[asterisk-users] WaitExten() always times out
kathrynster at gmail.com
Tue Aug 24 15:07:56 CDT 2010
As far as I can tell Asterisk recives media perfectly. For outgoing calls it
looks something like this:
-- Executing [xxx at proxy:5] WaitExten("SIP/voiptrunk-00000083", "7") in
DEBUG: rtp.c:1032 process_rfc2833: - RTP 2833 Event: 00000001 (len =
DEBUG: rtp.c:880 send_dtmf: Sending dtmf: 49 (1), at xx.xx.xxx.x
On incoming, as far as I can tell, Asterisk does not recieve anything. I
just don't know why.
I have added exceptions in firewall and network to allow voip traffic,
successfully allowing incoming and outgoing calls. Just no DTMF on incoming
My tests consist of a regular landline, I dial a DID and successfully reach
my asterisk box. Everything is fine until I come to user input. None is
recognized. I get a "-User entered nothing" and timeout.
On Mon, Aug 23, 2010 at 8:07 AM, Miguel Molina <mmolina at millenium.com.co>wrote:
> 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: 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
> Ing. Miguel Molina
> Grupo de Tecnología
> Millenium Phone Center
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