Asterisk   Asterisk Support


Topic Replies Activity
PJSIP multiple AOR: avoid calling other contacts when busy 3 June 28, 2019
Using local channels to set up the CDR collection 3 June 28, 2019
Joinempty = inuse not working 3 June 28, 2019
Configure OPUS codec 3 June 28, 2019
Can Asterisk Record voice with the functionality of Jitter Buffer 5 June 28, 2019
Music-On-Hold Tracking 2 June 27, 2019
Missing "SIP.conf" file 3 June 27, 2019
Honor the codec preference 3 June 27, 2019
Unable to see the number people are calling me 3 June 27, 2019
G729 Licence Registration 3 June 26, 2019
Wider columns in PJSIP CLI output? 2 June 26, 2019
Playback & Background billing 12 June 26, 2019
Callers wait a lot before be dispatched to queue member 6 June 26, 2019
No cdr when dialstatus is CANCEL 4 June 26, 2019
Need Support and possible upgrade from old ver 3 June 25, 2019
MixMonitor - Understanding about operation 2 June 25, 2019
Queue Retry Option 2 June 25, 2019
Audio format and codecs 2 June 25, 2019
Nature of Address header on SIP channel 5 June 25, 2019
Play specific file with StartMusicOnHold instead of files on folder 2 June 25, 2019
Sounds in queue at different times 4 June 25, 2019
Get QueueMember TalkTime from `QueueStatus` API 4 June 24, 2019
Channel.c: Exceptionally long voice queue length queuing to Local 8 June 24, 2019
Get trunk name on Monitor Trunk Failure callback 4 June 24, 2019
Get more information about SIP_CAUSE 2 June 21, 2019
Mute participant on entry into confbridge 4 June 21, 2019
Opus: decoding: buffer too small 6 June 21, 2019
Playback files from http server in an streaming way possible? 5 June 20, 2019
Meetme Users Limit 4 June 20, 2019
Cracked audio in bridged calls 2 June 19, 2019