<div dir="ltr">You may need to change the host name from the CLI. I had to use the set network hostname after discovering I could not ping the new host name from the CLI. UCM <a href="http://6.1.2.">6.1.2.</a><br><br>Mike<br>
<br><div class="gmail_quote">On Thu, Oct 2, 2008 at 4:28 PM, <span dir="ltr"><<a href="mailto:cisco-voip-request@puck.nether.net">cisco-voip-request@puck.nether.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Send cisco-voip mailing list submissions to<br>
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or, via email, send a message with subject or body 'help' to<br>
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When replying, please edit your Subject line so it is more specific<br>
than "Re: Contents of cisco-voip digest..."<br>
<br>
<br>
Today's Topics:<br>
<br>
1. Re: Outbound Dial with Carrier SIP Trunk (Wilusz, Mike)<br>
2. Re: Outbound Dial with Carrier SIP Trunk (Ryan West)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Thu, 2 Oct 2008 16:20:05 -0400<br>
From: "Wilusz, Mike" <<a href="mailto:mikewilusz@pricechopper.com">mikewilusz@pricechopper.com</a>><br>
Subject: Re: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
To: "Ryan West" <<a href="mailto:rwest@zyedge.com">rwest@zyedge.com</a>>, "Matt Slaga \(US\)"<br>
<<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>>, <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
Message-ID:<br>
<<a href="mailto:CF128CB1374E15468AF9D1A2219BE71354D755@EXCHANGE.golub.com">CF128CB1374E15468AF9D1A2219BE71354D755@EXCHANGE.golub.com</a>><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
Ryan,<br>
<br>
<br>
<br>
There's no required authentication for the provider (username/password).<br>
The show command just shows empty.<br>
<br>
<br>
<br>
Thanks everyone for the input so far in case I forgot to say that!<br>
<br>
<br>
<br>
Mike Wilusz, CCNA<br>
<br>
Networking Systems Programmer<br>
<br>
Price Chopper Supermarkets / The Golub Corporation<br>
<br>
From: Ryan West [mailto:<a href="mailto:rwest@zyedge.com">rwest@zyedge.com</a>]<br>
Sent: Thursday, October 02, 2008 4:17 PM<br>
To: Wilusz, Mike; Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
<br>
<br>
Are you required to register your sip-ua? If so, please also post 'show<br>
sip-ua reg stat'<br>
<br>
<br>
<br>
-ryan<br>
<br>
<br>
<br>
From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><br>
[mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Wilusz, Mike<br>
Sent: Thursday, October 02, 2008 4:11 PM<br>
To: Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: Re: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
<br>
<br>
Matt,<br>
<br>
<br>
<br>
Tried that and I still get the same result. The voice config from the<br>
CUBE is this:<br>
<br>
<br>
<br>
voice service voip<br>
<br>
address-hiding<br>
<br>
allow-connections h323 to h323<br>
<br>
allow-connections h323 to sip<br>
<br>
allow-connections sip to h323<br>
<br>
allow-connections sip to sip<br>
<br>
!<br>
<br>
dial-peer voice 2 voip<br>
<br>
destination-pattern 1..........<br>
<br>
session protocol sipv2<br>
<br>
session target ipv4:<a href="http://64.206.114.6" target="_blank">64.206.114.6</a> (Carrier Gateway)<br>
<br>
dtmf-relay rtp-nte<br>
<br>
codec g711ulaw<br>
<br>
!<br>
<br>
dial-peer voice 3 voip<br>
<br>
destination-pattern 3474179896<br>
<br>
session protocol sipv2<br>
<br>
session target ipv4:<a href="http://10.50.120.33" target="_blank">10.50.120.33</a> (Our CUCM)<br>
<br>
dtmf-relay rtp-nte<br>
<br>
codec g711ulaw<br>
<br>
!<br>
<br>
!<br>
<br>
sip-ua<br>
<br>
registrar ipv4:<a href="http://64.206.114.6" target="_blank">64.206.114.6</a> expires 3600<br>
<br>
<br>
<br>
<br>
<br>
Mike Wilusz, CCNA<br>
<br>
Networking Systems Programmer<br>
<br>
Price Chopper Supermarkets / The Golub Corporation<br>
<br>
<br>
<br>
From: Matt Slaga (US) [mailto:<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>]<br>
Sent: Thursday, October 02, 2008 4:06 PM<br>
To: Wilusz, Mike; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
<br>
<br>
Try adding this to your config:<br>
<br>
<br>
<br>
voice class codec 1<br>
<br>
codec preference 1 g711ulaw<br>
<br>
codec preference 2 g729br8<br>
<br>
codec preference 3 g729r8<br>
<br>
<br>
<br>
<br>
<br>
And to your voip dial peers:<br>
<br>
<br>
<br>
voice-class codec 1<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
From: Wilusz, Mike [mailto:<a href="mailto:mikewilusz@pricechopper.com">mikewilusz@pricechopper.com</a>]<br>
Sent: Thursday, October 02, 2008 3:42 PM<br>
To: Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
<br>
<br>
Yup... set here in this dial-peer. Which is matched properly according<br>
to the debug.<br>
<br>
<br>
<br>
dial-peer voice 2 voip<br>
<br>
destination-pattern 1..........<br>
<br>
session protocol sipv2<br>
<br>
session target ipv4:<a href="http://64.206.114.6" target="_blank">64.206.114.6</a><br>
<br>
dtmf-relay rtp-nte<br>
<br>
codec g711ulaw<br>
<br>
<br>
<br>
<br>
<br>
Mike Wilusz, CCNA<br>
<br>
Networking Systems Programmer<br>
<br>
Price Chopper Supermarkets / The Golub Corporation<br>
<br>
<br>
<br>
From: Matt Slaga (US) [mailto:<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>]<br>
Sent: Thursday, October 02, 2008 3:40 PM<br>
To: Wilusz, Mike; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
<br>
<br>
Where are you setting the codec? The line below says you are not using<br>
VCC, are your dial peers specifying a codec?<br>
<br>
<br>
<br>
Oct 2 10:21:18: //10/36ECCAF0801A/SIP/Info/sipSPIGetCallConfig: Not<br>
using Voice Class Codec<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
From: Wilusz, Mike [mailto:<a href="mailto:mikewilusz@pricechopper.com">mikewilusz@pricechopper.com</a>]<br>
Sent: Thursday, October 02, 2008 3:35 PM<br>
To: Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
<br>
<br>
Matt,<br>
<br>
<br>
<br>
Not sure what means. I see g726r32 is selected as a codec...but why? I<br>
don't have the set as my codec on the dial-peer and the carrier is set<br>
(or supposed to be) to the same as me. I guess I'm not really sure what<br>
that line is telling me.<br>
<br>
<br>
<br>
Mike Wilusz, CCNA<br>
<br>
Networking Systems Programmer<br>
<br>
Price Chopper Supermarkets / The Golub Corporation<br>
<br>
<br>
<br>
From: Matt Slaga (US) [mailto:<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>]<br>
Sent: Thursday, October 02, 2008 3:14 PM<br>
To: Wilusz, Mike; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
<br>
<br>
Probably this:<br>
<br>
<br>
<br>
<br>
<br>
Oct 2 10:21:18: //9/36ECCAF0801A/SIP/Info/sipSPISelectDynamicPayload:<br>
Selecting Dynamic Payload : 96 for Codec: g726r32<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><br>
[mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Wilusz, Mike<br>
Sent: Thursday, October 02, 2008 10:31 AM<br>
To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
<br>
<br>
Hey All,<br>
<br>
<br>
<br>
We're in the process of trialing a SIP trunk solution from our provider,<br>
PAETEC. I followed the Cisco docs for configuring a CUBE, but we can<br>
only receive calls and not place them. The setup is CUCM 6.1(2) -> SIP<br>
Trunk -> CUBE (2811) -> SIP Trunk -> PAETEC SIP Gateway. I'm very new<br>
to SIP and am trying to analyze the output from debug ccsip all.<br>
Nothing is jumping out at me as the exact reason why the calls outbound<br>
from our CUCM aren't working. Anybody have any ideas?<br>
<br>
<br>
<br>
<br>
<br>
________________________________<br>
<br>
Disclaimer: This e-mail communication and any attachments may contain<br>
confidential and privileged information and is for use by the designated<br>
addressee(s) named above only. If you are not the intended addressee,<br>
you are hereby notified that you have received this communication in<br>
error and that any use or reproduction of this email or its contents is<br>
strictly prohibited and may be unlawful. If you have received this<br>
communication in error, please notify us immediately by replying to this<br>
message and deleting it from your computer. Thank you.<br>
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<br>
------------------------------<br>
<br>
Message: 2<br>
Date: Thu, 2 Oct 2008 16:28:13 -0400<br>
From: Ryan West <<a href="mailto:rwest@zyedge.com">rwest@zyedge.com</a>><br>
Subject: Re: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
To: "Wilusz, Mike" <<a href="mailto:mikewilusz@pricechopper.com">mikewilusz@pricechopper.com</a>>, "Matt Slaga (US)"<br>
<<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>>, "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>"<br>
<<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
Message-ID:<br>
<6E21B2BDEF6E714EA0B5BA8D5D0E140163BF5B96@zy-ex1.zyedge.local><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
Try adding this to your config:<br>
Dial-peer voice 2 voip<br>
progress_ind setup enable 3<br>
progress_ind connect enable 8<br>
<br>
<br>
-ryan<br>
From: Wilusz, Mike [mailto:<a href="mailto:mikewilusz@pricechopper.com">mikewilusz@pricechopper.com</a>]<br>
Sent: Thursday, October 02, 2008 4:20 PM<br>
To: Ryan West; Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Ryan,<br>
<br>
There's no required authentication for the provider (username/password). The show command just shows empty.<br>
<br>
Thanks everyone for the input so far in case I forgot to say that!<br>
<br>
Mike Wilusz, CCNA<br>
Networking Systems Programmer<br>
Price Chopper Supermarkets / The Golub Corporation<br>
From: Ryan West [mailto:<a href="mailto:rwest@zyedge.com">rwest@zyedge.com</a>]<br>
Sent: Thursday, October 02, 2008 4:17 PM<br>
To: Wilusz, Mike; Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Are you required to register your sip-ua? If so, please also post 'show sip-ua reg stat'<br>
<br>
-ryan<br>
<br>
From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Wilusz, Mike<br>
Sent: Thursday, October 02, 2008 4:11 PM<br>
To: Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: Re: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Matt,<br>
<br>
Tried that and I still get the same result. The voice config from the CUBE is this:<br>
<br>
voice service voip<br>
address-hiding<br>
allow-connections h323 to h323<br>
allow-connections h323 to sip<br>
allow-connections sip to h323<br>
allow-connections sip to sip<br>
!<br>
dial-peer voice 2 voip<br>
destination-pattern 1..........<br>
session protocol sipv2<br>
session target ipv4:<a href="http://64.206.114.6" target="_blank">64.206.114.6</a> (Carrier Gateway)<br>
dtmf-relay rtp-nte<br>
codec g711ulaw<br>
!<br>
dial-peer voice 3 voip<br>
destination-pattern 3474179896<br>
session protocol sipv2<br>
session target ipv4:<a href="http://10.50.120.33" target="_blank">10.50.120.33</a> (Our CUCM)<br>
dtmf-relay rtp-nte<br>
codec g711ulaw<br>
!<br>
!<br>
sip-ua<br>
registrar ipv4:<a href="http://64.206.114.6" target="_blank">64.206.114.6</a> expires 3600<br>
<br>
<br>
Mike Wilusz, CCNA<br>
Networking Systems Programmer<br>
Price Chopper Supermarkets / The Golub Corporation<br>
<br>
From: Matt Slaga (US) [mailto:<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>]<br>
Sent: Thursday, October 02, 2008 4:06 PM<br>
To: Wilusz, Mike; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Try adding this to your config:<br>
<br>
voice class codec 1<br>
codec preference 1 g711ulaw<br>
codec preference 2 g729br8<br>
codec preference 3 g729r8<br>
<br>
<br>
And to your voip dial peers:<br>
<br>
voice-class codec 1<br>
<br>
<br>
<br>
<br>
From: Wilusz, Mike [mailto:<a href="mailto:mikewilusz@pricechopper.com">mikewilusz@pricechopper.com</a>]<br>
Sent: Thursday, October 02, 2008 3:42 PM<br>
To: Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Yup... set here in this dial-peer. Which is matched properly according to the debug.<br>
<br>
dial-peer voice 2 voip<br>
destination-pattern 1..........<br>
session protocol sipv2<br>
session target ipv4:<a href="http://64.206.114.6" target="_blank">64.206.114.6</a><br>
dtmf-relay rtp-nte<br>
codec g711ulaw<br>
<br>
<br>
Mike Wilusz, CCNA<br>
Networking Systems Programmer<br>
Price Chopper Supermarkets / The Golub Corporation<br>
<br>
From: Matt Slaga (US) [mailto:<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>]<br>
Sent: Thursday, October 02, 2008 3:40 PM<br>
To: Wilusz, Mike; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Where are you setting the codec? The line below says you are not using VCC, are your dial peers specifying a codec?<br>
<br>
Oct 2 10:21:18: //10/36ECCAF0801A/SIP/Info/sipSPIGetCallConfig: Not using Voice Class Codec<br>
<br>
<br>
<br>
<br>
From: Wilusz, Mike [mailto:<a href="mailto:mikewilusz@pricechopper.com">mikewilusz@pricechopper.com</a>]<br>
Sent: Thursday, October 02, 2008 3:35 PM<br>
To: Matt Slaga (US); <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Matt,<br>
<br>
Not sure what means. I see g726r32 is selected as a codec...but why? I don't have the set as my codec on the dial-peer and the carrier is set (or supposed to be) to the same as me. I guess I'm not really sure what that line is telling me.<br>
<br>
Mike Wilusz, CCNA<br>
Networking Systems Programmer<br>
Price Chopper Supermarkets / The Golub Corporation<br>
<br>
From: Matt Slaga (US) [mailto:<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>]<br>
Sent: Thursday, October 02, 2008 3:14 PM<br>
To: Wilusz, Mike; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Probably this:<br>
<br>
<br>
Oct 2 10:21:18: //9/36ECCAF0801A/SIP/Info/sipSPISelectDynamicPayload: Selecting Dynamic Payload : 96 for Codec: g726r32<br>
<br>
<br>
<br>
From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Wilusz, Mike<br>
Sent: Thursday, October 02, 2008 10:31 AM<br>
To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: [cisco-voip] Outbound Dial with Carrier SIP Trunk<br>
<br>
Hey All,<br>
<br>
We're in the process of trialing a SIP trunk solution from our provider, PAETEC. I followed the Cisco docs for configuring a CUBE, but we can only receive calls and not place them. The setup is CUCM 6.1(2) -> SIP Trunk -> CUBE (2811) -> SIP Trunk -> PAETEC SIP Gateway. I'm very new to SIP and am trying to analyze the output from debug ccsip all. Nothing is jumping out at me as the exact reason why the calls outbound from our CUCM aren't working. Anybody have any ideas?<br>
<br>
<br>
<br>
<br>
________________________________<br>
<br>
Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.<br>
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End of cisco-voip Digest, Vol 60, Issue 22<br>
******************************************<br>
</blockquote></div><br><br clear="all"><br>-- <br>Best Regards,<br><br>Mike Lydick<br><br>
</div>