[Asterisk-Users] connecting to nortel CS1000 (half way there)

Jerry Geis geisj at pagestation.com
Thu Jun 2 07:32:30 MST 2005


I am connecting to a Nortel CS 1000. I can place calls out to an extension
so we are half way there. When calling into the box I get the following from
sip debug ip X.

I get dead air when calling into the box.

In my sip.conf I have a context of nortel and in extensions.conf the nortel
context just has a s,1,Playback(demo-congrats).

Any suggestions as to why the call in might not be working but the
call out works just fine is appreciated.

Jerry

------------------------------------------

*CLI>
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.168.45.194 SIP/2.0
Via: SIP/2.0/UDP 161.49.198.102:5060;branch=z9hG4bK6a2f6214
From: "asterisk" <sip:asterisk at 161.49.198.102>;tag=as4c29ec16
To: <sip:192.168.45.194>
Contact: <sip:asterisk at 161.49.198.102>
Call-ID: 2a67a2493820c3c153d0b87c7a97ceca at 161.49.198.102
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 02 Jun 2005 14:02:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 192.168.45.194:5060


Sip read:
SIP/2.0 200 OK
From: "asterisk"<sip:asterisk at 161.49.198.102>;tag=as4c29ec16
To: <sip:192.168.45.194>;tag=c22da8c0-13c4-429ecb30-59f22c0-57e8
Call-ID: 2a67a2493820c3c153d0b87c7a97ceca at 161.49.198.102
CSeq: 102 OPTIONS
Allow: 
INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Via: SIP/2.0/UDP 161.49.198.102:5060;branch=z9hG4bK6a2f6214
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW: release=4.0 version=sse-4.00.31
Content-Length: 0


10 headers, 0 lines
Destroying call '2a67a2493820c3c153d0b87c7a97ceca at 161.49.198.102'



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