08-14-2007 02:49 AM - edited 03-14-2019 11:03 PM
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.
08-14-2007 03:15 AM
Hi, not sure it that is you case, but try:
voice service voip
no supplementary-service sip moved-temporarily
Let us know if it works...
08-14-2007 06:26 AM
Hi, unfortunately that hasn't helped :(
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
!
The voice class is g711alaw, followed by g711ulaw.
I'm not sure what the 'refer target dial-peer' does :/
I'm now running CME 4.2, with 12.4(11)XW2.
08-14-2007 06:44 AM
Hi,
the thing is that cisco doesn't officially support third-party SIP phones for CME so when something doesn't work you're on your own.
Good luck!
08-17-2007 02:18 AM
Well, maybe someone who isn't so Cisco-centric may wish to help in this instance :)
I can't be the only person with these troubles, considering SIP is /meant/ to be a standard.
12-06-2007 07:42 PM
Hi, can someone help on this. I am having the same issue. A SIP phone can transfer to a skinny phone, but a skinny phone cannot transfer a call to a sip phone. however skinny phone can call directly no problem.
The IOS I am using is 12.4(15)T1.
12-18-2011 05:30 AM
Hi
I have come across similar issue Cisco 8961 (SIP phone) fails to transfer an external caller to a Cisco SCCP phone; I get the following message:
subscription-state terminated reason noresource
Does anybody know what the fix is?
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide