[asterisk-users] Call to Asterisk registered sofphone from an independent unregistered Endpoint

Amar Akshat amar.akshat at gmail.com
Sun Nov 13 23:10:12 CST 2011


Sammy, Thanks much.

allowguest in sip.conf, is by default enabled, and I was not aware of
this option. So you are correct, this is on.

Yes, adding the rule in extensions works for me withslight
modification to the context by changing the default behaviour.

Thanks for the suggestion.

On Mon, Nov 14, 2011 at 1:41 PM, Sammy Govind <govoiper at gmail.com> wrote:
> 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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>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

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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