[Freeswitch-users] Freeswitch as Registrar for SIP Trunk

[Sergey Safarov]

What is way you planing to use for link DID with user?

Sergey

On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:

> Hello,
> sorry for asking again :
>
> I hav an asterisk that register on Freeswitch (as a user).
>
> When a call is incoming to FS, FS send it to asterisk : In asterisk, it is
> the s extension.
>
> Here my bridge tests
>
> <action application="bridge" data="user/myuserid"/>
>
> => s extension in asterisk instead of extension
>
> <action application="bridge" data="sofia/mycontext/myextension@
> <ASTERISK_IP>"/>
>
> => Not authenticated in asterisk (because no IP authentication in asterisk)
>
> Have you an idea how to send real extension instead of s extension  ?
> Thanks
>
>
>
>
>
>
>
> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>
>> Hello,
>>
>> i'm discovering FS. I hav read a lot about users and gateways.
>>
>> I would like to  FS act as registrar for authenticated SIP trunking.
>>
>> - Customers IPBX would register with login/password to Freeswitch.
>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>
>> directory/users does not seem to be the solution because in dialplan,
>> destination DID can't be defined, only user id :
>>
>> <action application="bridge" data="user/myuserid"/>
>>
>> gateway seems to be designed for SIP trunking to remote SIP gateway, not
>> for FS to act as registrar.
>>
>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>
>> Thanks a lot,
>> Marc
>>
>>
>>
>
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