[asterisk-users] DID number

[michel freiha]

On Wed, Sep 3, 2008 at 11:56 PM, michel freiha <michofr at gmail.com> 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 <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
>>
>> _______________________________________________
>> -- 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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