|Summary:||ASTERISK-17254: Dial MulticastRTP channel with A option can't play the file|
|Date Opened:||2011-01-17 01:02:02.000-0600||Date Closed:||2012-09-25 07:00:27|
|Environment:||Attachments:||( 0) trace.cap|
I am trying to play a message to a multicast channel:
I can talk to the channel but the message is never played:
Using SIP RTP CoS mark 5
-- Executing [1001@default:1] Dial("SIP/10.147.248.29-00000003", "MulticastRTP/basic/126.96.36.199:5004,,A(demo-moreinfo)") in new stack
-- Called basic/188.8.131.52:5004
-- MulticastRTP/0x2306b728 answered SIP/10.147.248.29-00000003
[Jan 17 07:18:30] WARNING: file.c:751 ast_readaudio_callback: Failed to write frame
-- <MulticastRTP/0x2306b728> Playing 'demo-moreinfo.ulaw' (language 'en')
[Jan 17 07:18:30] ERROR: app_dial.c:2324 dial_exec_full: error streaming file 'demo-moreinfo' to callee
|Comments:||By: Leif Madsen (lmadsen) 2011-01-17 09:04:20.000-0600|
Please capture and upload the multicast traffic here.
By: Yohann (wybecom) 2011-01-18 00:43:54.000-0600
10.147.248.29 is the Asterisk.
10.151.248.21 is the phone calling.
By: Leif Madsen (lmadsen) 2011-01-19 14:29:33.000-0600
Well I was talking to Russell, and he basically said if there is traffic in the capture (which I see), then the problem is a configuration issue on the phone listening to the multicast traffic.
By: Casey Morford (cmorford) 2011-01-19 15:15:29.000-0600
I have this same issue on 184.108.40.206
Talking seems to transmit just fine over multicast without the A() option, so if the issue is on the listening phone side, then I would expect talking would fail as well.
By: Yohann (wybecom) 2011-01-20 00:21:42.000-0600
Yes there is traffic in the capture but it's not the traffic caused by the A option. It's the traffic generated by the caller.
So, when dial a multicast destination with A option or not, the caller is always heard from the listening phone. It's not a configuration issue on the listening side. The fact is that you can't stream a file to the multicast channel.