You seem like you're trying to fly before you can crawl with FreeSWITCH.  You have some base level technological understanding to acquire prior to getting to the end results. You probably didn't read what I wrote in my previous email.   Your device is registering as sip:s@5.6.7.8, so we're gonna follow the rules and call you back at sip:s@5.6.7.8

You can however in FreeSWITCH force the auth user to match the user and there by having exactly one DID per sip account, or you could change the RURI (which an example of how to do this is in the vanilla config we ship)

Here it is:

    <!--                                                                                                                                                        

        This is an example of how to override the RURI on an outgoing invite to a registered contact.                                                           

    -->

                                                                                                                                                   

    <extension name="ruri">                                                                                                                                     

      <condition field="destination_number" expression="^ruri$">                                                                                                

        <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>                      

      </condition>                                                                                                                                              

    </extension>                                                                                                                                                

                                                                                                                                                                

    <extension name="7004">                                                                                                                                     

      <condition field="destination_number" expression="^7004$">                                                                                                

        <action application="set" data="ruri_profile=default"/>                                                                                                 

        <action application="set" data="ruri_user=2000"/>                                                                                                       

        <action application="set" data="ruri_contact=1001@${domain_name}"/>                                                                                     

        <action application="execute_extension" data="ruri"/>                                                                                                   

      </condition>                                                                                                                                              

    </extension> 




On Tue, Feb 2, 2016 at 2:42 PM, Marc S <therebel22@gmail.com> wrote:
Thanks for all replies.

FS would be used as SBC.

I would like my customers' IPBX (Asterisk, Alcatel, ..)  register against FS, as a SIP trunk and then i would like routing several DID (not only s) on incoming call to FS  to customers IPBX with bridge action.

I prefer IPBX login/pass auth against FS rather than IPBX IP auth only, because of security.

But registring user in FS seems to expect SIP phone only as user to register, not IPBX as user :

Here my asterisk Register :

REGISTER sip:1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport
From: <sip:testipbx2@1.2.3.4>;tag=as47a2d14b
To: <sip:testipbx2@1.2.3.4>
Call-ID: 6896d4fe6ab4904b7830df881a25e4cf@127.0.1.1
CSeq: 103 REGISTER
User-Agent: Asterisk
Max-Forwards: 70
Authorization: Digest username="testipbx2", realm="1.2.3.4", algorithm=MD5, uri="sip:1.2.3.4", nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1", response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8", nc=00000001
Expires: 120
Contact: <sip:s@5.6.7.8>
Event: registration
Content-Length: 0

Here FS->Asterisk invite on incoming call

INVITE sip:s@5.6.7.8 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c
Max-Forwards: 69
From: "45678776" <sip:45678776@1.2.3.4>;tag=4v1S65HQ930mF
To: <sip:s@5.6.7.8>
Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1
CSeq: 86886190 INVITE
Contact: <sip:mod_sofia@1.2.3.4:5060>
User-Agent: FS/1.6
..
Thanks



 

2016-02-02 21:29 GMT+01:00 Oz Mortimer <omortimer@gmail.com>:
I think you need to explain what you are trying to achieve.. 
What you are doing currently is registering a user against freeswitch - not dissimilar to logging in to Skype. 
In your bridge statement you are calling the registered user - again like you would in Skype.
Is there a reason you want asterisk to register against freeswitch? The only thing I can think of is that it's on a private LAN and that your asterisk box does something "special"..


On 2 Feb 2016, at 19:28, Marc S <therebel22@gmail.com> wrote:

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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_________________________________________________________________________
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_________________________________________________________________________
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http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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FreeSWITCH-users mailing list
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_________________________________________________________________________
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consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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FreeSWITCH-users mailing list
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