Whilst I've been messing about with registering a Grandstream GXP2000 to our CME 4.1 server's SIP service, I've found quite an annoying problem.
The phone registers fine and can make external calls via the SIP trunk absolutely fine. The phone will also accept transfers of internal calls (ephones), <strong>but</strong> if I attempt to transfer an external call form an ephone, to the Granstream - the external call is cut-off. I've isolated the 'debug ccsip messages' output which describes what is happening, but I'm by no means an expert in debugging SIP output.
Aug 14 09:16:49.807: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REFER sip:07xxxxxxxxx@cme-router:5060 SIP/2.0
Via: SIP/2.0/UDP cme-router:5060;branch=z9hG4bK3432633
From: <sip:0116@cme-router>;tag=2CD733C8-216A
To: "Tom" <sip:07xxxxxxxxx@cme-router>;tag=as221e2abe
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 102 REFER
Max-Forwards: 70
Contact: <sip:0116@cme-router:5060>
User-Agent: Cisco-SIPGateway/IOS-12.x
Timestamp: 1187083009
Refer-To: sip:2007@cme-router?Replaces=E72682C6-497D11DC-90D890FC-79F613F8%40cme-router%3Bto-tag%3Ddb0457357be0d469%3Bfrom-tag%3D2CD77068-25C3
Referred-By: <sip:0116@cme-router>
Content-Length: 0
Aug 14 09:16:49.811: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP cme-router:5060;branch=z9hG4bK3432633;received=cme-router
From: <sip:0116@cme-router>;tag=2CD733C8-216A
To: "Tom" <sip:07xxxxxxxxx@cme-router>;tag=as221e2abe
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 102 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07xxxxxxxxx@asterisk-sip-gateway>
Content-Length: 0
Aug 14 09:16:49.811: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
NOTIFY sip:0116@cme-router:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-sip-gateway:5060;branch=z9hG4bK49b17aad;rport
From: "Tom" <sip:07xxxxxxxxx@asterisk-sip-gateway>;tag=as221e2abe
To: <sip:0116@cme-router>;tag=2CD733C8-216A
Contact: <sip:07xxxxxxxxx@asterisk-sip-gateway>
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=102
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 49
SIP/2.0 481 Call leg/transaction does not exist
Aug 14 09:16:49.815: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP asterisk-sip-gateway:5060;branch=z9hG4bK49b17aad;rport
From: "Tom" <sip:07xxxxxxxxx@asterisk-sip-gateway>;tag=as221e2abe
To: <sip:0116@cme-router>;tag=2CD733C8-216A
Date: Tue, 14 Aug 2007 09:16:49 GMT
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 104 NOTIFY
Content-Length: 0
Aug 14 09:16:49.843: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:0116@cme-router:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-sip-gateway:5060;branch=z9hG4bK136ae0fa;rport
From: "Tom" <sip:07xxxxxxxxx@asterisk-sip-gateway>;tag=as221e2abe
To: <sip:0116@cme-router>;tag=2CD733C8-216A
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Where 'cme-router' is the router on which CME 4.1 is installed and running, and 'asterisk-sip-gateway' is our SIP gateway (which the CME 4.1 SIP-UA connects to) that handles more advanced call routing features.
If anyone has any ideas - I'd appreciate the help! It's not a massive problem, but I'm keen to see this one through. I should be able to upgrade to CME 4.2 at some point this week, so we'll have to see if that helps matters.
