<?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[Supervised transfer]]></title><description><![CDATA[<p dir="auto">Hi ,</p>
<p dir="auto">Having small issues, when our reception do supervised transfer and end up can get the customer call line back in below case</p>
<p dir="auto">Reception(9000) highlight extension ex 9001 and press supervised transfer button on FOP2 and put client on hold<br />
9001 rining and pressed 'x' on the phone reject the call<br />
9000 end up went to 9001's voice message bank and asking for leaving the message and client's line is not coming back.</p>
<p dir="auto">I tried many ways and can't get the client line back to reception line if user pressed 'x' on the phone.<br />
At moment reception is using pick up link from Main Trunk to get that line back .</p>
<p dir="auto">Is there any smart way to fix this?</p>
<p dir="auto">Thanks<br />
Jacky</p>
]]></description><link>http://localhost:4567/topic/2463/supervised-transfer</link><generator>RSS for Node</generator><lastBuildDate>Tue, 27 Oct 2026 13:08:43 GMT</lastBuildDate><atom:link href="http://localhost:4567/topic/2463.rss" rel="self" type="application/rss+xml"/><pubDate>Tue, 10 Jan 2017 02:07:35 GMT</pubDate><ttl>60</ttl><item><title><![CDATA[Reply to Supervised transfer on Sat, 14 Jan 2017 02:23:42 GMT]]></title><description><![CDATA[<p dir="auto">The PlayDTMF manager action does not work as you would expect, you cannot use it to "mimic" dialing from a real phone. Never had success using it that way.</p>
]]></description><link>http://localhost:4567/post/10853</link><guid isPermaLink="true">http://localhost:4567/post/10853</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Sat, 14 Jan 2017 02:23:42 GMT</pubDate></item><item><title><![CDATA[Reply to Supervised transfer on Fri, 13 Jan 2017 05:01:08 GMT]]></title><description><![CDATA[<p dir="auto">Hi altsupport,</p>
<p dir="auto">Are you using vanilla asterisk ? If yes, edit features.conf. There is a star code there to cancel the supervised transfer. Not sure if it's the default but on our system we use *0.</p>
<p dir="auto">It would be a very good improvement if FOP2 would add a cancel button that would send the dtmf we define using the AMI on the live channel. That way we could push a button to have the transfer cancel.</p>
<p dir="auto">Keep me posted if you can't find the starcode or your system. You could print the output of 'core show features'</p>
]]></description><link>http://localhost:4567/post/10852</link><guid isPermaLink="true">http://localhost:4567/post/10852</guid><dc:creator><![CDATA[Dimitripietro]]></dc:creator><pubDate>Fri, 13 Jan 2017 05:01:08 GMT</pubDate></item><item><title><![CDATA[Reply to Supervised transfer on Wed, 11 Jan 2017 11:13:51 GMT]]></title><description><![CDATA[<p dir="auto">I think the asterisk dialplan should be modiifed in this case, so if a supervised transfer is detected, it won't send the call to voicemail if its not answered, to allow the original call to come back to the transferee.</p>
<p dir="auto">If you attempt to cancel the transfer once voicemail kicks in, I guess you are lost.. you can try to dial the hangup feature code from the phone (could be ## or ** , that is configurable in features.conf).</p>
<p dir="auto">Best regards,</p>
]]></description><link>http://localhost:4567/post/10851</link><guid isPermaLink="true">http://localhost:4567/post/10851</guid><dc:creator><![CDATA[nicolas]]></dc:creator><pubDate>Wed, 11 Jan 2017 11:13:51 GMT</pubDate></item></channel></rss>