<?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[PIAF - Attended transfer not working]]></title><description><![CDATA[<p dir="auto">Hello,<br />
I have a PIAF purple server with asterisk Ver. 11.16.0 and FOP2 Version: 2.30.06<br />
The attended transfer button doesn't work; the original caller only hears the DTMF tones.<br />
I have polycom IP331 desk phones managed with the OSS plugin and the attended transfer works fine.</p>
<p dir="auto">I looked at freepbx Module Admin and I see "In-Call Asterisk Attended Transfer" enabled with the default value of *2<br />
I tried to change it to *# with no luck<br />
in feature.conf I have the attended transfer enabled.<br />
In FOP2 I tried with both my user account (fewer permission) as well as the administator account but the result is the same.</p>
<p dir="auto">What have I missed?</p>
]]></description><link>https://forum.fop2.com/topic/2349/piaf-attended-transfer-not-working</link><generator>RSS for Node</generator><lastBuildDate>Tue, 27 Oct 2026 20:02:57 GMT</lastBuildDate><atom:link href="https://forum.fop2.com/topic/2349.rss" rel="self" type="application/rss+xml"/><pubDate>Fri, 09 Sep 2016 14:42:12 GMT</pubDate><ttl>60</ttl><item><title><![CDATA[Reply to PIAF - Attended transfer not working on Mon, 12 Sep 2016 11:38:12 GMT]]></title><description><![CDATA[<p dir="auto">You must pass the tT options to dial options in Asterisk for feature code transfer to work. It is not a FOP2 issue, just the lack of tT options in dial string.</p>
<p dir="auto">Best regards,</p>
]]></description><link>https://forum.fop2.com/post/10314</link><guid isPermaLink="true">https://forum.fop2.com/post/10314</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Mon, 12 Sep 2016 11:38:12 GMT</pubDate></item></channel></rss>