Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k).
Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no canreinvite=no dtmfmode=rfc2833 fromuser=user username=user secret=123 disallow=all allow=gsm allow=ulaw allow=alaw dtmfmode=rfc2833 usereqphone=yes context=from-nortel asterisk*CLI> sip debug ip 10.0.0.10 SIP Debugging Enabled for IP: 10.0.0.10 The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. Audio is at 192.168.10.2 port 17492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.10:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: "Shawn Ip" <sip:[EMAIL PROTECTED]>;tag=as25dd7670 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Nov 2007 18:24:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI> <--- SIP read from 10.0.0.10:5060 ---> SIP/2.0 486 Busy Here From: "Shawn Ip"<sip:[EMAIL PROTECTED]>;tag=as25dd7670 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.10.2:5060;rport=5060;branch=z9hG4bK489024dd User-Agent: Asterisk PBX Max-Forwards: 70 Supported: replaces Date: Wed, 28 Nov 2007 18:24:14 GMT Allow: NOTIFY Content-Type: application/SDP Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Transmitting (no NAT) to 10.0.0.10:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: "Shawn Ip" <sip:[EMAIL PROTECTED]>;tag=as25dd7670 o: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI> <--- SIP read from 10.0.0.10:5060 ---> SIP/2.0 486 Busy Here From: "Shawn Ip"<sip:[EMAIL PROTECTED]>;tag=as25dd7670 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.10.2:5060;rport=5060;branch=z9hG4bK489024dd User-Agent: Asterisk PBX Max-Forwards: 70 Supported: replaces Date: Wed, 28 Nov 2007 18:24:14 GMT Allow: NOTIFY Content-Type: application/SDP Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Transmitting (no NAT) to 10.0.0.10:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: "Shawn Ip" <sip:[EMAIL PROTECTED]>;tag=as25dd7670 o: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Transmitting (no NAT) to 10.0.0.10:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: "Shawn Ip" <sip:[EMAIL PROTECTED]>;tag=as25dd7670 o: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users .
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