[asterisk-users] SIP Registration and INVITE question

[Martin]

Have you looked at the syntax of register => keyword ?

register => [transport://]user[:secret[:authuser]]@host[:port][/extension]
; If no extension is given, the 's' extension is used.

There you have it ... Contact: <sip:s ....

set the extension and you should be fine

Martin

On Mon, Apr 6, 2009 at 7:45 AM, Steve Davies <davies147 at gmail.com> wrote:
> I have an ITSP we are trying to work with that has an "Unusual" way of
> working, but that said my understanding of their behaviour is that it
> is fully RFC compliant. Can someone suggest how I might be able to
> interoperate under these circumstances:
>
> We register fine with them, and send the default asterisk Contact: header of:
>     Contact: <sip:s at x.x.x.x>
>
> This then causes ALL calls from the ITSP inbound to us to be addressed:
>
>     INVITE sip:s at x.x.x.x:5060;transport=udp SIP/2.0
>     To: <sip:44123456789 at x.x.x.x:5060;transport=udp>
>     [other headers omitted]
>
> In fact, whatever we send in the Contact: header is reflected in the
> INVITE for inbound calls, and the actual number dialled is always
> placed in the To: header. What 99.9% of our ITSPs would send is:
>
>     INVITE sip:44123456789 at x.x.x.x:5060;transport=udp SIP/2.0
>     To: <sip:44123456789 at x.x.x.x:5060;transport=udp>
>     [other headers omitted]
>
> As you can see, the correct destination number is placed into the
> INVITE header AND the To: header, and Asterisk routes it correctly
> based on the INVITE.
>
> My questions:
>
> - Is there a way of telling chan_sip to register with multiple
> Contact: headers in the registration request, so that all of the
> acceptable DDI numbers can be presented to the ITSP (This is what the
> RFC seems to suggest is the "correct" way to operate.)
>
> - Alternatively, has anyone encountered this previously, and perhaps
> created an "s" extension that then digs into the To: header, and
> routes according to that? Examples, workarounds and solutions are all
> welcome!
>
> Help?
>
> Thanks,
> Steve
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>