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
