[Freeswitch-users] Sending SAVPF INVITE to Opensips

[Stanislav Sinyagin]

in your ws-Opensips gateway definition, you have "register" set to "false"
If you change it to "true", FreeSWITCH will authenticate on that gateway.

is that what you try to achieve?





________________________________
 From: James Mortensen <james.mortensen at synclio.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Saturday, September 28, 2013 2:26 AM
Subject: [Freeswitch-users] Sending SAVPF INVITE to Opensips
 


Hello,

I have a bandwidth.com number pointed to opensips, and a WebRTC peer registered with Opensips.  I'm trying to dial the 10 digit number from a cell phone and connect the call through FreeSWITCH to the Chrome WebRTC client.


I defined opensips as a gateway, in the external profile:

<include>
   <gateway name="ws-Opensips">
     <!-- <param name="from-user" value="fromuser"/> -->
     <param name="from-domain" value="54.X.X.75"/>
     <param name="proxy" value="54.X.X.75"/>
     <param name="expire-seconds" value="600"/>
     <param name="register" value="false"/>
     <param name="retry_seconds" value="30"/>
     <param name="extension" value="18257773456"/>
     <param name="context" value="public"/>
     <param name="avpf" value="yes"/>
     <param name="username" value="11234"/>
     <param name="password" value="password"/>
   </gateway>
</include>


In the public dialplan context, I added in a condition to catch the INVITE coming in from opensips and pass it to a context I've called "default-inbound". See the second condition:

 <extension name="from_opensips">
    <condition field="network_addr" expression="^54\.X\.X\.75$" break="never"> <!--CUSTOMIZE-->
      <action application="transfer" data="${destination_number} XML default"/>
    </condition>
    <condition field="network_addr" expression="^54\.X\.X\.111$"> <!--CUSTOMIZE Use a third context here -->
      <action application="transfer" data="${destination_number} XML default-inbound"/>
    </condition>
  </extension>


Then, in the default-inbound context, I match the dialed number, answer the call leg from the PSTN, and then try to transfer back through opensips to oversip and to Chrome.  The problem is that I either end up sending back AVP INVITES, or Opensips refuses to authenticate the user. 

<extension name="bandwidth.com inbound bridge">
    <condition field="destination_number" expression="^\+1(5035551212)$">

       <action application="answer" />

       <action application="set" data="variable_sip_auth_username=11234"/>
       <action application="set" data="variable_sip_auth_password=password"/>
       <action application="bridge" data="sofia/external/ws-Opensips/11234 at 54.X.X.75"/>

    </condition>
  </extension>


As you can see, I've hard-coded my peer, 11234, in the configuration. This is a registered user on Opensips.  

How can I get FreeSWITCH to send the SAVPF INVITE through to Opensips for the WebRTC portion of the call leg?  I apologize if this is covered somewhere, but I've been wracking my brain on this for days and am not getting anywhere.

The Opensips configuration I have works with existing Asterisk 11 servers, and I'm hoping I can just simply plug in FreeSWITCH servers seamlessly into the mix.

Thank you!


James



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