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asterisk-chan-voicemngr
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Created date
send dtmf rtp_event packet from DSP for 2-way call
!70
· created
Apr 24, 2024
by
Wenpeng Song
Merged
Approved
5
updated
Apr 25, 2024
fix a regression from previous commit
!71
· created
Apr 29, 2024
by
Wenpeng Song
Merged
0
updated
Apr 29, 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
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
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
Implement ubus method to retrieve sip registration status
!75
· created
May 15, 2024
by
Iryna Antsyferova
asterisk_chan_voicemngr_rdkb
Merged
0
updated
May 16, 2024
Support dynamic DTMF payload type
!76
· created
May 15, 2024
by
Yalu Zhang
Merged
0
updated
May 15, 2024
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