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@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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