[Asterisk-Users] cisco/asterisk interop issues?

[James Burke]

hi,

i have an issue that when making a call from a SIP phone going as follows:

phone --> asterisk --> cisco(192.168.0.1) --> terminating voip platform(10.0.0.1)

i get the cisco sending up an invite to the voip platform followed 
directly with a CANCEL message, as follows:

Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKF325E4
Remote-Party-ID: "device" 
<sip:200 at 192.168.0.1>;party=calling;screen=no;privacy=off
From: "device" <sip:200 at 192.168.0.1>;tag=B2A336CC-413
To: <sip:5551234567 at 10.0.0.1>
Date: Thu, 05 Jan 2006 15:09:08 GMT
Call-ID: D8E85DC-7D3411DA-BC0AE3D2-F59304C1 at 192.168.0.1
Supported: 100rel,timer,resource-priority
Min-SE:  1800
Cisco-Guid: 227404060-2100564442-3154699218-4120052929
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REG
ISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1136473748
Contact: <sip:200 at 192.168.0.1:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 285


Jan  5 15:09:10.642: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:5551234567 at 10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKF325E4
From: "device" <sip:200 at 192.168.0.1>;tag=B2A336CC-413
to: <sip:5551234567 at 10.0.0.1>
Date: Thu, 05 Jan 2006 15:09:08 GMT
Call-ID: D8E85DC-7D3411DA-BC0AE3D2-F59304C1 at 192.168.0.1
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1136473750
Reason: Q.850;cause=0
Content-Length: 0

the asterisk reports the following:

     -- Executing Dial("SIP/200-c5c4", "SIP/5551234567 at 192.168.0.1") in new stack
     -- Called 5551234567 at 192.168.0.1
     -- SIP/192.168.0.1-a928 is making progress passing it to SIP/200-c5c4
     -- Got SIP response 500 "Internal Server Error" back from 192.168.0.1
     -- SIP/192.168.0.1-a928 is circuit-busy
   == Everyone is busy/congested at this time (1:0/1/0)

if i send it as follows:

phone --> asterisk --> cisco(192.168.0.1) --> pstn

all is good and call is processed normally.

any help would be appreciated..