[Freeswitch-users] Freeswitch as Registrar for SIP Trunk

[Sergey Safarov]

Read theread "Conectin a SPA3102 as pstn gateway on FS
<http://lists.freeswitch.org/pipermail/freeswitch-users/2015-August/115045.html>
"

{sip_invite_to_uri=<sip:${destination_number}@mydomain.org>}user/
reg_user at mydomain.org

If requred mutal autertication

{sip_auth_username='login_name',sip_auth_password='strong_password_here',sip_invite_to_uri=<sip:${
destination_number}@mydomain.org>}user/ reg_user at mydomain.org


On Tue, Feb 2, 2016 at 10:03 PM, 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
>>>>
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
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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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