[asterisk-users] DID number
michel freiha
michofr at gmail.com
Thu Sep 4 06:28:02 CDT 2008
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:961555555 at voxbone.com <sip%3A961555555 at voxbone.com>
>;tag=48201
To: <sip:DID_Number at 87.236.144.14 <sip%3ADID_Number at 87.236.144.14>
>;tag=as3fc2e680
Call-ID: 8b40a2a5a4f6082b5d33a47d01bd111d at 83.202.82.39
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
'8b40a2a5a4f6082b5d33a47d01bd111d at 83.202.82.39' in 32000 ms (Method: INVITE)
localhost*CLI>
<--- SIP read from 83.202.82.39:5060 --->
ACK sip:DID_Number at 87.236.144.14 <sip%3ADID_Number at 87.236.144.14> SIP/2.0
Call-ID: 8b40a2a5a4f6082b5d33a47d01bd111d at 83.202.82.39
CSeq: 102 ACK
From: "961555555" <sip:961555555 at voxbone.com <sip%3A961555555 at voxbone.com>
>;tag=48201
To: <sip:DID_Number at 87.236.144.14 <sip%3ADID_Number at 87.236.144.14>
>;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 <vicky.r at gmail.com> 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 <emistz at gmail.com> 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 <emistz at gmail.com
> >> <mailto:emistz at gmail.com>> 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
>
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