I had a similiar situation , I had my local * server registering to
another, and if I left off the part where he has /sip then all calls
came in destined for the s extension. I solved this by simply adding my
number at the end of the registration request, then ${EXTEN} actually
contained the dialed number I wanted to use.-e It's seems fuzzy now but I think on Tue, Mar 15, 2005 at 09:40:02AM +0200 , Dmitry Mishchenko said: > On Tuesday 15 March 2005 08:38, Harald Milz wrote: > > Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > > > That pattern does not match your incoming number. A pattern that would > > > match would be _ZXXX., so that the pattern will match a number of any > > > length that starts with zero and has at least three digits following the > > > zero. > > > > Kevin, I sent you a personal e-mail asking for more information. On my > > system, and with the configs attached, there is no match whatsoever except > > against "s", which does not contain the dialed number. This is with > > asterisk-1.0.6 as well as with the said CVS-HEAD. > > > > To the CVS maintainer: I have not gotten any answer so far telling me how > > exactly the patch I sent recently can be avoided in such a situation. Maybe > > sipgate.de is somewhat special. Nevertheless this patch works fine and can > > help other sipgate users too. > > I saw the same way of passing number not only with sipgate but with other SIP > providers which use SER. > > Dmitry > _______________________________________________ > Asterisk-Dev mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev _______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
