On Wed, Sep 3, 2008 at 11:56 PM, michel freiha <[EMAIL PROTECTED]> wrote:
> Hello Air,Hi, > I created an extension like ths: [442033553] user=442033553 type=pusers secret=1234 host=dynamic context=users nat=yes when calling the DID number from an extension registered on asterisk server everything looks fine...When dilaing the number fromPSTN number I'm still getting the the below erroe: [Sep 3 20:56:00] NOTICE[18440]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '442033553' rejected because extension not found. Do you think i should define a context to receive calls from outside the asterisk server?If yes do you have any context sample definition? Regards > > 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]>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 -- >> > >> > 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 >> >> 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 >> >> _______________________________________________ >> -- 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