On Wed, 2009-09-02 at 21:31 +0000, Ott Rose wrote: > i have posted this before but was unable to resolve it. i have some > new info so i figured i would try again. the trace from bandwidth.com > are below. they are telling me that the ip that is bold should be our > ip not bandwidth.com. i have changed every setting that i can see and > nothing fixes this. Where would i change this at? they cannot tell me. > > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport > From:"8592192438"<sip:[email protected]>;tag=as0707d433 > To:<sip:[email protected]> > Contact:<sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Wed, 02 Sep 2009 21:10:39 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 412 > > v=0 > o=root 3831 3831 IN IP4 216.82.224.202 > s=session > c=IN IP4 216.82.224.202 > t=0 0 > m=audio 17050 RTP/AVP 0 8 3 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > m=video 12426 RTP/AVP 31 34 103 > a=rtpmap:31 H261/90000 > a=rtpmap:34 H263/90000 > a=rtpmap:103 h263-1998/90000 > a=sendrecv > <snip> I know very little about how ringing works but are they providing any kind of status information to you? Do you need to furnish the ring if they are not? It seems to me I saw quite a few articles about providing ring tone, what causes it to fail, and how to work around it. I assume you've searched for those already. Just a few thoughts - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 [email protected]
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