<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:courier,monaco,monospace,sans-serif;font-size:12pt">please retest with latest trunk as a few issues were fixed with transfers today.<br><br><br>Be aware that the transfers work best with bridged calls.<br>It sounds like that is what you are doing so it should be ok but there are a few things to consider when calling an extension that is just "playback" for example.<br><br>Say you call an extension that has playback foo.wav:<br><br>Technically the file is playing on the inbound leg and this is considered a 1 legged call.<br><br>so if you blind transfer this extension there is nothing to transfer cos there is only 1 leg so in this case we will not allow it.<br><br>if you attended transfer this extension, we will transfer the new channel you called to the same extension as the one you were on and hangup the original so the playback will start over for
the called party.<br><br><br>if you want to call and ivr or playfile extension and then transfer it you have to make a hairpin call back into the system.<br><br>so instead of making 1000 lead to playback foo.wav<br>make it lead to sofia/default/playback-foo@same.box<br><br>then you have 2 legs and it's possible to transfer the playback.<br><br><br><br><br><br><br><div> </div><div>Anthony Minessale II<br><br><span>FreeSWITCH <a target="_blank" href="http://www.freeswitch.org/">http://www.freeswitch.org/</a></span><br><span>ClueCon <a target="_blank" href="http://www.cluecon.com/">http://www.cluecon.com/</a></span><br><br>AIM: anthm<br>MSN:anthony_minessale@hotmail.com<br>GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<br>IRC: irc.freenode.net #freeswitch</div><div><br>FreeSWITCH Developer
Conference<br>sip:888@conference.freeswitch.org<br>iax:guest@conference.freeswitch.org/888<br>googletalk:conf+888@conference.freeswitch.org<br>pstn:213-799-1400</div><div style="font-family: courier,monaco,monospace,sans-serif; font-size: 12pt;"><br><br><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;">----- Original Message ----<br>From: Jonas Gauffin <jonas.gauffin@gmail.com><br>To: freeswitch-users@lists.freeswitch.org<br>Sent: Thursday, October 25, 2007 6:55:45 AM<br>Subject: Re: [Freeswitch-users] Call problems<br><br>
Ok. I've created a jira entry.<br><br>On 10/25/07, Michael Jerris <<a ymailto="mailto:mike@jerris.com" href="mailto:mike@jerris.com">mike@jerris.com</a>> wrote:<br>> We would need a full trace from freeswitch including the TPORT_LOG to<br>> determine that.<br>><br>><br>> On 10/25/07, Jonas Gauffin < <a ymailto="mailto:jonas.gauffin@gmail.com" href="mailto:jonas.gauffin@gmail.com">jonas.gauffin@gmail.com</a>> wrote:<br>> ><br>> > Hello<br>> ><br>> > I have three phones, all registered with FreeSwitch.<br>> ><br>> > I did the following:<br>> > a) Called from phone A to phone B<br>> > b) B picked up<br>> > c) A put B on hold<br>> > c) A called phone C<br>> > d) C picked up<br>> > e) Pressed transfer on A's phone<br>> ><br>> > Both B and C got disconnected instead of being connected to each
other.<br>> ><br>> > I did another variant:<br>> > A calls B, B calls C, B presses transfer.<br>> > A things that he is still connected, B and C gets disconnected.<br>> ><br>> > Are these problems in my phones, or are they FS problems?<br>> ><br>> > //Jonas<br>> ><br>> > _______________________________________________<br>> > Freeswitch-users mailing list<br>> > <a ymailto="mailto:Freeswitch-users@lists.freeswitch.org" href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>> ><br>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>> ><br>>
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