[cisco-voip] Need help with CUBE config

Mac GroupStudy mac.groupstudy at gmail.com
Tue Jan 25 16:58:32 EST 2011


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<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 <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<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 <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.
>
>
>
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>
>


-- 
Charles S. Dye
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