<?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[Transfer Calls not working after FreePBX upgrade]]></title><description><![CDATA[<p dir="auto">My FreePBX server was a few versions behind, so I updated it.   It all went very smooth, but now FOP won't transfer calls.  I upgraded to 2.28, erased the management module from FreePBX.  (love the web based config) It functions well, I feel like there is a disconnect between FOP and Asterisk that I can't put my finger on.  The config files all have transfers turned on, as I have read older threads with similar problems.</p>
<p dir="auto">I also upgraded Asterisk to version 13.  Not sure if that caused some of the problems.</p>
<p dir="auto">What can I do to debug this problem?</p>
]]></description><link>http://localhost:4567/topic/1856/transfer-calls-not-working-after-freepbx-upgrade</link><generator>RSS for Node</generator><lastBuildDate>Tue, 27 Oct 2026 10:53:39 GMT</lastBuildDate><atom:link href="http://localhost:4567/topic/1856.rss" rel="self" type="application/rss+xml"/><pubDate>Thu, 20 Nov 2014 14:32:13 GMT</pubDate><ttl>60</ttl><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Thu, 18 Dec 2014 23:12:34 GMT]]></title><description><![CDATA[<p dir="auto">Your description fits almost exactly on the case I described above. How do your queue members look like? Are you using Local/xxx@yyy/n devices?</p>
<p dir="auto">asterisk -rx "queue show"</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/8071</link><guid isPermaLink="true">http://localhost:4567/post/8071</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Thu, 18 Dec 2014 23:12:34 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Thu, 18 Dec 2014 20:53:34 GMT]]></title><description><![CDATA[<p dir="auto">Incomming call (from dahdi) -&gt; direct to SIP extension -&gt; blind transfer to conference = all ok - external call in conference and SIP extension line released.</p>
<p dir="auto">Incomming call (from dahdi) -&gt; queue -&gt; SIP extension -&gt; blind transfer to conference = problem - external call in conference but SIP extension not released line.</p>
<p dir="auto">Screen in attach.</p>
]]></description><link>http://localhost:4567/post/8070</link><guid isPermaLink="true">http://localhost:4567/post/8070</guid><dc:creator><![CDATA[filippov]]></dc:creator><pubDate>Thu, 18 Dec 2014 20:53:34 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Thu, 18 Dec 2014 18:52:52 GMT]]></title><description><![CDATA[<p dir="auto">I am not sure if I understand correctly the "flow" of the problem. Not sure who are you transferring nor how into the conference?</p>
<p dir="auto">A line being busy after a transfer is made, and remaining busy until the transferred call finishes, is a classic asterisk "feature" when using Local/xxxx@yyyy/n channels. It is not a FOP2 problem, is a byproduct of /n in a Local channel inside Asterisk, and as such, cannot be fixed within FOP2, nor it is a FOP2 bug.</p>
<p dir="auto">However, I am not sure if your description of the problem has to do with that or not.</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/8069</link><guid isPermaLink="true">http://localhost:4567/post/8069</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Thu, 18 Dec 2014 18:52:52 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Thu, 18 Dec 2014 09:18:02 GMT]]></title><description><![CDATA[<p dir="auto">About spy in conference understand.<br />
I have error with blind transfer from user in queues to conference - after transfer incoming call to conference, sip line of user form queues is not released. The line of user remains busy until incoming call not complet.</p>
]]></description><link>http://localhost:4567/post/8068</link><guid isPermaLink="true">http://localhost:4567/post/8068</guid><dc:creator><![CDATA[filippov]]></dc:creator><pubDate>Thu, 18 Dec 2014 09:18:02 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Wed, 17 Dec 2014 13:13:32 GMT]]></title><description><![CDATA[<p dir="auto">You cannot spy on a conference in fop2, you can join the conference and listen in (same effect).</p>
]]></description><link>http://localhost:4567/post/8067</link><guid isPermaLink="true">http://localhost:4567/post/8067</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Wed, 17 Dec 2014 13:13:32 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Wed, 17 Dec 2014 12:25:25 GMT]]></title><description><![CDATA[<p dir="auto">No sounds in spy mode (listen / whisper) only in conferences. Button listen / whisper work well with other internal numbers. If i call 555 + number conference sounds is good, but it is inconvenient for the user.</p>
]]></description><link>http://localhost:4567/post/8066</link><guid isPermaLink="true">http://localhost:4567/post/8066</guid><dc:creator><![CDATA[filippov]]></dc:creator><pubDate>Wed, 17 Dec 2014 12:25:25 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Tue, 16 Dec 2014 20:12:58 GMT]]></title><description><![CDATA[<p dir="auto">You cannot spy on conferences? Only extensions. If you want to hear a conference, just join it.</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/8065</link><guid isPermaLink="true">http://localhost:4567/post/8065</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Tue, 16 Dec 2014 20:12:58 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Tue, 16 Dec 2014 19:37:59 GMT]]></title><description><![CDATA[<p dir="auto">All the same, there is no sound. Output for asterisk -rx "core show channels concise" with and without force_parameter_delimiter=','</p>
<pre><code>DAHDI/pseudo-922601214!default!s!1!Rsrvd!(None)!!!!!3!47!!1418758260.206057
DAHDI/i2/889658064166-31d!from-internal!STARTMEETME!4!Up!MeetMe!201,oTMr,!89658064166!!!3!30!!1418758277.206075
DAHDI/pseudo-2103507193!default!s!1!Rsrvd!(None)!!!!!3!47!!1418758260.206059
DAHDI/i2/88162280360-31c!from-internal!STARTMEETME!4!Up!MeetMe!201,oTMr,!88162280360!1418758252.206045!1418758252.206045!3!56!!1418758252.206045
SIP/1002-00000317!from-internal-increase-vol!!1!Up!ChanSpy!CONFERENCE/201,dq!1002!!!3!12!!1418758295.206085
</code></pre>
]]></description><link>http://localhost:4567/post/8064</link><guid isPermaLink="true">http://localhost:4567/post/8064</guid><dc:creator><![CDATA[filippov]]></dc:creator><pubDate>Tue, 16 Dec 2014 19:37:59 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Mon, 15 Dec 2014 12:19:02 GMT]]></title><description><![CDATA[<p dir="auto">asterisk -rx "core show channels concise"</p>
<p dir="auto">When you are listening to a call, look for the chanspy session and the parameter delimiter, is it a comma or a pipe, if a pipe, in fop2.cfg set force_parameter_delimiter=','</p>
]]></description><link>http://localhost:4567/post/8063</link><guid isPermaLink="true">http://localhost:4567/post/8063</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Mon, 15 Dec 2014 12:19:02 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Mon, 15 Dec 2014 06:13:53 GMT]]></title><description><![CDATA[<p dir="auto">Hello, transfer is now working, but no sound in listen/whisper mode. If cal to 555 (freepbx spy channel) + number extension - all good.</p>
]]></description><link>http://localhost:4567/post/8062</link><guid isPermaLink="true">http://localhost:4567/post/8062</guid><dc:creator><![CDATA[filippov]]></dc:creator><pubDate>Mon, 15 Dec 2014 06:13:53 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Fri, 12 Dec 2014 12:58:26 GMT]]></title><description><![CDATA[<p dir="auto">Can you send me privately your /etc/freepbx.conf file ? Or please contact me via the live help. I am online now.</p>
]]></description><link>http://localhost:4567/post/8061</link><guid isPermaLink="true">http://localhost:4567/post/8061</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Fri, 12 Dec 2014 12:58:26 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Fri, 12 Dec 2014 07:21:03 GMT]]></title><description><![CDATA[<p dir="auto">The autoconfig-users runs fine, but the autoconfig-buttons doesn't:</p>
<pre><code>./autoconfig-buttons.sh
# Problem connecting to mysql

! Cannot connect to Fo2 Manager database
</code></pre>
<p dir="auto">This is what it looks like now:</p>
<pre><code>if [ -e /etc/freepbx.conf ]; then
DBNAME=`cat /etc/freepbx.conf | sed 's/ //g' | grep AMPDBNAME | cut -d= -f2 | tail -n1`
DBUSER=`cat /etc/freepbx.conf | sed 's/ //g' | grep AMPDBUSER | cut -d= -f2 | tail -n1`
DBPASSLINE=`cat /etc/freepbx.conf | grep AMPDBPASS | tail -n1`
DBSTRIP=`echo $DBPASSLINE | cut -d= -f1`
DBPASS=`echo $DBPASSLINE | sed "s/$DBSTRIP=//g"`
DBHOST=`cat /etc/freepbx.conf | sed 's/ //g' | grep ^AMPDBHOST | cut -d= -f2 | tail -n1`
</code></pre>
]]></description><link>http://localhost:4567/post/8060</link><guid isPermaLink="true">http://localhost:4567/post/8060</guid><dc:creator><![CDATA[svan]]></dc:creator><pubDate>Fri, 12 Dec 2014 07:21:03 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Thu, 11 Dec 2014 17:44:19 GMT]]></title><description><![CDATA[<p dir="auto">Hi Svan,</p>
<p dir="auto">Try editing /usr/local/fop2/autoconfig-users.sh and remove the caret from the AMPDBNAME, AMPDBUSER and AMPDBPASS variables, so it looks like this (at the top of the file).</p>
<pre><code>if [ -e /etc/freepbx.conf ]; then
DBNAME=`cat /etc/freepbx.conf | sed 's/ //g' | grep AMPDBNAME | cut -d= -f2 | tail -n1`
DBUSER=`cat /etc/freepbx.conf | sed 's/ //g' | grep AMPDBUSER | cut -d= -f2 | tail -n1`
DBPASSLINE=`cat /etc/freepbx.conf | grep AMPDBPASS | tail -n1`
</code></pre>
<p dir="auto">Then do the same in <a href="http://autoconfig-buttons.sh" rel="nofollow ugc">autoconfig-buttons.sh</a></p>
<p dir="auto">To test it out, run the script in the command line and inspect the ouptut</p>
<p dir="auto">[fixed]<br />
/usr/local/fop2/autoconfig-users.sh<br />
[/fixed]</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/8059</link><guid isPermaLink="true">http://localhost:4567/post/8059</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Thu, 11 Dec 2014 17:44:19 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Thu, 11 Dec 2014 10:27:26 GMT]]></title><description><![CDATA[<p dir="auto">Hi, I'm having the same issue: call transfer and pickup do not work with fop2 2.28 and FreePBX 12. However after upgrading to 2.29 beta (the CentOS version, I get the following error when running "/usr/local/fop2/fop2_server --test":</p>
<p dir="auto">Flash Operator Panel 2 - White Label Version.<br />
unable to initialize libusb: -99<br />
Flash Operator Panel 2 - Valid License (7)</p>
<p dir="auto">ERROR 1102 (42000) at line 1: Incorrect database name ''<br />
ERROR 1102 (42000) at line 1: Incorrect database name ''<br />
Connection to manager OK!</p>
<p dir="auto">And no buttons display in fop2.<br />
Any ideas?</p>
]]></description><link>http://localhost:4567/post/8058</link><guid isPermaLink="true">http://localhost:4567/post/8058</guid><dc:creator><![CDATA[svan]]></dc:creator><pubDate>Thu, 11 Dec 2014 10:27:26 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Sun, 07 Dec 2014 20:40:20 GMT]]></title><description><![CDATA[<p dir="auto"><a href="http://download.fop2.com/fop2-2.29-debian-x86_64.tgz" rel="nofollow ugc">http://download.fop2.com/fop2-2.29-debian-x86_64.tgz</a></p>
<p dir="auto">Just download, extract and run make. The license will upgrade itself, you are free to upgrade for ONE year after purchase.</p>
]]></description><link>http://localhost:4567/post/8057</link><guid isPermaLink="true">http://localhost:4567/post/8057</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Sun, 07 Dec 2014 20:40:20 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Sun, 07 Dec 2014 10:31:14 GMT]]></title><description><![CDATA[<p dir="auto">OK, pls give me URL to FOP 2.29 Beta (Debian/Ubuntu x64). And what to do with the activation code.<br />
I have the key to 2.27. Two weeks ago I bought a key for upgrade to 2.28 (on main page <a href="http://fop2.com" rel="nofollow ugc">fop2.com</a> claimed that supported Asterisk 12).<br />
How to deal with the new version 2.29?</p>
]]></description><link>http://localhost:4567/post/8056</link><guid isPermaLink="true">http://localhost:4567/post/8056</guid><dc:creator><![CDATA[filippov]]></dc:creator><pubDate>Sun, 07 Dec 2014 10:31:14 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Sat, 06 Dec 2014 03:23:39 GMT]]></title><description><![CDATA[<p dir="auto">You can roll back to Asterisk 11 by running asterisk-version-switch and choosing 11. I just did that after having this problem and the buttons started working.</p>
]]></description><link>http://localhost:4567/post/8055</link><guid isPermaLink="true">http://localhost:4567/post/8055</guid><dc:creator><![CDATA[FreerPBXer]]></dc:creator><pubDate>Sat, 06 Dec 2014 03:23:39 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Wed, 03 Dec 2014 14:11:54 GMT]]></title><description><![CDATA[<p dir="auto">FOP 2.29 will address some changes in Asterisk 12 and 13. It is available for download, but in beta.</p>
]]></description><link>http://localhost:4567/post/8054</link><guid isPermaLink="true">http://localhost:4567/post/8054</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Wed, 03 Dec 2014 14:11:54 GMT</pubDate></item><item><title><![CDATA[Reply to Transfer Calls not working after FreePBX upgrade on Mon, 01 Dec 2014 19:48:11 GMT]]></title><description><![CDATA[<p dir="auto">Same problem for me. Asterisk 12.6.1 FreePBX 12.0.13 Ubuntu 14.10 FOP 2.28. Not working transfer (blind and attended) and no sound (and not working mic) in 'listen&amp;whisper' mode for conference (the button listen and 'listen&amp;whisper' are working).</p>
]]></description><link>http://localhost:4567/post/8053</link><guid isPermaLink="true">http://localhost:4567/post/8053</guid><dc:creator><![CDATA[filippov]]></dc:creator><pubDate>Mon, 01 Dec 2014 19:48:11 GMT</pubDate></item></channel></rss>