This is a call coming in through the ISDN to 7940's. Answering with non-codec capability 1 - Is that the problem?
SIP Debugging Enabled We're at 10.1.0.11 port 18406 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 <<<<<<------------- 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as03605c88 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 16 Apr 2004 19:21:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 258 v=0 o=root 14316 14316 IN IP4 10.1.0.11 =sessionI> c=IN IP4 10.1.0.11 t=0 0 m=audio 18406 RTP/AVP 8 0 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.119:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as03605c88 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0 9 headers, 0 lines pbx01*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as03605c88 To: <sip:[EMAIL PROTECTED]>;tag=000e3857223c0238011930bc-566c64f8 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0 9 headers, 0 lines We're at 10.1.0.11 port 18198 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as286f917d To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 16 Apr 2004 19:21:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 258 v=0 o=root 14316 14316 IN IP4 10.1.0.11 =sessionI> c=IN IP4 10.1.0.11 t=0 0 m=audio 18198 RTP/AVP 8 0 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.120:5060 pbx01*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as286f917d To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as286f917d To: <sip:[EMAIL PROTECTED]>;tag=000f23ad6e25021c1c1f7e2d-532b7f03 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0 9 headers, 0 lines We're at 10.1.0.11 port 29654 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as19596a6b To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 16 Apr 2004 19:21:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 258 v=0 o=root 14316 14316 IN IP4 10.1.0.11 =sessionI> sip debug c=IN IP4 10.1.0.11 t=0 0 m=audio 29654 RTP/AVP 8 0 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.125:5060 pbx01*CLI> sip debug Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as19596a6b To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0 9 headers, 0 lines pbx01*CLI> sip debug Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as19596a6b To: <sip:[EMAIL PROTECTED]>;tag=000f23ac489f00c519541b4d-016de7d7 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0 -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tracy R Reed Sent: 16 April 2004 19:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly: > When we receive or make a call to the outside - they can hear us, but we > cant hear them. I have had this problem several times and so far no resolution. However for me it has always been with IAX. I have been told that IAX is supposed to be NAT-safe but that does not seem to be the case for me. For example: SIP (grandstream, snom) ->Asterisk->NAT->Asterisk->SIP (grandstream, snom) He can hear me but I can't hear him. In another case I had: IAXclient (soft phone)->NAT->Asterisk->Snom And I could hear him but he could not hear me. Same phone system and settings as above. However as soon as I switched the first users phone to talk directly to my Asterisk box with SIP it worked perfectly. And when I switched the user in the second case to a SIP based soft phone it also worked just fine. SIP has worked better through NAT than IAX (with nat=yes in sip.conf) which is bizarre and contrary to what I have read where IAX should be NAT-safe and SIP not. I have dreams of a world fully converted to IPv6 where NAT no longer exists. Alas, it is but a dream. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
