Bad SIP Transfers in CME 4.1

Unanswered Question
Aug 14th, 2007
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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:[email protected]-router:5060 SIP/2.0

Via: SIP/2.0/UDP cme-router:5060;branch=z9hG4bK3432633

From: <sip:[email protected]-router>;tag=2CD733C8-216A

To: "Tom" <sip:[email protected]-router>;tag=as221e2abe

Call-ID: [email protected]-sip-gateway

CSeq: 102 REFER

Max-Forwards: 70

Contact: <sip:[email protected]-router:5060>

User-Agent: Cisco-SIPGateway/IOS-12.x

Timestamp: 1187083009

Refer-To: sip:[email protected][email protected]%3Bto-tag%3Ddb0457357be0d469%3Bfrom-tag%3D2CD77068-25C3

Referred-By: <sip:[email protected]-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:[email protected]-router>;tag=2CD733C8-216A

To: "Tom" <sip:[email protected]-router>;tag=as221e2abe

Call-ID: [email protected]-sip-gateway

CSeq: 102 REFER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:[email protected]-sip-gateway>

Content-Length: 0




Aug 14 09:16:49.811: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

NOTIFY sip:[email protected]-router:5060 SIP/2.0

Via: SIP/2.0/UDP asterisk-sip-gateway:5060;branch=z9hG4bK49b17aad;rport

From: "Tom" <sip:[email protected]-sip-gateway>;tag=as221e2abe

To: <sip:[email protected]-router>;tag=2CD733C8-216A

Contact: <sip:[email protected]-sip-gateway>

Call-ID: [email protected]-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:[email protected]-sip-gateway>;tag=as221e2abe

To: <sip:[email protected]-router>;tag=2CD733C8-216A

Date: Tue, 14 Aug 2007 09:16:49 GMT

Call-ID: [email protected]-sip-gateway

CSeq: 104 NOTIFY

Content-Length: 0




Aug 14 09:16:49.843: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:[email protected]-router:5060 SIP/2.0

Via: SIP/2.0/UDP asterisk-sip-gateway:5060;branch=z9hG4bK136ae0fa;rport

From: "Tom" <sip:[email protected]-sip-gateway>;tag=as221e2abe

To: <sip:[email protected]-router>;tag=2CD733C8-216A

Call-ID: [email protected]-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.

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paolo bevilacqua Tue, 08/14/2007 - 03:15
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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...



firebadger Tue, 08/14/2007 - 06:26
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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.

paolo bevilacqua Tue, 08/14/2007 - 06:44
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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!

firebadger Fri, 08/17/2007 - 02:18
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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.

jasondu88 Thu, 12/06/2007 - 19:42
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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.

gsidhu Sun, 12/18/2011 - 05:30
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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?

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