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Call forward disconnects after 30 seconds

I have a UC540 with a SIP trunk. The incoming calls are answered by CUE, the caller hears a greeting message and then is connected to the operator.

If the operator activate the CFA, the calls are forwarded to her mobile but the call is disconnected after 30 seconds.

If I program the incoming calls to go directly to operator phone (without greeting message on CUE) then there is no problem with CFA.

The Provider of SIP trunk make his tests and it says that the UC540 disconnects the call. I think they have right because the UC540 sends first the BYE message. See below debug ccsip messages:

000705: Sep 15 14:33:29.860: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:25811102@217.27.60.210:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.16.145:5060;branch=z9hG4bK5EE12C4
From: 35722100444 <sip:35722100444@217.27.60.204;user=phone>;tag=E27CCBC0-20C2
To: 25811102 <sip:25811102@sip.thunderworx.net;user=phone>;tag=103465381
Date: Wed, 15 Sep 2010 14:32:57 GMT
Call-ID: 3623d547-7609173d-129733c9-cbf@sip.thunderworx.net
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Route: <sip:217.27.60.204;lr=on;ftag=103465381;nat=yes>
Timestamp: 1284561209
CSeq: 102 BYE
Reason: Q.850;cause=86
Content-Length: 0

10 Replies 10

antyeung
Level 4
Level 4

Can you get the sip trace for the entire duration of the call?

Reason: Q.850;cause=86

Call having the requested call identity has been cleared

Typical scenarios include:

Network timeout

Call cleared by remote user.

86

Indicates that the network has received a call identity information  element indicating a suspended call that has in the meantime been  cleared wile suspended.

Thanks for your reply. I am attaching you the sip trace for the entire duration of the call.

On the INVITE that is sent out to the CFA destination the UC500 includes the following header indicating support for session refresh-

002032: Sep 17 04:09:19.189: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:99676711@217.27.60.204:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.16.145:5060;branch=z9hG4bK6FB40D
Remote-Party-ID: "24367040 " <24367040>;party=calling;screen=no;privacy=off
From: "24367040 " <24367040>;tag=EA8EC9EC-1D0F
To: <99676711>
Date: Fri, 17 Sep 2010 04:09:19 GMT
Call-ID: 2DA2D964-C14811DF-9AB3F1D6-E171076B@172.21.16.145

...   

Content-Disposition: session;handling=required

The 200 OK that we receive from the provider indicating that the far-end answered specifies the session refresher parameters where the UAC (UC500) should be the session refresher-

002035: Sep 17 04:09:32.394: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 172.21.16.145:5060;rport=58590;received=172.21.16.145;branch=z9hG4bK6FB40D
From: "24367040 " <24367040>;tag=EA8EC9EC-1D0F
To: <99676711>;tag=547697925
Call-ID: 2DA2D964-C14811DF-9AB3F1D6-E171076B@172.21.16.145
CSeq: 101 INVITE
Contact:  <>095935799676711@sip.thunderworx.net:5060>
Record-Route: <217.27.60.204>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
Supported: timer
Session-Expires: 86400;refresher=uac

...

We later then send a REINVITE to refresh the session-

002037: Sep 17 04:09:32.418: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:24367040@217.27.60.210:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.16.145:5060;branch=z9hG4bK6FD25E7
Remote-Party-ID: <401>;party=calling;screen=no;privacy=off
From: 35722100444 <35722100444>;tag=EA8EA1D0-9DE
To: 24367040 <>24367040@sip.thunderworx.net;user=phone>;tag=906549646
Date: Fri, 17 Sep 2010 04:09:32 GMT

...

But we don't get a response so we disconnect the call-

002039: Sep 17 04:10:04.451: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:24367040@217.27.60.210:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.16.145:5060;branch=z9hG4bK6FE1D06
From: 35722100444 <35722100444>;tag=EA8EA1D0-9DE
To: 24367040 <>24367040@sip.thunderworx.net;user=phone>;tag=906549646
Date: Fri, 17 Sep 2010 04:09:32 GMT
Call-ID: 6cb34611-63ad8441-54eea1-561a@sip.thunderworx.net


We have two options to fix this-


1. Figure out why the provider is not responding to the reINVITE to refresh the session. Maybe they don't support the reINVITE method?

2. Try to find a way for the UC500 to not advertise session handling as a requirement.

For (2), can you send attach the running configurations excluding passwords and public IP addresses?

-Anthony

Thank you very much Anthony for your help.

I will speak with the provider for the reINVITE.

In the case they do not support it, I am attaching you the configuration of UC500.

Sorry, I misquoted. "Content-Disposition: session;handling=required" doesn't refer to session refresh but to the SDP. The "timer" timer extension however is what we want - " Supported: 100rel,timer,resource-priority,replaces,sdp-anat". Are you using CCA to configure the UC500?

No, I am using CLI to configure the UC500.

Hi Anthony,

I spoke with my provider and we find out that the problem is that their softswitch does not support re-invite messages. As shown below UC500 sends a re-invite message for the first incoming leg and does not get a reply.

Do you know how can I disable the re-invite messages from UC500 using CLI?

002037: Sep 17 04:09:32.418: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:24367040@217.27.60.210:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.16.145:5060;branch=z9hG4bK6FD25E7
Remote-Party-ID: <401>;party=calling;screen=no;privacy=off
From: 35722100444 <35722100444>;tag=EA8EA1D0-9DE
To: 24367040 <>24367040@sip.thunderworx.net;user=phone>;tag=906549646
Date: Fri, 17 Sep 2010 04:09:32 GMT
Call-ID: 6cb34611-63ad8441-54eea1-561a@sip.thunderworx.net
Route: <217.27.60.204>
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0662402989-3242725855-2594763222-3782281067
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1284696572
Contact: <22100444>
Expires: 180
Allow-Events: telephone-event
Session-Expires:  86400;refresher=uac
Content-Type: application/sdp
Content-Length: 244

Can you try using sip-profiles-

voice class sip-profiles 1
request INVITE sip-header Supported remove

and apply this to the outgoing sip dial-peer

I configure a sip-profile but it still sending the reinvite message.

Is it sending the "Supported: timer" header?

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