Solved: Cisco IOS 15 Cube Gateway and SKype SIP Proxy issue

charles-moore · ‎10-08-2012

Hi All

I currently have an issue with making outgoing calls via SIP Skype ITSP.

I have a working SIP ITSP but I would like to get familiar with SKYPE. The SIP-UA with SKYPE is registered and can bee seen as registration status ok when I browse to the SKYPE Manager web page. Incoming calls are successful however outgoing calls fail with the following error detailed below.

The disconnect Cause code found in the VOIP CCAPI trace, relates to SIP Proxy Authentication error, found within the CCSIP trace Messages

during the setup of the call.

Disconnect Cause (CC) : 47

Disconnect Cause (SIP) : 407

Received:

SIP/2.0 407 Proxy Authentication Required

From: <sip:9905XXXXXXXXXX@76.63.XX.01>;tag=37D7EF0-13F3

To: <sip:44789xxxxxxx@sip.skype.com>;tag=44da78c1-13c4-5071c43f-e8a4c7dd-2087e3ba

Call-ID: 506A1FDE-FE011E2-B83AD438-7F20DAC@76.63.XX.01

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="76.63.XX.01", nonce="5071c45d00012d3e6e3d6c88ebd5ca4ab8e97b136a45bbee", algorithm=MD5

Via: SIP/2.0/UDP 76.63.XX.01:5060;branch=z9hG4bK9137C86

Content-Length: 0

The fault is active on the IOS 15, if I roll back to IOS 12, the outgoing SIP calls are successful.

Please be advised Cisco have introduced the following with the IOS 15 Train Multi-registrar’s url below:

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-multi-registrars.html.

This allows users to have a choice of several SIP ITSP connected to their CUBE. I have tried several IOS 15 train all with the same results, when initiating a call with Skype.

With reference to the issue I’m currently facing. I have a SIP Dial-peer from HQ and an E-Phone currently configured on HQ.

Incoming calls is reaching the E-Phone successfully, however outgoing calls are failing. The Outside facing interface on the HQ Router is configured for NAT. I have decided to configure the SKYPE SIP configuration on my Voice Router GRY9, and connected to the HQ Router. I have completed the same configuration for the Remote site Router CUBE. Both of these Voice Gateways, incoming and outgoing calls to Skype are successful.

The only difference in the configuration relates to the HQ Perimeter facing Router which is configured with NAT, although I dont believe this to be an issue.

I run a debug CCSIP Call, you will see in the setup message “State Dead” and the Source Address is My Perimeter Public facing IP address.

If I debug the GRY9 and the CUBE voice gateways you see the internal IP address associated to the Voice Router interface.

The IP address subnet range for the routers are actually configured Subnet’s (Subinterfaces ) “Nat Inside” off the HQ. perimeter router.

During the SIP Call setup process i.e. "Sent invite", "Received Trying" The Proxy-Authenticate Digest realm, fails to populate the SIP ITSP DNS name. All that is populated is the IP address.

i.e. “76.XX.XX.01@sip.skype.com” or “76.xx.xx.01@sipgate.com”

The SIP-UA, Authentication username and password are correctly configured not sure how to resolve the issue.

Please review attached debug documentaion which highlights a successful call via two SIP ITSP on a 12T train an the failed call to Skype on the 15T.

Please note the second ITSP calls is successful on the 15 Train, the issue relates to SKYPE calls only.

brmeade · ‎10-08-2012

You can use a SIP profile to modify the From field to the proper domain.

voice class sip-profiles 100

 request INVITE sip-header From modify "<>" "<>"

voice service voip
 sip
  sip-profiles 100

View solution in original post

brmeade · ‎10-08-2012

Charles,

You can also apply this under the specific outbound voip dial-peer so you can just apply this to your SIP dial-peer.

dial-peer voice 555 voip
   voice-class sip-profiles 100

Brian

View solution in original post

brmeade · ‎10-08-2012

You can use a SIP profile to modify the From field to the proper domain.

voice class sip-profiles 100

 request INVITE sip-header From modify "<>" "<>"

voice service voip
 sip
  sip-profiles 100

charles-moore · ‎10-08-2012

Hi Brian,

thanks for the quick response, I have a query with reference to the SIP Profile the setting

I assume relates to the request INVITE sipheader modify 
"<>". I'm currently working with two SIP ITSp, would this mean all SIP headers will be modified.
If this is the case can I be more specific i.e.

voice class sip-profiles 100
 request INVITE sip-header From modify "<>" "<9905XXXXXXXXXX>

 Regards  Charles

charles-moore · ‎10-08-2012

Hi Brian,

thanks for the quick response, I have a query with reference to the SIP Profile the setting

I assume relates to the request INVITE sipheader modify

voice class sip-profiles 100

request INVITE sip-header From modify "<>" "<>9905XXXXXXXXXX@sip.skype.com>"

brmeade · ‎10-08-2012

Charles,

You can also apply this under the specific outbound voip dial-peer so you can just apply this to your SIP dial-peer.

dial-peer voice 555 voip
   voice-class sip-profiles 100

Brian

charles-moore · ‎10-08-2012

Hi Brian,

Thanks the issue is now resolved, I have placed the following under the Global Voice class profiles:

voice class sip-profiles 9905

request INVITE sip-header From modify "<>" "<>9905xxxxxxxxxx@sip.skype.com>"

Cheers

Charles

brmeade · ‎10-08-2012

Great to hear!

armin.wagenfuehr · ‎12-18-2013

Hi, I have nearly the same issue. For outgoing I must use also the Username in the outgoing invite. Works fine the Profile in the top of this thread. Thanks.

When we get a imcoming invite from the SIP Provider we get also all the time in the Invite Header the Username for registration. The customer have ddid from 00-29 at sip trunk. So how we could differ between this numbers, because we get all the time the username.

I wounder me why I get all the time fast busy and no dial-peer are match. I done debug voice dial-peer and had saw:

549051: Dec 18 18:27:54.702: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=D200010111100, Called Number=D200010111100, Peer Info Type=DIALPEER_INFO_SPEECH

The from and to address in the invite is correct. So, I have a change that the VGW looks at the "to" address instead of the invite header?

Have you an Idea how I could solve this issue in VGW?

Received:

INVITE sip:D2000101000@213.777.22.11:5060 SIP/2.0

Record-Route: <213.218.12.2>

Record-Route: <213.218.28.100>

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bKbade.0e72b251.0

Via: SIP/2.0/UDP 213.218.28.100;branch=z9hG4bKbade.3d2d6e45.0

Via: SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C2FD9043ABA97588

From: "+49123456789" <>+49123456789@voice.de;user=phone>;tag=00E0F5100556C2FD9043259E617A

To: <>+49987654321@voice.de>

Call-ID: e410e00004f5-52b1d8f9-472e60b7-33428f00-2e52e84@127.0.0.1

CSeq: 17749 INVITE

Contact: <>

Max-Forwards: 68

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE

Content-Type: application/sdp

Supported: 100rel, timer, replaces

P-Asserted-Identity: <>+4971919004598@toplink-voice.de>

Remote-Party-ID: <>+4971919004598@toplink-voice.de>;party=calling;screen=no;privacy=off

Content-Length: 441

v=0

o=- 3616975336 0 IN IP4 213.218.12.2

s=session

c=IN IP4 213.218.12.2

t=0 0

m=audio 12340 RTP/AVP 8 0 18 2 96

c=IN IP4 213.218.12.2

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:2 G726-32/8000

a=sendrecv

a=rtpmap:96 telephone-event/8000

m=image 12342 udptl t38

c=IN IP4 213.218.12.2

a=T38FaxVersion:0

a=T38MaxBitRate:14400

a=T38FaxUdpEC:t38UDPRedundancy

a=T38FaxRateManagement:transferredTCF

549047: Dec 18 18:18:51.204: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bKbade.0e72b251.0,SIP/2.0/UDP 213.218.28.100;branch=z9hG4bKbade.3d2d6e45.0,SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C2FD9043ABA97588

From: "+49123456789" <>+49123456789@voice.de;user=phone>;tag=00E0F5100556C2FD9043259E617A

To: <>+49987654321@voice.de>

Date: Wed, 18 Dec 2013 17:18:51 GMT

Call-ID: e410e00004f5-52b1d8f9-472e60b7-33428f00-2e52e84@127.0.0.1

CSeq: 17749 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

549048: Dec 18 18:18:51.240: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bKbade.0e72b251.0,SIP/2.0/UDP 213.218.28.100;branch=z9hG4bKbade.3d2d6e45.0,SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C2FD9043ABA97588

From: "+49123456789" <>+49123456789@voice.de;user=phone>;tag=00E0F5100556C2FD9043259E617A

To: <>+49987654321@voice.de>

Date: Wed, 18 Dec 2013 17:18:51 GMT

Call-ID: e410e00004f5-52b1d8f9-472e60b7-33428f00-2e52e84@127.0.0.1

CSeq: 17749 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

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HTH, please rate all useful posts and right answers.