It's a nortel phone switch (ie: phone company), not a nortel pbx.
On Wed, Nov 28, 2007 at 08:55:09PM -0600, Jonn R Taylor wrote: > What LAN and you using? ELAN or HSP Are you trying to connect to a signaling > server? Please provide Nortel config. > > Jonn > > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL > PROTECTED] > Sent: Wednesday, November 28, 2007 2:06 PM > To: [email protected] > Subject: [asterisk-users] Asterisk <-> Nortel Phone Switch > > 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 > > > > > _______________________________________________ > --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
