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), but 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: ;tag=2CD733C8-216A
To: "Tom" ;tag=as221e2abe
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 102 REFER
Max-Forwards: 70
Contact:
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:
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: ;tag=2CD733C8-216A
To: "Tom" ;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:
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" ;tag=as221e2abe
To: ;tag=2CD733C8-216A
Contact:
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" ;tag=as221e2abe
To: ;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" ;tag=as221e2abe
To: ;tag=2CD733C8-216A
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


My voice service voip config is as below:

voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
sip
registrar server expires max 240 min 60
no call service stop


And the Grandstream phone config is:

voice register dn 1
number 2007
allow watch
refer target dial-peer
mwi
!
voice register pool 1
id mac 000B.820D.0536
number 1 dn 1
template 1
dtmf-relay rtp-nte
voice-class codec 1
description Grandstream


Where ‘cme-router’ is the router on which CME 4.2 is installed and running, and ‘asterisk-sip-gateway’ is our SIP gateway (which the CME 4.2 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.