It seems that "freeswitch gateway username" return all results about setup username and password on FS to register against external SIP gateway, am i wrong ?

2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer@gmail.com>:
Setup username / password authentication on asterisk and set the corresponding user & pass in your freeswitch gateway. 
I'm sure a google for "asterisk username authentication" and "freeswitch gateway username" will give you plenty of examples - you will want freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
Google is your friend.. 


On 2 Feb 2016, at 19:03, Marc S <therebel22@gmail.com> wrote:

I want to use mod xml curl to generate dynamic dialplan xml like this :

 <extension name="inbound_12345678">
        <condition field="destination_number" expression="^(12345678)$">
           <action application="bridge" data="user/myuserid"/>
        </condition>
</extension>

Thanks



2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov@gmail.com>:
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@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@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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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