[asterisk-users] DID number

[Jaswinder Singh]

[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
>>     >
>>     >
>>     >
>>     ------------------------------------------------------------------------
>>     >
>>     > _______________________________________________
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>>
>>     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/> --
>>
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>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> -- 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
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>>    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
>