asterisk users - Nov 2011 - Call to Asterisk registered sofphone from an independent unregistered Endpoint

Hi,
I have an Endpoint written in C, which simply sends out SIP invite to
another endpoint and also sets up media session after the call is
initiated. Now I am using this endpoint to call to the Asterisk PBX.
And the call is successfull.

Now, I have a softphone registered with asterisk with extension 1234,
and I want to call that softphone from my external endpoint which is
not resgistered with Asterisk. So I am sending an invite to the SIP
URI

sip:1234@<host-ip>:<port>, however, this call does not ring the
Softphone with extension, and the call is auto answered by Asterisk.
How can I configure/enable Asterisk to forward that call to the
softphone, rather than answering itself.

-- 

Thank you...

Amar Akshat

Please excuse any spelling mistakes, as this email was sent from a
"not so good" mobile device.

Hi,
The end-point which isn't registered in asterisk will hit the
"default"
context in asterisk. This is the one which you've defined in sip.conf
general section i.e

[general]
...
context=my-context

Also, if your calls are successful from any unregistered endpoint then I
think you've enable allowguest in sip.conf.

So if you need to bridge the call to 1234 extension make sure you've a
dialplan like this in extensions.conf

[my-context]
exten => 1234,1,Dial(SIP/1234)
same =>         n,Hangup()
OR
exten => _X.,1,Dial(SIP/${EXTEN}) ;<== Security Warning, don't use in
production server.

Hope this helps,

--
Regards,
Sammy

On Mon, Nov 14, 2011 at 8:25 AM, Amar Akshat <amar.akshat at gmail.com>
wrote:
> Hi,
> I have an Endpoint written in C, which simply sends out SIP invite to
> another endpoint and also sets up media session after the call is
> initiated. Now I am using this endpoint to call to the Asterisk PBX.
> And the call is successfull.
>
> Now, I have a softphone registered with asterisk with extension 1234,
> and I want to call that softphone from my external endpoint which is
> not resgistered with Asterisk. So I am sending an invite to the SIP
> URI
>
> sip:1234@<host-ip>:<port>, however, this call does not ring the
> Softphone with extension, and the call is auto answered by Asterisk.
> How can I configure/enable Asterisk to forward that call to the
> softphone, rather than answering itself.
>
> --
>
> Thank you...
>
> Amar Akshat
>
> Please excuse any spelling mistakes, as this email was sent from a
> "not so good" mobile device.
>
> --
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