Have you looked at the syntax of register => keyword ? register => [transport://]user[:secret[:authuse...@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 <davies...@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...@x.x.x.x> > > This then causes ALL calls from the ITSP inbound to us to be addressed: > > INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0 > To: <sip:44123456...@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:44123456...@x.x.x.x:5060;transport=udp SIP/2.0 > To: <sip:44123456...@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 > _______________________________________________ -- 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