[asterisk-users] WaitExten() always times out

[Miguel Molina]

El 20/08/10 16:14, Kathryn Jones escribió:
> Thanks for all the help, but I still can't find what's wrong.
>
> I enabled console => notice,warning,error,debug,dtmf like Miguel 
> suggested. The output is attached.
>
> I noticed that the rtp.c session never starts, which as I understand 
> is what catches the dtmf tone, but I could not find how to start it :s.
>
> The Answer() and waitExten(5,m) didn't fix my problem. I hope someone 
> can help me see the problem after looking at the attached console output.
The following line brought my attention:

[Aug 20 16:50:04] DEBUG[5319]: channel.c:1882 __ast_answer: Didn't receive a media frame from SIP/xx.xx.xxx.xx-00000026 within 500 ms of answering. Continuing anyway



Are your sure that RTP audio (media) is correctly received in asterisk? 
I suspect network or firewall problems. Also, you said that you were 
going to receive calls from the PSTN, but are you testing from a SIP 
endpoint?

Regards,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center