[asterisk-users] Asterisk <-> Nortel Phone Switch

[shawnl at up.net]

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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of shawnl at up.net
> Sent: Wednesday, November 28, 2007 2:06 PM
> To: asterisk-users at lists.digium.com
> 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:5551212 at 10.0.0.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport
> From: "Shawn Ip" <sip:user at 192.168.10.2>;tag=as25dd7670
> To: <sip:3538379 at 10.0.0.10>
> Contact: <sip:user at 192.168.10.2>
> Call-ID: 0a72cba50ae922e206a7f65c69bc4670 at 192.168.10.2
> 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:user at 192.168.10.2>;tag=as25dd7670
> To: <sip:5551212 at 10.0.0.10>
> Call-ID: 0a72cba50ae922e206a7f65c69bc4670 at 192.168.10.2
> 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:3538379 at 10.0.0.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport
> From: "Shawn Ip" <sip:user at 192.168.10.2>;tag=as25dd7670
> o: <sip:5551212 at 10.0.0.10>
> Contact: <sip:user at 192.168.10.2>
> Call-ID: 0a72cba50ae922e206a7f65c69bc4670 at 192.168.10.2
> 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