<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.
<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@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!