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Today's Topics:
1. Ciso IOS Gateway and SKype SIP configuration. (Charles)
2. Diff bet FCoE and DCB (Shaihan Jaffrey)
3. Re: Ciso IOS Gateway and SKype SIP configuration. (Charles)
4. Re: Ciso IOS Gateway and SKype SIP configuration. (Gavin Henry)
5. Cheap Platform for High Density FXS? (chris)
6. Re: Cheap Platform for High Density FXS? (Lelio Fulgenzi)
7. Re: Cheap Platform for High Density FXS? (Doug McIntyre)
----------------------------------------------------------------------
Message: 1
Date: Sat, 6 Oct 2012 22:18:52 +0100
From: "Charles" <[email protected]>
To: <[email protected]>
Subject: [cisco-voip] Ciso IOS Gateway and SKype SIP configuration.
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
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
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------------------------------
Message: 2
Date: Sun, 7 Oct 2012 04:24:23 +0500
From: Shaihan Jaffrey <[email protected]>
To: Cisco VOIP <[email protected]>
Subject: [cisco-voip] Diff bet FCoE and DCB
Message-ID:
<CAPXwYGTR==+XAvxmeNq5Zd931iDdrOZ_dLpjSnJVgFrM=D2L=a...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
Whats the difference between FCoE and DCB?
Regards.
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------------------------------
Message: 3
Date: Sun, 7 Oct 2012 12:02:33 +0100
From: "Charles" <[email protected]>
To: <[email protected]>
Cc: [email protected]
Subject: Re: [cisco-voip] Ciso IOS Gateway and SKype SIP
configuration.
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
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
_____
From: [email protected]
[mailto:[email protected]] On Behalf Of Charles
Sent: 06 October 2012 22:19
To: [email protected]
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
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------------------------------
Message: 4
Date: Sun, 7 Oct 2012 13:22:49 +0100
From: Gavin Henry <[email protected]>
To: Charles <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Ciso IOS Gateway and SKype SIP
configuration.
Message-ID: <3561070627146935093@unknownmsgid>
Content-Type: text/plain; charset="windows-1252"
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 [email protected]
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 <[email protected]> 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:* [email protected] [
mailto:[email protected]<[email protected]>]
*On Behalf Of *Charles
*Sent:* 06 October 2012 22:19
*To:* [email protected]
*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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------------------------------
Message: 5
Date: Sun, 7 Oct 2012 09:55:27 -0400
From: chris <[email protected]>
To: [email protected]
Subject: [cisco-voip] Cheap Platform for High Density FXS?
Message-ID:
<CAKnNFz826W9MhSMy81et0xm8ZO3jELjNeV+KYGBV7=pm7ik...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
I am looking for any advice on a cheap platform to connect high
amounts of FXS. The solution thats worked for me thus far has been
2600xm with nm-hda and the em-8fxs expansion gets me up to 12 fxs. It
seems to handle well and 2600xm is cheap in the graymarket plus small
form factor is nice for mounting in termination areas. I also like
that the NM-HDA uses the amphenol connector so its easy to connect to
a 66 block and let the wiring tech just punch down there.
I am just wondering if there is any way to get any more FXS out of
that platform? The part that sort of confuses me is that the IAD comes
in a 16 and 24fxs model and it seems like the IAD is basically a fixed
configuration of the 2600. I read in the datasheet for the NM-HDA that
you cant install more than one 8 port expansion module into the NM-hda
which is a shame it would have been nice to get 20 fxs that way
I would be open to other platforms if the price is fairly similar, I
looked at 2811 since it has more slots but the base platform is a bit
more costly and the associated NM and VIC2 cards are higher in price
as well
thanks
chris
------------------------------
Message: 6
Date: Sun, 7 Oct 2012 10:16:00 -0400 (EDT)
From: Lelio Fulgenzi <[email protected]>
To: chris <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Cheap Platform for High Density FXS?
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii
I did a price per port comparison a few years back and with the excellent price
on the VG224 4-pack, you couldn't beat the price per port.
However, I have not updated the chart with the new VG350 w/ SM FXS modules.
Sent from my iPhone...
"There's no place like 127.0.0.1"
On Oct 7, 2012, at 9:55 AM, chris <[email protected]> wrote:
> I am looking for any advice on a cheap platform to connect high
> amounts of FXS. The solution thats worked for me thus far has been
> 2600xm with nm-hda and the em-8fxs expansion gets me up to 12 fxs. It
> seems to handle well and 2600xm is cheap in the graymarket plus small
> form factor is nice for mounting in termination areas. I also like
> that the NM-HDA uses the amphenol connector so its easy to connect to
> a 66 block and let the wiring tech just punch down there.
>
> I am just wondering if there is any way to get any more FXS out of
> that platform? The part that sort of confuses me is that the IAD comes
> in a 16 and 24fxs model and it seems like the IAD is basically a fixed
> configuration of the 2600. I read in the datasheet for the NM-HDA that
> you cant install more than one 8 port expansion module into the NM-hda
> which is a shame it would have been nice to get 20 fxs that way
>
> I would be open to other platforms if the price is fairly similar, I
> looked at 2811 since it has more slots but the base platform is a bit
> more costly and the associated NM and VIC2 cards are higher in price
> as well
>
> thanks
> chris
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
Message: 7
Date: Sun, 7 Oct 2012 10:04:05 -0500
From: Doug McIntyre <[email protected]>
To: [email protected]
Subject: Re: [cisco-voip] Cheap Platform for High Density FXS?
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii
On Sun, Oct 07, 2012 at 09:55:27AM -0400, chris wrote:
> I am looking for any advice on a cheap platform to connect high
> amounts of FXS. The solution thats worked for me thus far has been
> 2600xm with nm-hda and the em-8fxs expansion gets me up to 12 fxs..
Do you need single box? Or how about two boxen?
FXS channel bank into T1 into a T1 voice-card.
Well, at least channel banks used to be fairly cheap used. I don't see
too many on eBay right now.
------------------------------
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End of cisco-voip Digest, Vol 108, Issue 7
******************************************
