[Asterisk-Users] SIP DID routing

[denis at isolve.com.br]

Hi Chris.

You have been facing the same problem of mine. I was encouraged to use the
CVS HEAD version that includes an application called SIP_HEADER. With
SIP_HEADER we can handle SIP Headers fields.

If you get success on it, please let me know. I will do the same.

Regards,

Deniss Galvãao.

> How do you get the called number on incoming SIP calls?  I've never
> had multiple DID's via SIP from one provider before and somehow I
> never realized that with IAX it just works, and SIP is different.
>
> If I don't set an extension in the register command the incoming
> invite has <sip:s at me.com> in the To field.  Now if I have multiple
> DID's that I want routed to different extensions, what's the solution?
>  Is there a SIP header that is normally used to pass the called number
> in?
>
> Hope that makes sense..
>
> Chris
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