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

[Jonn R Taylor]

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