<?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[Asterisk 13 PJSIP with FOP 2.30.06 ]]></title><description><![CDATA[<p dir="auto">I would like to know if it is possible to run fop2 latest version with asterisk 13 with pjsip ? I'm using a vanilla installation.</p>
<p dir="auto">[SIP/100]<br />
type=extension<br />
extension=100</p>
<p dir="auto">when I try to use pjsip it make the button disappear</p>
<p dir="auto">[PJSIP/100]<br />
type=extension<br />
extension=100</p>
]]></description><link>http://localhost:4567/topic/2224/asterisk-13-pjsip-with-fop-2.30.06</link><generator>RSS for Node</generator><lastBuildDate>Tue, 27 Oct 2026 07:48:10 GMT</lastBuildDate><atom:link href="http://localhost:4567/topic/2224.rss" rel="self" type="application/rss+xml"/><pubDate>Tue, 26 Apr 2016 14:59:44 GMT</pubDate><ttl>60</ttl><item><title><![CDATA[Reply to Asterisk 13 PJSIP with FOP 2.30.06  on Fri, 06 May 2016 14:02:22 GMT]]></title><description><![CDATA[<p dir="auto">In that case, you can start FOP2 in debug mode and capture events to see if you have Peer registration events in PJSIP. Maybe there are no such events, or maybe they are different to the chan_sip ones and FOP2 needs to be updated to support them... if you can make a test try this:</p>
<p dir="auto">service fop2 stop<br />
cd /usr/local/fop2<br />
script capture.log<br />
./fop2_server -X 1</p>
<p dir="auto">At this point FOP2 will start and you will see output with AMI events on the console. Then log to FOP2 with your browser and check sip status. Then fire up some soft phone and register to Asterisk, then unregister (that should fire some AMI events about the SIP device status). Then stop capturing and resume normal FOP2 by issuing these commands:</p>
<p dir="auto">ctrl-C<br />
exit<br />
service fop2_start</p>
<p dir="auto">You will then end up with a capture.log file that I can inspect for such registration events. You can send that file via email to nicolas at house dot com dot ar or post it here (although it might contain some information that you might want to not publish in a public forum).</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/9773</link><guid isPermaLink="true">http://localhost:4567/post/9773</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Fri, 06 May 2016 14:02:22 GMT</pubDate></item><item><title><![CDATA[Reply to Asterisk 13 PJSIP with FOP 2.30.06  on Fri, 06 May 2016 13:57:13 GMT]]></title><description><![CDATA[<p dir="auto">Well when I log in fop2 and after that I restart asterisk the button show correctly if they are connected they show green if not the button are faded.</p>
<p dir="auto">But right not when I log in Fop2 they all show green even if the extension isn't connected.</p>
<p dir="auto">Yes, I do see call activity when i make call. red for busy, green for not busy</p>
]]></description><link>http://localhost:4567/post/9772</link><guid isPermaLink="true">http://localhost:4567/post/9772</guid><dc:creator><![CDATA[clinnovations]]></dc:creator><pubDate>Fri, 06 May 2016 13:57:13 GMT</pubDate></item><item><title><![CDATA[Reply to Asterisk 13 PJSIP with FOP 2.30.06  on Tue, 03 May 2016 19:35:31 GMT]]></title><description><![CDATA[<p dir="auto">You mean the sip peer registration status? Do you see call activity (red for busy, green for not busy)?</p>
]]></description><link>http://localhost:4567/post/9771</link><guid isPermaLink="true">http://localhost:4567/post/9771</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Tue, 03 May 2016 19:35:31 GMT</pubDate></item><item><title><![CDATA[Reply to Asterisk 13 PJSIP with FOP 2.30.06  on Tue, 03 May 2016 12:34:45 GMT]]></title><description><![CDATA[<p dir="auto">The extension button in fop2 with pjsip always stay green even if the endpoints isn't connected. What can I do to fix it ?</p>
]]></description><link>http://localhost:4567/post/9770</link><guid isPermaLink="true">http://localhost:4567/post/9770</guid><dc:creator><![CDATA[clinnovations]]></dc:creator><pubDate>Tue, 03 May 2016 12:34:45 GMT</pubDate></item><item><title><![CDATA[Reply to Asterisk 13 PJSIP with FOP 2.30.06  on Sat, 30 Apr 2016 18:44:13 GMT]]></title><description><![CDATA[<p dir="auto">Works fine with pjsip, but your button definition might be wrong, you must define always extension, context and label for a button to show, not matter if its sip or pjsip.</p>
<p dir="auto">Also, if you use any kind of filtering group, you will have to adapt/change the channel there also.</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/9769</link><guid isPermaLink="true">http://localhost:4567/post/9769</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Sat, 30 Apr 2016 18:44:13 GMT</pubDate></item></channel></rss>