[Freeswitch-users] Freeswitch as Registrar for SIP Trunk

Oz Mortimer omortimer at gmail.com
Tue Feb 2 22:23:16 MSK 2016


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 at 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 at 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 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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>>> 
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>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at 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
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at 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
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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