AudioSocket From Trunk RTP Problem

Asterisk Asterisk APIs
1

I wrote the dialplan below;

exten => 102,1,Verbose("Call to AudioSocket via Dialplan Application")
 same => n,Answer()
 same => n,AudioSocket(40325ec2-5efd-4bd3-805f-53576e581d13,192.168.0.170:8181)
 same => n,Hangup()

exten => s,1,Verbose("Call to AudioSocket via Dialplan Application")
 same => n,Answer()
 same => n,AudioSocket(40325ec2-5efd-4bd3-805f-53576e581d13,192.168.0.170:8181)
 same => n,Hangup()

When I’ve call the extension 102 from 101, Calls being connected to AudioSocket application correctly and I can see the audio data in the application.

But at the second dialplan, when I’ve called the trunk from my cellphone, It connects me to the AudioSocket application but audios are not sending the audiosocket appilcation. When I tracked the call in sngrep, no rtp is starting. What is the reason of that and what should I do?

2

Is Asterisk behind NAT, and the provider isn’t? Has Asterisk been configured to know it is behind NAT, and ports forwarded?

3

Yes, Asterisk is behind the NAT. But, I don’t know what should I do for this. Also, I’ve forgot the mention about that I’ve tried to use trunk with this dialplan;

exten => s,1,Verbose("Call to AudioSocket via Dialplan Application")
 same => n,Answer()
 same => n,Dial(SIP/101)
 same => n,Hangup()

This dialplan worked and I made the call successfully. There were no rtp problems at all. The problem is only exists when I’ve tried to use audiosocket application with the incoming call.

4

The phone is probably sending media, which will allow the provider to make use of symmetric media to compensate for the invalid media address in the SDP. I suspect your audio socket is one way, so there is no outgoing media to the provider.

Also note that the source code for chan_sip has been removed in the latest version of Asterisk, and it wasn’t being supported, even before then.

Reading trunk to mean provider.

5

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