As far as I can tell Asterisk recives media perfectly. For outgoing calls it looks something like this:
-- Executing [...@proxy:5] WaitExten("SIP/voiptrunk-00000083", "7") in new stack DEBUG[28557]: rtp.c:1032 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4) DEBUG[28557]: 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 calls. 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 <mmol...@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[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 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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