[cisco-voip] Need help with CUBE config

[Nick Matthews]

I would try removing the sip profile under voice service voip.  I don't
think that's a good policy to apply globally.  What may be happening is
you're receiving the call to 1501 at 112.112.112.112 and then we're returning
1501 at domain.com and the other SIP user agent gets confused with why you're
messing with the headers.

Or, it could be that you really do need the domain conversion before the
call will complete successfully, and your pattern doesn't match correctly.

This: To: sip:1501 at 112.112.112.112
does not match:
"<sip:1501 at .*>" "<sip:1501 at domain.com <sip%3A1501 at domain.com>>"

Because of the lack of <> brackets.

It's hard to say without more details.


As well, your dtmf is misconfigured.

dial-peer voice 1 voip

 description Inbound calls

 incoming called-number .

 codec g711ulaw

!


Most people enable rtp-nte for SIP.  I would add this to dial peer 1 and
remove the h245 alpha and sip-notify from your outgoing dial peer.


As well, 'no vad' is best practice as vad varies from carrier to carrier as
well as to remove possibilities for voice quality issues.


-nick


On Tue, Jan 25, 2011 at 5:20 PM, Mark Holloway <mh at markholloway.com> wrote:

> Change the session target to IPV4 and see if the call completes.  If so
> then you have a DNS problem.  If I remember correctly you can set the dial
> peer to be a sip-server instead of DNS and under sip-ua enter the DNS name
> there.  Doing this support DNS SRV records where doing DNS in the dial-peer
> does not.  Also, "debug ccsip messages" is helpful to see what is going on.
>
> On Jan 25, 2011, at 2:58 PM, Mac GroupStudy wrote:
>
> 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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>>
>>
>
>
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