if those two calls are all that are in the conference then you just need to uuid_bridge them. Ex from fs_cli<br><br><font style="font-family:courier new,monospace" size="2">uuid_bridge 8b0b8262-67df-11e1-89a5-2dc9f6a91475 8b09249a-67df-11e1-899d-</font><font size="1"><font><span style="font-family:courier new,monospace">2dc9f6a91475</span></font><br>
<br></font>You can also do an API from a Lua script:<br><font><span style="font-family:courier new,monospace">api = freeswitch.API()</span><br style="font-family:courier new,monospace"><span style="font-family:courier new,monospace">res = api:execute('uuid_bridge','8b0b8262-67df-11e1-89a5-2dc9f6a91475 8b09249a-67df-11e1-899d-</span></font><font><span style="font-family:courier new,monospace">2dc9f6a91475')</span><br>
</font><br>-MC<br><br><div class="gmail_quote">On Tue, Mar 6, 2012 at 3:28 PM, Bob McCarthy <span dir="ltr"><<a href="mailto:bob.mccarthy@experient.com">bob.mccarthy@experient.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<u></u>
<div>
I have some customers that are using SLA but want to Blind transfer the call after the supervisor drops out of the conference. (2 legs left in call) <br>
<br>
in Asterisk I would use the bridge command which would pull the channels out of the conference and bridge them together. I tried to do the same thing in Freeswitch from a lua script but I get the error: [ERR} switch_cpp.cpp:1277 Channels not ready<br>
<br>
<br>
<br>
I parsed the channel names out of the Conference list:<br>
Conference 8b09249a-67df-11e1-899d-2dc9f6a91475 (2 members rate: 16000)<br>
95;sofia/internal/<a href="mailto:sip%3ACO999x1001.3@192.168.57.102" target="_blank">sip:CO999x1001.3@192.168.57.102</a>;8b0b8262-67df-11e1-89a5-2dc9f6a91475;Outbound Call;<a href="tel:303-444-5555" value="+13034445555" target="_blank">303-444-5555</a>;hear|speak|talking|floor;0;0;0;200<br>
94;sofia/internal/<a href="mailto:101@192.168.57.31" target="_blank">101@192.168.57.31</a>;8b09249a-67df-11e1-899d-2dc9f6a91475;911 Trunk 01;<a href="tel:303-444-5555" value="+13034445555" target="_blank">303-444-5555</a>;hear|speak;0;0;0;200<br>
<br>
Some debug output to make sure I am using the correct channel names:<br>
channelarray[1]=sofia/internal/<a>sip:CO999x1001.3@192.168.57.102</a><br>
channelarray[2]=sofia/internal/<a href="mailto:101@192.168.57.31" target="_blank">101@192.168.57.31</a><br>
<br>
Here is the relevant script commands<br>
session1 = freeswitch.Session(channelarray[1]);<br>
session1 = freeswitch.Session(channelarray[1]);<br>
freeswitch.bridge(session1, session2);<br>
<br>
</div>
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<br></blockquote></div><br>