[cisco-voip] Need help with CUBE config

[Brad Ellis]

How are you doing your DNS resolution?  Do you have a name server
configured?

 

thanks,

Brad Ellis

CCIE#5796 (R&S / Security)

CCSI# 30482

CEO / President

CCBOOTCAMP - Cisco Learning Solutions Partner (CLSP)

Email: brad at ccbootcamp.com

Toll Free: 877-654-2243

International: +1-702-968-5100

Skype: skype:ccbootcamp?call

FAX: +1-702-446-8012

YES! We take Cisco Learning Credits!

Training And Remote Racks: http://www.ccbootcamp.com
<http://www.ccbootcamp.com/> 

Routing and Switching Forums: http://www.routerie.com
<http://www.routerie.com/> 

Security Forums: http://www.securityie.com <http://www.securityie.com/> 

Voice Forums: http://www.voiceie.com <http://www.voiceie.com/> 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sandy Lee
Sent: Tuesday, January 25, 2011 4:33 PM
To: Mac GroupStudy
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Need help with CUBE config

 

Hi,

Yes I want to call From: 4420 To: 1501, but the problem is that I see 

 

>From :4420 at 10.0.17.60 which is my UCM

To: 1501 at 112.112.112.112 which is my CUBE

Shouldn't it be To: 1501 at sip.domain.com which is the SiteB SIP proxy
server ?

 

This is new for me, so I'm very confused.

Thanks. 

 

Is DNS able to resolve sip.domain.com? Otherwise, somewhere under the
voice service voip hierarchy you can define what sip.domain.com actually
is (I see where one target is an IP and the other is an FQDN). Also, I
am not sure why you say you see what you do. I mean, I see a call
From:4420 To:1501. IS that not what it should have been?

 

On Tue, Jan 25, 2011 at 1:30 PM, Sandy Lee <Sandy.Lee at dti.ulaval.ca>
wrote:

Hi,

I have the following setup: UCM -- sip trunk - CUBE. 

I need to reach another site which has this setup : SIP Proxy -sip trunk
- UCM.

 

On my CUBE, I have several dial-peers to send the calls to the SIP
proxy. Here's my config:

 

!

version 15.1

!

hostname TEL-CUBE-PR01

!

!

no ipv6 cef

ip source-route

no ip cef

!

voice-card 0

 dspfarm

 dsp services dspfarm

!

voice service voip

 media statistics

 allow-connections sip to sip

 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback
none

 sip

  bind control source-interface GigabitEthernet0/0

  bind media source-interface GigabitEthernet0/0

  sip-profiles 100

!

voice class sip-profiles 100

 request INVITE sip-header To modify "<sip:1501 at .*>"
"<sip:1501 at domain.com <mailto:sip%3A1501 at domain.com> >" 

!

!

interface GigabitEthernet0/0

 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

 ip address 112.112.112.112 255.255.255.0

 ip route-cache same-interface

 duplex full

 speed 100

 service-policy output AutoQoS-Policy-Trust

!

dial-peer voice 1 voip

 description Inbound calls

 incoming called-number .

 codec g711ulaw

!

dial-peer voice 1500 voip

 description Outbound to SiteA

 destination-pattern 15..

 session protocol sipv2

 session target dns:sip.domain.com

 dtmf-relay h245-alphanumeric sip-notify rtp-nte

 codec g711ulaw

!

!

dial-peer voice 4000 voip

 description Inbound From SiteA 

 destination-pattern 44..

 session protocol sipv2

 session target ipv4:10.0.16.60

 dtmf-relay h245-alphanumeric sip-notify rtp-nte

 codec g711ulaw

!

gatekeeper

 shutdown

!

 

So, when I try to call DN 1501 from my extension 4420, I see it as:

 

From: "SANDY LEE" <sip:4420 at 10.0.17.60 <mailto:sip%3A4420 at 10.0.17.60>
>;tag=a1be1450-93a3-47d3-9429-252be029c8ef-34475608

Allow-Events: presence, kpml                       

P-Asserted-Identity: "SANDY LEE" <sip:4420 at 10.0.17.60
<mailto:sip%3A4420 at 10.0.17.60> >

Supported: timer,resource-priority,replaces

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Min-SE:  1800

Remote-Party-ID: "SANDY LEE" <sip:4420 at 10.0.17.60
<mailto:sip%3A4420 at 10.0.17.60> >;party=calling;screen=yes;privacy=off

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: sip:1501 at 112.112.112.112

 

It looks like I'm sending the call to myself, what am I doing wrong? Any
idea what might be my problem ? When the SiteA calls me, I have
"Disconnect Cause (SIP)   : 403". The only thing I came up with is that
the CUBE sees the call, but forbids it for a reason that I don't know.

 

Any help would be appreciated.

Thanks and regards.

 

Sandy.

 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
Charles S. Dye
Contracts Officer
InSync Training LLC
(v) (860) 553-3521
(m) (860) 303-2617
(f) (775) 522-2740
contracts at insynctraining.com

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20110127/af36c945/attachment.html>