<?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[Intermitent transfer problem]]></title><description><![CDATA[<p dir="auto">Got several complaints that when the operator is on a call and another one comes in, they will answer the call, the original call goes on hold, they click on the destination user than click on the transfer button intending to transfer the caller on the line  BUT the actual caller transferred is the one on hold NOT the one live on the line.</p>
<p dir="auto">This does not happen all the time, anyone seeing this?<br />
best yet how do I fix it? <img src="http://localhost:4567/assets/plugins/nodebb-plugin-emoji/emoji/android/1f61e.png?v=9c14fadbfa0" class="not-responsive emoji emoji-android emoji--disappointed" style="height:1.4em;width:auto;vertical-align:middle" title=":(" alt="😞" /></p>
]]></description><link>http://localhost:4567/topic/261/intermitent-transfer-problem</link><generator>RSS for Node</generator><lastBuildDate>Tue, 27 Oct 2026 08:49:45 GMT</lastBuildDate><atom:link href="http://localhost:4567/topic/261.rss" rel="self" type="application/rss+xml"/><pubDate>Wed, 17 Feb 2010 13:15:59 GMT</pubDate><ttl>60</ttl><item><title><![CDATA[Reply to Intermitent transfer problem on Wed, 17 Feb 2010 15:39:21 GMT]]></title><description><![CDATA[<p dir="auto">In sip_general_custom.conf set</p>
<p dir="auto">callevents=yes</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/1225</link><guid isPermaLink="true">http://localhost:4567/post/1225</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Wed, 17 Feb 2010 15:39:21 GMT</pubDate></item><item><title><![CDATA[Reply to Intermitent transfer problem on Wed, 17 Feb 2010 15:33:58 GMT]]></title><description><![CDATA[<p dir="auto">I use polycom 430.<br />
I have not made any changes to the sip.conf<br />
what is the exact setting I need to put there? I use freepbx (piaf) and changes to file directly can be overwritten.<br />
if you think this can cure my issue where can I add this to?<br />
One of those perhaps?<br />
sip_custom.conf<br />
sip_custom_post.conf<br />
sip_general_additional.conf<br />
sip_general_custom.conf</p>
]]></description><link>http://localhost:4567/post/1224</link><guid isPermaLink="true">http://localhost:4567/post/1224</guid><dc:creator><![CDATA[sergiocesar]]></dc:creator><pubDate>Wed, 17 Feb 2010 15:33:58 GMT</pubDate></item><item><title><![CDATA[Reply to Intermitent transfer problem on Wed, 17 Feb 2010 15:17:37 GMT]]></title><description><![CDATA[<p dir="auto">Hi Sergio,</p>
<p dir="auto">FOP2 will try to transfer the active channel only. What kind of phones are you using? SIP? Have you enabled callevents in sip.conf ? Best regards,</p>
]]></description><link>http://localhost:4567/post/1223</link><guid isPermaLink="true">http://localhost:4567/post/1223</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Wed, 17 Feb 2010 15:17:37 GMT</pubDate></item></channel></rss>