[Freeswitch-users] Freeswitch as Registrar for SIP Trunk

Marc S therebel22 at gmail.com
Tue Feb 2 23:06:01 MSK 2016


Thanks, Asterisk is already registred to FS.

But when incoming call to FS is bridged to registered Asterisk :

<action application="bridge" data="user/myuserid"/>

FS send SIP Message to asterisk :

INVITE s@<asterisk_ip>

instead of :

INVITE 12345678@<asterisk_ip>

I would like to get 12345678 in asterisk..




2016-02-02 20:57 GMT+01:00 Alexandr Usov <blessendor at gmail.com>:

> You must have in Asterisk configs something as in my examples.
>
> ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc.
> With static IP of both FS and Asterisk you not need to do register from
> asterisk.
> register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch
>
>
> ;; type=friend means that this peer can be used for outbound and inbound
> calls, so we don't need to create two peer settings block ( [freeswitch-in]
> and [freeswitch-out] )
>
> [freeswitch] ;; you maybe want to use here your public DID number, as well
> as in username/fromuser settings
> type=friend
> username=asterisk2fs
> fromuser=asterisk2fs
> fromdomain=ip_of_freeswitch
> host=ip_of_freeswitch
> context=from-freeswitch
> secret=strongpassword
> insecure=port,invite
> qualify=yes
> port=5060
>
>
> Your context [from-freeswitch] must have an extension, named as
> 'freeswitch' (or your DID number) for incoming calls operations.
>
>
>
>
> 2016-02-02 21:28 GMT+02:00 Marc S <therebel22 at gmail.com>:
>
>> 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 at 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 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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>>>
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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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>>
>
>
> _________________________________________________________________________
> 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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