[cisco-voip] CLID issue with sip

[Dane Newman]

Im sorry about that yes I am using CUBE with the configuration below

I have also attached my capture.


voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol pass-through g711ulaw
 h323
  emptycapability
  h225 id-passthru
  h245 passthru tcsnonstd-passthru
 sip
!
!
!
voice class codec 1
 codec preference 2 g711ulaw
!
!
!
!
voice class h323 50
  h225 timeout tcp establish 3
!
!
!
!
!
!
!
!
!
!
!
voice translation-rule 1
 rule 1 /.*/ /190/
!
voice translation-rule 2
 rule 1 /.*/ /1&/
!
!
voice translation-profile aa
 translate called 1
!
voice translation-profile addone
 translate called 2
!

!
!
dial-peer voice 100 voip
 description AA Publisher
 preference 1
 destination-pattern 1..
 voice-class h323 50
 session target ipv4:10.1.80.10
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 1000 voip
 description incoming Call
 translation-profile incoming aa
 preference 1
 rtp payload-type nse 101
 rtp payload-type nte 100
 incoming called-number 6782282221
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 101 voip
 description AA Subscriber
 preference 2
 destination-pattern 1..
 voice-class h323 50
 session target ipv4:10.1.80.11
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 2000 voip
 description outbound
 translation-profile outgoing addone
 preference 1
 destination-pattern .T
 progress_ind setup enable 3
 progress_ind progress enable 8
 rtp payload-type nse 101
 rtp payload-type nte 100
 voice-class codec 1
 session protocol sipv2
 session target dns:sipconnect.ipcomms.net
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 150 voip
 description incoming outbound
 preference 1
 voice-class h323 50
 incoming called-number .T
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
!
sip-ua
 credentials username ********* password 7 *********  realm
sipconnect.ipcomms.net
 authentication username********* password 7 1*********
 authentication username *********  password 7 *********  realm
sipconnect.ipcomms.net
 registrar dns:sipconnect.ipcomms.net expires 60
 sip-server dns:sipconnect.ipcomms.net:5060
!
!
On Fri, Dec 4, 2009 at 11:24 AM, Ryan Ratliff <rratliff at cisco.com> wrote:

> The gateway screenshot appears to be an H.323 gateway, not a SIP trunk.
>  Additionally, you have only the inbound call routing info included, not
> outbound.
>
> If you are using CUBE to go from H.323 to SIP that's rather important and
> something you will want to mention.
>
> In order to use the external phone number mask you need to tick that check
> box on the route pattern.
>
>  -Ryan
>
>   On Dec 4, 2009, at 11:04 AM, Dane Newman wrote:
>
> Hello,
>
> I am having an odd issue and I am wondering why it is happening.  I have a
> sip trunk to the provider.  I call out a phone and it passes the CLID of my
> internal extention rather then the clid im setting on my gateway under
> prefix dn and External Phone Number Mask.
>
> I took a packet capture and in the sip invite message its showing SIP >From
> 101 at sipconnect.ipcomms.net which 101 is my internal extention.
>
> Is anyone able to comment?
>
> Dane
> <dn.jpg><ext.jpg>_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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