On Sat, 7 Jun 2003, shido wrote:
> This is the sip debug when the call went through........ > > Sip read: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Call-ID: [EMAIL PROTECTED] > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 157 > Content-Type: application/sdp > CSeq: 1 INVITE > From: <sip:[EMAIL PROTECTED]>;tag=402ada92-5 > To: <sip:[EMAIL PROTECTED]> > User-Agent: Quintum/1.0.0 > Via: SIP/2.0/UDP 64.42.218.146;branch=z9hG4bK-tenor-64.42.218.146-5 > Quintum: 0c01030b0239380501 > > v=0 > o=Quintum 4 4 IN IP4 64.42.218.146 > s=VoipCall > c=IN IP4 64.42.218.146 > t=0 0 > m=audio 10240 RTP/AVP 0 > c=IN IP4 64.42.218.146 > a=rtpmap:0 pcmu/8000/1 > > 11 headers, 8 lines > Using latest request as basis request > Sending to 64.42.218.146 : 5060 (non-NAT) > Capabilities: us - 4, them - 4, combined - 4 > Non-codec capabilities: us - 1, them - 0, combined - 0 > > Funnily enough I've been looking at the same problem. Will get a chance to look a bit more tomorrow. Steve _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
