Well we made the move from 1.8 to a new server running 11.21 asterisk on Debian Jesse. ConfBridge seems pretty straight forward – I left the defaukts for now and I have this in the extension.conf
exten => 2828,1,ConfBridge(2828)
Dial it and it connects but there is no audio–
Thanks in advance for any ideas. Other than that it works well.
Thanks,
Tom
What is the console output, do you have all modules loaded? Is timing working on the system? (timing test from CLI will test it)
Timer info:
Attempting to test a timer with 50 ticks per second.
Using the ‘timerfd’ timing module for this test.
It has been 1000 milliseconds, and we got 50 timer ticks
CLI output of a client joining the conference:
– Executing [2828@Mississippi:1] ConfBridge(“SIP/28026-00000013”, “2828”) in new stack
– <SIP/28026-00000013> Playing ‘confbridge-join.gsm’ (language ‘en’)
> 0xb6d3cc40 – Probation passed - setting RTP source address to 107.xxx.xxx.xxx:10026
[2016-08-24 09:07:27] NOTICE[30909][C-00000013]: res_rtp_asterisk.c:4444 ast_rtp_read: Unknown RTP codec 126 received from ‘107.xxx.xxx.xxx:10026’
> 0xb6d3cc40 – Probation passed - setting RTP source address to 107.xxx.xxx.xxx:10026
– <Bridge/0xa816aa4-input> Playing ‘confbridge-join.gsm’ (language ‘en’)
And what does “rtp set debug on” show? Is media flowing out to the device, and do you hear the join sound?
Looks pretty good:
Got RTP packet from 96.19.104.60:33082 (type 18, seq 005794, ts 161434408, len 000020)
Sent RTP packet to 107.131.65.77:10200 (type 18, seq 024909, ts 9604304, len 000020)
Got RTP packet from 107.131.65.77:10200 (type 18, seq 044999, ts 3165775224, len 000020)
Got RTP packet from 107.131.65.77:10200 (type 18, seq 045000, ts 3165775384, len 000020)
Got RTP packet from 107.131.65.77:10200 (type 18, seq 045001, ts 3165775544, len 000020)
Got RTP packet from 104.232.168.90:48398 (type 18, seq 011140, ts 2402728736, len 000020)
Sent RTP packet to 67.61.220.85:10300 (type 18, seq 038366, ts 9209440, len 000020)
Got RTP packet from 104.232.170.26:28438 (type 18, seq 063773, ts 2255972534, len 000020)
Sent RTP packet to 104.232.170.26:28438 (type 18, seq 045466, ts 10286864, len 000020)
Sent RTP packet to 198.0.28.161:60742 (type 18, seq 012660, ts 9901504, len 000020)
Got RTP packet from 198.0.28.161:60742 (type 18, seq 041859, ts 149496354, len 000020)
Sent RTP packet to 96.19.104.60:33082 (type 18, seq 034760, ts 9450048, len 000020)
Sent RTP packet to 66.175.147.162:13096 (type 18, seq 062350, ts 9518816, len 000020)
Sent RTP packet to 104.232.168.90:48398 (type 18, seq 006909, ts 9473240, len 000020)
Got RTP packet from 208.60.31.66:10030 (type 18, seq 015626, ts 2436391834, len 000020)
Sent RTP packet to 208.60.31.66:10030 (type 18, seq 064704, ts 9459576, len 000020)
I do hear the join beep as well
Nothing abnormal there. Media certainly looks as though it is flowing. Have you confirmed bidirectional flow in Asterisk using the Echo() application?
Echo test works perfectly and I can make point to point calls without issue
Very interesting. Nothing else springs to mind and I don’t think I’ve seen any issues reporting a problem like that…
Is it possible to try to narrow things down further by using a codec like ulaw or g722? That eliminates G729 from the equation (just in case).
My similarly light weight (and working) confbridge includes references to Answer and Hangup. Not sure if Hangup is really needed, but I was under the impression that Answer was.
exten => 8200,1,Answer(250)
same => n,ConfBridge(8200,,,sample_user_menu)
same => n,Hangup()
This works for me in Asterisk 11.19 running in an Amazon EC2 instance booting Amazon’s Linux.