Asterisk PBX -- Re: DID number

On Wed, Sep 3, 2008 at 11:56 PM, michel freiha <michofr@xxxxxxxxx> 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@xxxxxxxxx> 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

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