<?xml version="1.0" encoding="UTF-8"?><rss xmlns:dc="http://purl.org/dc/elements/1.1/" xmlns:content="http://purl.org/rss/1.0/modules/content/" xmlns:atom="http://www.w3.org/2005/Atom" version="2.0"><channel><title><![CDATA[Built in Softphone for (full) Operator's Console]]></title><description><![CDATA[<p dir="auto">Hi,</p>
<p dir="auto">We just got FOP2 up and running and haven't had that much time to work with it yet so my question may be premature but, in looking at it thus far I was surprised to see there isn't a built in soft-phone.  Our previous system was an Avaya and the operator worked the system directly from the console itself as an extension on the system.</p>
<p dir="auto">Any plans for that?</p>
]]></description><link>http://localhost:4567/topic/1201/built-in-softphone-for-full-operator-s-console</link><generator>RSS for Node</generator><lastBuildDate>Tue, 27 Oct 2026 04:03:57 GMT</lastBuildDate><atom:link href="http://localhost:4567/topic/1201.rss" rel="self" type="application/rss+xml"/><pubDate>Thu, 12 Jul 2012 18:47:41 GMT</pubDate><ttl>60</ttl><item><title><![CDATA[Reply to Built in Softphone for (full) Operator's Console on Mon, 18 Mar 2024 15:31:58 GMT]]></title><description><![CDATA[<p dir="auto">Reviving old thread sorry.<br />
<a class="plugin-mentions-user plugin-mentions-a" href="/user/nicolas" aria-label="Profile: nicolas">@<bdi>nicolas</bdi></a> "Also sip or pjsip devices need to be configured accordingly."<br />
We are currently unable to configure an extension to work with WebRTC and an hard phone at the same time.<br />
According to <a href="https://community.freepbx.org/t/fop2-webrtc-with-asterisk-18-3-0-and-freepbx15-working-settings-and-a-question/73872/2" title="this thread" rel="nofollow ugc">this thread</a> in freepbx forum, it's not possible. Any luck ?<br />
Having our agents use FOP2 at home and a hardphone in the office would be our ideal scenario.</p>
]]></description><link>http://localhost:4567/post/5411</link><guid isPermaLink="true">http://localhost:4567/post/5411</guid><dc:creator><![CDATA[francois]]></dc:creator><pubDate>Mon, 18 Mar 2024 15:31:58 GMT</pubDate></item><item><title><![CDATA[Reply to Built in Softphone for (full) Operator's Console on Tue, 03 Dec 2019 10:44:08 GMT]]></title><description><![CDATA[<p dir="auto">Hi Nicolas,<br />
Great news! I tried to configure it but I can't see any registration attempt. How can I debug the webrtc registration process?<br />
Asterisk has valid certificates.<br />
Thanks in advance,</p>
]]></description><link>http://localhost:4567/post/5410</link><guid isPermaLink="true">http://localhost:4567/post/5410</guid><dc:creator><![CDATA[santi.anton]]></dc:creator><pubDate>Tue, 03 Dec 2019 10:44:08 GMT</pubDate></item><item><title><![CDATA[Reply to Built in Softphone for (full) Operator's Console on Sun, 29 Sep 2019 22:00:29 GMT]]></title><description><![CDATA[<p dir="auto">Hi,</p>
<p dir="auto">Just reviving this old thread to let users know that as of today (2019), there is a free plugin (WebRTC phone) that adds softphone capabilities straight into the FOP2 switchboard.</p>
<p dir="auto"><img src="https://forum.fop2.com/uploads/editor/2u/nsl9yocjq5kb.png" alt="" class="img-markdown  img-fluid" /></p>
<p dir="auto">It requires Asterisk to be configured specifically for webrtc, as described in the <a href="http://www.fop2.com/docs/webrtc_guide.php" title="configuration guide" rel="nofollow ugc">configuration guide</a>. Also sip or pjsip devices need to be configured accordingly.</p>
<p dir="auto">Its a simple phone but gets the job done. With it you can create a zero configuration contact center in minutes, without the need to install and configure softphones or hardphones. Agents can just log into FOP2 and start working.</p>
<p dir="auto">Mixing the plugin with the FOP2 Dialer and Scripter you can have a complete solution for phone based call and contact centers. If you are not interested in outbound, then using the <a href="https://www.fop2.com/buy.php#queuelogin" title="Queue Login" rel="nofollow ugc">Queue Login</a> plugin will let you use FOP2 as the agent phone and console. And you can get full agent activity reporting using <a href="http://www.asternic.net" title="Asternic Call Center Stats." rel="nofollow ugc">Asternic Call Center Stats.</a></p>
]]></description><link>http://localhost:4567/post/5409</link><guid isPermaLink="true">http://localhost:4567/post/5409</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Sun, 29 Sep 2019 22:00:29 GMT</pubDate></item><item><title><![CDATA[Reply to Built in Softphone for (full) Operator's Console on Tue, 18 Dec 2012 02:41:33 GMT]]></title><description><![CDATA[<p dir="auto">We have an experimental version running rtmp for audio and freeswitch for rtmp to sip. It works, but it might not be production quality for phone calls (rtmp was not created for calls, but for audio/video streaming).</p>
<p dir="auto">We are playing also with webrtc. Asterisk 11 has native support for it. Browser are slowly adding that functionality. So, in a few months you will have audio and perhaps video integrated into fop2.</p>
]]></description><link>http://localhost:4567/post/5408</link><guid isPermaLink="true">http://localhost:4567/post/5408</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Tue, 18 Dec 2012 02:41:33 GMT</pubDate></item><item><title><![CDATA[Reply to Built in Softphone for (full) Operator's Console on Sat, 15 Dec 2012 05:59:30 GMT]]></title><description><![CDATA[<p dir="auto">Just a thought, invert the software!</p>
<p dir="auto">why not partner with a Softphone company (plenty of them around) and get them to integrate a browser within their OEM softphone, (say softphone on the right, browser on the left) which opens up the FOP2 screen as per IP setup in setting. The browser can also draw the SIP account details form the SIp phone..</p>
<p dir="auto">so it's one login to the windows based softphone which brings up the FOP2 panel and voila! you have one screen with all.</p>
<p dir="auto">thoughts?</p>
]]></description><link>http://localhost:4567/post/5407</link><guid isPermaLink="true">http://localhost:4567/post/5407</guid><dc:creator><![CDATA[illuder]]></dc:creator><pubDate>Sat, 15 Dec 2012 05:59:30 GMT</pubDate></item><item><title><![CDATA[Reply to Built in Softphone for (full) Operator's Console on Fri, 13 Jul 2012 01:25:58 GMT]]></title><description><![CDATA[<p dir="auto">Yes, there are plans, but the tech is not really there yet. Not sure how avaya worked, but fop2 is a "Zero install" operator console. There are no zero install softphones available right now that are good enough, but tech is getting closer.</p>
<p dir="auto">We have an experimental version that uses adobe flash and a special proxy to interface between rtmp clients and sip. The problem is that rtmp is not really made for real time communications but more for multimedia streaming.</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/5406</link><guid isPermaLink="true">http://localhost:4567/post/5406</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Fri, 13 Jul 2012 01:25:58 GMT</pubDate></item></channel></rss>