[Freeswitch-users] Freeswitch as Registrar for SIP Trunk

[Marc S]

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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