<div dir="ltr">Hi Stanislav,<div><br></div><div>Opensips doesn&#39;t register our media servers, it just passed traffic through it and is all based on IP address.  What I&#39;m trying to do is get FreeSWITCH to take an AVP INVITE from Bandwidth.com, answer that call leg, then send an SAVPF INVITE to Opensips then to OverSIP and finally to a Chrome client.  Instead, it&#39;s sending an AVP INVITE.</div>

<div><br></div><div>Hence, the INVITE reaches my Chrome client endpoint, but Chrome responds, rightly so, with a 488 Not Acceptable Here because Chrome only supports SAVPF profiles.</div><div><br></div><div>In Asterisk, in the gateway configuration, I would put avpf=yes to tell Asterisk to send the INVITE with SAVPF, but on FreeSWITCH, since WebRTC is so new, and since I&#39;m only about a week into my experiences with FreeSWITCH, the documentation for this is tough to find.</div>

<div><br></div><div>Hope this helps clarify, and thanks for responding.</div><div><br></div><div class="gmail_extra"><div><div dir="ltr"><div><br></div><div>James<br></div><div><div><br></div></div></div></div><br><div class="gmail_quote">

On Sat, Sep 28, 2013 at 3:29 AM, Stanislav Sinyagin <span dir="ltr">&lt;<a href="mailto:ssinyagin@yahoo.com" target="_blank">ssinyagin@yahoo.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div><div style="font-size:10pt;font-family:arial,helvetica,sans-serif">in your ws-Opensips gateway definition, you have &quot;register&quot; set to &quot;false&quot;<br>If you change it to &quot;true&quot;, FreeSWITCH will authenticate on that gateway.<br>

<br>is that what you try to achieve?<br><br><div><span><br></span></div><div><br></div>  <div style="font-family:arial,helvetica,sans-serif;font-size:10pt"> <div style="font-family:times new roman,new york,times,serif;font-size:12pt">

 <div dir="ltr"> <hr size="1">  <font face="Arial"> <b><span style="font-weight:bold">From:</span></b> James Mortensen &lt;<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>&gt;<br>

 <b><span style="font-weight:bold">To:</span></b> FreeSWITCH Users Help &lt;<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt; <br> <b><span style="font-weight:bold">Sent:</span></b> Saturday, September 28, 2013 2:26 AM<br>

 <b><span style="font-weight:bold">Subject:</span></b>
 [Freeswitch-users] Sending SAVPF INVITE to Opensips<br> </font> </div> <div><div><div class="h5"><br><div><div dir="ltr">Hello,<div><br></div><div>I have a <a rel="nofollow" href="http://bandwidth.com/" target="_blank">bandwidth.com</a> number pointed to opensips, and a WebRTC peer registered with Opensips.  I&#39;m trying to dial the 10 digit number from a cell phone and connect the call through FreeSWITCH to the Chrome WebRTC client.</div>



<div><br></div><div><br></div><div>I defined opensips as a gateway, in the external profile:</div><div><br></div><div><div>&lt;include&gt;</div><div>   &lt;gateway name=&quot;ws-Opensips&quot;&gt;</div><div>     &lt;!-- &lt;param name=&quot;from-user&quot; value=&quot;fromuser&quot;/&gt; --&gt;</div>



<div>     &lt;param name=&quot;from-domain&quot; value=&quot;54.X.X.75&quot;/&gt;</div><div>     &lt;param name=&quot;proxy&quot; value=&quot;54.X.X.75&quot;/&gt;</div><div>     &lt;param name=&quot;expire-seconds&quot; value=&quot;600&quot;/&gt;</div>



<div>     &lt;param name=&quot;register&quot; value=&quot;false&quot;/&gt;</div><div>     &lt;param name=&quot;retry_seconds&quot; value=&quot;30&quot;/&gt;</div><div>     &lt;param name=&quot;extension&quot; value=&quot;18257773456&quot;/&gt;</div>



<div>     &lt;param name=&quot;context&quot; value=&quot;public&quot;/&gt;</div><div>     &lt;param name=&quot;avpf&quot; value=&quot;yes&quot;/&gt;</div><div>     &lt;param name=&quot;username&quot; value=&quot;11234&quot;/&gt;</div>



<div>     &lt;param name=&quot;password&quot; value=&quot;password&quot;/&gt;</div><div>   &lt;/gateway&gt;</div><div>&lt;/include&gt;</div></div><div><br></div><div><br></div><div>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&#39;ve called &quot;default-inbound&quot;. See the second condition:</div>



<div><br></div><div><div> &lt;extension name=&quot;from_opensips&quot;&gt;</div><div>    &lt;condition field=&quot;network_addr&quot; expression=&quot;^54\.X\.X\.75$&quot; break=&quot;never&quot;&gt; &lt;!--CUSTOMIZE--&gt;</div>



<div>      &lt;action application=&quot;transfer&quot; data=&quot;${destination_number} XML default&quot;/&gt;</div><div>    &lt;/condition&gt;</div><div>    &lt;condition field=&quot;network_addr&quot; expression=&quot;^54\.X\.X\.111$&quot;&gt; &lt;!--CUSTOMIZE Use a third context here --&gt;</div>



<div>      &lt;action application=&quot;transfer&quot; data=&quot;${destination_number} XML default-inbound&quot;/&gt;</div><div>    &lt;/condition&gt;</div><div>  &lt;/extension&gt;</div><div><br></div><div><br></div><div>



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. </div>



<div><br></div><div><div>&lt;extension name=&quot;<a rel="nofollow" href="http://bandwidth.com/" target="_blank">bandwidth.com</a> inbound bridge&quot;&gt;</div><div>    &lt;condition field=&quot;destination_number&quot; expression=&quot;^\+1(5035551212)$&quot;&gt;<br>



</div><div>       &lt;action application=&quot;answer&quot; /&gt;<br></div><div>       &lt;action application=&quot;set&quot; data=&quot;variable_sip_auth_username=11234&quot;/&gt;</div><div>       &lt;action application=&quot;set&quot; data=&quot;variable_sip_auth_password=password&quot;/&gt;</div>



<div>       &lt;action application=&quot;bridge&quot; data=&quot;sofia/external/ws-Opensips/11234@54.X.X.75&quot;/&gt;</div><div><br></div><div>    &lt;/condition&gt;</div><div>  &lt;/extension&gt;</div></div><div><br></div>



<div><br></div><div>As you can see, I&#39;ve hard-coded my peer, 11234, in the configuration. This is a registered user on Opensips.  </div><div><br></div><div>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&#39;ve been wracking my brain on this for days and am not getting anywhere.</div>



<div><br></div><div>The Opensips configuration I have works with existing Asterisk 11 servers, and I&#39;m hoping I can just simply plug in FreeSWITCH servers seamlessly into the mix.</div><div><br></div><div>Thank you!</div>



<div><div dir="ltr"><div><br></div><div>James<br></div><div><br><div><br></div></div></div></div></div></div></div><br></div></div>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br>

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<br></blockquote></div><br></div></div>