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