[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>
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