Dear Sir, Please find below the error that we are getting when enabling 'sip set debug'.
localhost*CLI> <--- Reliably Transmitting (no NAT) to 83.202.82.39:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 83.202.82.39:5060 ;branch=z9hG4bK5ac79f249887f915005b5d34415b1a56;received=83.202.82.39 From: "961555555" <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >;tag=48201 To: <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >;tag=as3fc2e680 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Sep 4 11:13:05] NOTICE[10388]: chan_sip.c:14035 handle_request_invite: Call from 'sip_proxy' to extension 'DID_Number' rejected because extension not found. Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) localhost*CLI> <--- SIP read from 83.202.82.39:5060 ---> ACK sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK From: "961555555" <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >;tag=48201 To: <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >;tag=as3fc2e680 Via: SIP/2.0/UDP 83.202.82.39:5060 ;branch=z9hG4bK5ac79f249887f915005b5d34415b1a56 Max-Forwards: 69 User-Agent: Vox Callcontrol Content-Length: 0 Regards On 9/4/08, Jaswinder Singh <[EMAIL PROTECTED]> wrote: > > [442033553] > user=442033553 > type=pusers > secret=1234 > host=dynamic > context=users > nat=yes > > > make it context=stations , i am assuming this is how your DID provider > is sending u calls ? > > Let us know if your DID provider is just sending calls to your ip > address or you are registering asterisk server with the, . Keep > context=stations in extensions.conf global section . > > On Thu, Sep 4, 2008 at 2:41 AM, Igor Hernandez <[EMAIL PROTECTED]> wrote: > > 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 > > > > _______________________________________________ > -- 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
