<div dir="ltr"><span class="Apple-style-span" style="border-collapse: collapse; "><div>I have a call between two people. I know their channel identifier. I want to trasfer a call away from one person and pass it to another person.</div>
<div> </div><div>To start, let's talk about a <span class="nfakPe" style="background-image: initial; background-repeat: initial; background-attachment: initial; -webkit-background-clip: initial; -webkit-background-origin: initial; background-color: rgb(255, 255, 136); background-position: initial initial; ">blind</span> <span class="nfakPe" style="background-image: initial; background-repeat: initial; background-attachment: initial; -webkit-background-clip: initial; -webkit-background-origin: initial; background-color: rgb(255, 255, 136); background-position: initial initial; ">transfer</span>. My system places both outgoing calls to people and bridges them together (cheaper, works via AGI).</div>
<div> </div><div>Action: Redirect</div><div>Channel: <prospect></div><div>ExtraChannel: 0</div><div>Exten: SIP/transfer_to</div><div>Context: default</div><div>Priority: 1</div><div> </div><div>So that works just fine. I'm having an issue however that when the person who was orginally talking decides to hang up his call, Asterisk disconnects the other line as well, as if the ownership of that line is still controled by the orginal process. I'd love to solve that problem. Maybe putting the SIP/transfer_to into the ExtraChannel and then transfering them to a conference room. Suggestions welcome. Could also be that AGI maintains control of any channels it creates and when the main calling line dies, it kills all the others even if they've been transfered away.</div>
<div> </div><div>Okay, in the end, I'd like this to be assisted <span class="nfakPe" style="background-image: initial; background-repeat: initial; background-attachment: initial; -webkit-background-clip: initial; -webkit-background-origin: initial; background-color: rgb(255, 255, 136); background-position: initial initial; ">transfer</span>. Place the party on hold, call another party, and then bridge the two together. Whenever a channel is taken away from the current person, the call status is returned and my AGI script can continue. So I think it should be fine. Has anyone done anything like this? Any pointers would be great.</div>
<div><br></div><div>PS: (update since I wrote this original message a while back), via the web, you click a link. That creates a CALL file which calls your number. Once connected, it passes it to an extension that spawns an AGI program. That AGI program looks in the database for the number you wanted to call and places that phone call. You than chat with that person and decide that you're done with that call and want to go onto your next phone call. I use the Asterisk Manager Interface (AMI) to perform a Redirect on the person you're talking to. Doing this causes the AGI script to continue.</div>
</span><br>-- <br>Nicholas Blasgen<br><a href="mailto:nicholas@blasgen.com" target="_blank">nicholas@blasgen.com</a><br>408.497.9796 (c)<br>
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