Sip trunk to Skype problems since 5 May on UC560

Unanswered Question
May 17th, 2010


Skype made some changes to their platform a few weeks ago and we are not able to make outbound calls anymore.

Before the change we could make calls using the following FROM field in the sip header:

From: "Admin" <sip:[email protected].4> where is the WAN IP of our UC560

It worked perfectly for a few months until 5 May. Then is stopped. We contacted skype and they said that we should change the from field to:

From: "Admin" <sip:[email protected]>

how do I change sip header in UC500? Sounds like a trivial thing but can't seem to find a way

I would appreciate any useful advice. Skype says that they support cisco uc500 but their tech support was useless. They don't know themselves how to change it.

I have this problem too.
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Marcos Hernandez Tue, 05/18/2010 - 09:38

Do this:

voice service voip



Let me know how it goes.


inventica Tue, 05/18/2010 - 09:52

thank you Marcos, but I'm afraid that change didn't make any difference

000402: May 18 16:45:47.777: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK7E1D7FD
From: "Inventica Technologies" [email protected].45>;tag=1092E17C-1F72
To: [email protected]>
Date: Tue, 18 May 2010 16:45:47 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2738975647-1641222623-2503529596-0315902310
User-Agent: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1274201147
Contact: [email protected].45:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 286

Marcos Hernandez Tue, 05/18/2010 - 10:00

Hmm, OK. Here something else you can try:

Address of Record (AOR) usage over SIP trunk

What are the SP UNI requirements for AOR over SIP trunk?

Most SP UNI mandate that the host portion in the below headers of SIP messages represent the domain name of the proxy for authentication purposes

  • From header in INVITE message
  • Contact header in 3xx message
  • Refer-To & Referred-By headers in REFER message

How can CME send the right AOR over the SIP trunk?

This behavior can be achieved using the "host-registrar" CLI under sip-ua - the host portion will now reflect the domain name configured via the registrar-server command

If no registrar server command is present, then the "localhost" CLI will apply

voice service voip

inventica Tue, 05/18/2010 - 10:28

I've just tried


but it doesn't recognize registrar-server command

inventica Tue, 05/18/2010 - 10:41

no, there is no such command

currently I have:

registrar 1 expires 300
registrar 2 expires 300

Marcos Hernandez Tue, 05/18/2010 - 10:44

The registrar commands you have are the ones that I am talking about. Do you still see the IP in the From header?

Marcos Hernandez Tue, 05/18/2010 - 11:05

OK, try one more thing. Under the voip dial-peer (or peers) pointing to Skype, try this:

voice-class sip localhost


inventica Wed, 05/19/2010 - 04:06

Hi Marcos,

I'm afraid that didn't make any difference either

here is my dial peer

dial-peer voice 5023 voip
description Skype Out
translation-profile outgoing Skype-out2
destination-pattern 900...........
session protocol sipv2
session target
session transport udp
voice-class codec 1
voice-class sip localhost
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad

and here is the debug extract

000419: May 19 10:59:27.679: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
SIP/2.0 403 Forbidden
From: "Inventica" [email protected]>;tag=147C218C-20E0
To: [email protected]>;tag=ca90d13f-13c4-4bf3c48f-673ace76-2035561b
Call-ID: [email protected]
CSeq: 101 INVITE

Marcos Hernandez Wed, 05/19/2010 - 06:46

One of these methods should have worked. You will need to open a TAC case so we can determine what in your configuration is preventing the name from being added to the From: field.



inventica Thu, 05/20/2010 - 03:03

Solution found!

I had to add

voice-class sip localhost preffered

to dial peers. You were very close Marcos. It didn't work without preffered for some reason

Marcos Hernandez Thu, 05/20/2010 - 03:15

True, this is needed when multiple registrars are configured. Sorry for missing that.

Usage Guidelines

Use the voice-class sip localhost command in dial peer voice configuration mode to override the global configuration on Cisco IOS voice gateways, Cisco UBEs, or Cisco Unified CME and configure a DNS localhost name to be used in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages on a specific dial peer. When multiple registrars are configured for an individual dial peer you can then use the voice-class sip localhost preferred command to specify which host is preferred for that dial peer.


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