[cisco-voip] Ciso IOS Gateway and SKype SIP configuration.

Gavin Henry ghenry at suretec.co.uk
Sun Oct 7 08:22:49 EDT 2012


What does a full SIP trace show with a normal SIP call vs the Skype one?

Does where Skype sees you registered from match where the outgoing call is
coming from?

Are you authenticating again when asked?

Thanks.

--
Kind Regards,

Gavin Henry.
Managing Director.

T +44 (0) 1224 279484
M +44 (0) 7930 323266
F +44 (0) 1224 824887
E ghenry at suretec.co.uk

Open Source. Open Solutions(tm).

http://www.suretecsystems.com/

Suretec Systems is a limited company registered in Scotland. Registered
number: SC258005. Registered office: 24 Cormack Park, Rothienorman,
Inverurie,
Aberdeenshire, AB51 8GL.

Subject to disclaimer at http://www.suretecgroup.com/disclaimer.html

Do you know we have our own VoIP provider called SureVoIP? See
http://www.surevoip.co.uk

On 7 Oct 2012, at 12:40, Charles <charles at moore1.net> wrote:

  Hi All



*Previous comment posted:*

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



I already 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.  I would be most grateful if someone could point me in the right
direction.



*Disconnect Cause (CC)    : 47*

*Disconnect Cause (SIP)   : 407***



*Updated comment*



I thought I would elaborate on the previous email content, 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,* (Configured in GNS3) and connected to the HQ Router.
 I have completed the same configuration for the Remote site Router *CUBE*.
Both of these Voice Gateways are configured with a Physical handset and
incoming and outgoing calls to Skype are successful. The only difference in
the configuration is the HQ is the Perimeter facing Router and is
configured for NAT. Therefore If 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 for the routers are actually configured
Subnet’s (Subinterfaces ) *“Nat Inside”* on HQ. perimeter router.



I hope this explains things a little better.





Regards





Charles



*The Call Setup Information is*:

Call Control Block (CCB) : 0x4BAB5128

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 9905XXXXXXXXXX

Called Number            : 44XXXXXXXXXX

Source IP Address (Sig  ): 7x.9x.1xx.xx

Destn SIP Req Addr:Port  : 193.120.218.68:5060

Destn SIP Resp Addr:Port : 193.120.218.68:5060

Destination Name         : sip.skype.com



VG-2811HQ(config-sip-ua)#

000466: Oct  6 16:57:34.548:
//1566/5DFA75F28122/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : No Codec

Negotiated Codec Bytes   : 0

Nego. Codec payload      : 255 (tx), 255 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 7x.9x.1xx.xx

Source IP Port    (Media): 17298

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0



000467: Oct  6 16:57:34.548: //1566/5DFA75F28122/SIP/Call/sipSPICallInfo:

*Disconnect Cause (CC)    : 47*

*Disconnect Cause (SIP)   : 407***









<image002.jpg>






 ------------------------------

*From:* cisco-voip-bounces at puck.nether.net [
mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether.net>]
*On Behalf Of *Charles
*Sent:* 06 October 2012 22:19
*To:* cisco-voip at puck.nether.net
*Subject:* [cisco-voip] Ciso IOS Gateway and SKype SIP configuration.



Hi All,



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



I already have a working SIP ITSP but I would like to get familiar with
SKYPE. The SIP-UA with SKYPE is registered and show 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.  I would be most grateful if someone could
point me in the right direction.



*Disconnect Cause (CC)    : 47*

*Disconnect Cause (SIP)   : 407***





VG-2811HQ(config)#do sh sip-ua register status

--------------------- Registrar-Index  1 ---------------------



Line                             peer       expires(sec) registered
P-Associ-URI

================================ ========== ============ ==========
============

9.*                              9150       55           no

9035152222                       20005      55           no

90800*                           9101       55           no

90[2-68].........                9100       23           no

90[7].........*                  950        149          no

911                              20001      55           no

9905??????????           20039      115          yes

999                              20002      55           no







*The Call Setup Information is*:

Call Control Block (CCB) : 0x4BAB5128

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 9905XXXXXXXXXX

Called Number            : 44XXXXXXXXXX

Source IP Address (Sig  ): 7x.9x.1xx.xx

Destn SIP Req Addr:Port  : 193.120.218.68:5060

Destn SIP Resp Addr:Port : 193.120.218.68:5060

Destination Name         : sip.skype.com



VG-2811HQ(config-sip-ua)#

000466: Oct  6 16:57:34.548:
//1566/5DFA75F28122/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : No Codec

Negotiated Codec Bytes   : 0

Nego. Codec payload      : 255 (tx), 255 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 7x.9x.1xx.xx

Source IP Port    (Media): 17298

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0



000467: Oct  6 16:57:34.548: //1566/5DFA75F28122/SIP/Call/sipSPICallInfo:

*Disconnect Cause (CC)    : 47*

*Disconnect Cause (SIP)   : 407***



47         Resource unavailable



407       Proxy authentication required     eq 21    Call rejecte

<CCIEV TOPOLOGY-X.jpg>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20121007/a7998032/attachment.html>


More information about the cisco-voip mailing list