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@synclio.com>
To: FreeSWITCH Users Help <freeswitch-users@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@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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
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