[cisco-voip] Need help with CUBE config

[Sandy Lee]

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<mailto:1501 at 112.112.112.112> which is my CUBE
Shouldn't it be To: 1501 at sip.domain.com<mailto: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<http://sip.domain.com>? Otherwise, somewhere under the voice service voip hierarchy you can define what sip.domain.com<http://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<mailto: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<http://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<mailto: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<mailto:contracts at insynctraining.com>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20110125/3f6c7020/attachment.html>