CUBE DID Numbers

[Ermir Morina]

Greetings!

I have an issue with calls coming from the ITSP, they gave us two DID Numbers and the format that they send that number towards us is +38xxxxxxxxx.

So, I'm having problems with this because this format isn't supported by CUBE on it's translation patterns and I can't redirect this number towards my operator.

Received:

INVITE sip:+38xxxxxxxxx@

And in order for this to work I need te sip invite to be delivered as INVITE sip:38xxxxxxxxx@

Do you guys have any ideas or suggestions/workaround to fix this issue?

Looking forward to hearing from you.

Kind regards,

Ermir.

[Roger Kallberg]

What do you mean by it's not supported by translation in Cube? You can for sure translate +38xxxxxxxxx to 38xxxxxxxxx with a translation rule. See below for an example.

voice translation-rule 10
 rule 1 /^\+\(38.*\)/ /\1/

If this does not do it for your issue you could look at using a SIP profile to rewrite the SIP headers. For details on this please have a look at my all time most linked to document. In Depth Explanation of Cisco IOS and IOS-XE Call Routing - Cisco

[Ermir Morina]

Hi Roger, thank you a lot for your reply!

The problem is that I want to translate this number +38xxxxxxxxx (as this is the format the SIP Provider sends us) directly to the DN of my operator, would that be possible?

[Roger Kallberg]

Can you please explain what you mean by this "directly to the DN of my operator"?

[Ermir Morina]

@Roger Kallberg It is as @Nithin Eluvathingal explained, I need that call to be directed toward my deskphone's Extension.

And to be able to do that, I will need to have a translation profile which translates +38xxxxxxxx to for example  77600 (IP Phone DN).

[Roger Kallberg · Accepted solution]

In this case create this and assign it to the inbound direction of the connection from PSTN.

voice translation-rule 10
 rule 1 /^\+38.*/ /77600/
!
voice translation-profile PSTN-IN
 translate called 10
!
dial-peer voice <dial peer number> voip
 translation-profile incoming PSTN-IN

This assumes that you already have the needed configuration in place for call routing to the operator phone number. If you do not have that you'd need to take care of that also.

An advice in all friendliness, this is not really what you asked about in your original post.

---

@Ermir Morina wrote:

INVITE sip:+38xxxxxxxxx@

And in order for this to work I need te sip invite to be delivered as INVITE sip:38xxxxxxxxx@

---

To an outsider of your line of thought this would likely be read as that you want to translate +38xxxxxxxxx to 38xxxxxxxxx and this is what I answered in my first post.

Please try to give enough information upfront for us to best be able to help you out, without wasting yours and our time on irrelevant things.

[Ermir Morina]

Greetings @Roger Kallberg !
Thank you very much for the reply, that translation rule did work for me, I am so sorry for the late reply and also for not providing the right information upfront, I will try my best in the future thanks a lot for the advice too.

@Nithin Eluvathingal  At this moment everything seems to be working just fine, I am still working on some issues with the ITSP, thank you a lot for your prompt replies!

[Nithin Eluvathingal]

Would you mind sharing  your CUBE configuration  ?

[Nithin Eluvathingal]

I assume your requirement as when some one call +38xxxxxxxxx, you need to translate to your internal extension number  and ring the Deskphone.  is that correct ?

Could you share your Gateway Configuration  ?

If its all about stripping "+", @Roger Kallberg  provided you a solution. Did you tried that ?