Hey, Did you reload asterisk after changing the extensions.conf?
Also, if you try it with "sip set debug" on the console what do you see? michel freiha wrote: > Hello Air, > > I did what you asked for but I got the following error: > > extensions.conf: > > [stations] > exten => 442033553,1,Answer > exten => 442033553,n,Playback(demo-nogo) > > Error message: > [Sep 3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite: > Call from '' to extension '442033553' rejected because extension not found. > Regards > On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>> wrote: > > michel freiha wrote: > > Hi All, > > I bought a DID number from VOxbone...this number could be dialed from > > any PSTN line and could be forwarded to any SIP server like asterisk > > server...Now I need to forward this number to my asterisk server > so when > > a customer dial this number from his GSM or Land line PSTN number the > > call will be forwarde to my asterisk server and I need to play a wav > > file for example.. > > Can you please give me some tips about how to accomplish this task? > > > > Regards > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > <http://www.api-digital.com/> -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net <http://www.astricon.net/> > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > Hello, > > I have never used that provider but usually either the provider knows > your switch's ip and routes the did traffic to it or you have asterisk > register with the provider so that it knows where to route the calls. > > Once thats done you can do something like > > exten => XXXXXXXXXX,1,Answer > exten => XXXXXXXXXX,n,Playback(file) > > Where the x's are the number that you see coming in from your provider. > If you're routed all your dids from what looks like one > number(callcentric does this) then you might need to use the sip header > to route your did to the particular extension you want. You shouldn't > have to bother with this if you only have one did. > > > Regards, > > -- > Igor Hernandez > Escape Communications > http://www.escapetel.com <http://www.escapetel.com/> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com > <http://www.api-digital.com/> -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net <http://www.astricon.net/> > > 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 -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
