[Freeswitch-users] Freeswitch as Registrar for SIP Trunk

Marc S therebel22 at gmail.com
Wed Feb 3 11:25:18 MSK 2016


I will try soon, thanks

2016-02-03 7:42 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:

> 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
>>>>>
>>>>>
>>>>>
>>>>
>>>>
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>>>
>>>
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>>>
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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
>>
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>> 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
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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