 06, 2024
Fix issues found in internal call processing
!88
· created
Jul 30, 2024
by
Grzegorz Sluja
asterisk_chan_voicemngr_rdkb
Merged
1
Approved
updated
Jul 30, 2024
Queue DTMF frames to the channel if the call is terminated by Asterisk core or application
!87
· created
Jul 11, 2024
by
Yalu Zhang
Merged
2
updated
Jul 11, 2024
Fix a crash when an internal call between two DECT handsets is terminated
!85
· created
Jun 27, 2024
by
Yalu Zhang
Merged
updated
Jun 27, 2024
Forward RTP telephone event packets to voicemngr
!84
· created
Jun 24, 2024
by
Yalu Zhang
Merged
updated
Jun 25, 2024
Implement ubus method to trigger "pjsip send register"
!83
· created
Jun 19, 2024
by
Iryna Antsyferova
asterisk_chan_voicemngr_rdkb
Merged
updated
Jun 20, 2024
Make max session per extension configurable
!68
· created
Mar 26, 2024
by
Bogdan Bogush
asterisk_chan_voicemngr_rdkb
Merged
updated
Jun 03, 2024
Add line invoke support, REF 14346
!82
· created
May 24, 2024
by
Wenpeng Song
asterisk_chan_voicemngr_rdkb
Merged
3
updated
May 28, 2024
Termination digit logic modify, REF 14468
!81
· created
May 23, 2024
by
Wenpeng Song
asterisk_chan_voicemngr_rdkb
Merged
updated
May 23, 2024
Fix a deadlock issue triggered by R1
!80
· created
May 22, 2024
by
George Yang
Merged
2
updated
May 23, 2024
Fix a deadlock issue triggered by R1
!79
· created
May 22, 2024
by
George Yang
release-7.3
Merged
updated
May 23, 2024
Implement ubus method to retrieve sip registration status
!75
· created
May 15, 2024
by
Iryna Antsyferova
asterisk_chan_voicemngr_rdkb
Merged
updated
May 16, 2024
Support dynamic DTMF payload type
!76
· created
May 15, 2024
by
Yalu Zhang
Merged
updated
May 15, 2024
14115 add config for mid call tone
!74
· created
May 13, 2024
by
Wenpeng Song
asterisk_chan_voicemngr_rdkb
Merged
2
updated
May 14, 2024
Change maximum number of sip clients supported to 10
!73
· created
May 08, 2024
by
Grzegorz Sluja
asterisk_chan_voicemngr_rdkb
Merged
1
updated
May 08, 2024
Generate call waiting tone by Asterisk instead of by DSP
!72
· created
Apr 30, 2024
by
Yalu Zhang
Merged
1
updated
May 02, 2024
fix a regression from previous commit
!71
· created
Apr 29, 2024
by
Wenpeng Song
Merged
updated
Apr 29, 2024
send dtmf rtp_event packet from DSP for 2-way call
!70
· created
Apr 24, 2024
by
Wenpeng Song
Merged
5
Approved
updated
Apr 25, 2024
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