<div>Asterisk creates various channel names based on the situation, like the way call is invoked , if you are seeing the Local channel then call is actually invoked through Local channel driver . As I don't know whats your exact setup is and why you are expecting to see a SIP channel driver in channel names. After call has properly established just run the following command </div>
<div> </div>
<div>core show channels </div>
<div> </div>
<div>or </div>
<div> </div>
<div>show channels </div>
<div> </div>
<div>If the above command outputs the SIP channel name you are looking then you can fetch the same from AMI and set absolute time out on that. if you don't see that SIP/NT000 thing then Local channel name is what you really have to set absolute time out on.<br>
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<div class="gmail_quote">On Mon, Jan 24, 2011 at 12:49 AM, ABBAS SHAKEEL <span dir="ltr"><<a href="mailto:shakeel.abbas.qau@gmail.com">shakeel.abbas.qau@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div dir="ltr">Thank you very much for the prompt response.
<div><br></div>
<div>I also tried this i.e in beginning i used Noop(${CHANNEL}) to get channel name but it display some thing like ("Local/99449051010033@default-8307,2"</div>
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<div>But i was expecting some thing like SIP/NT000</div>
<div><br></div>
<div>I spend alot of time on it but still in vain
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<div class="h5"><br><br>
<div class="gmail_quote">On Sun, Jan 23, 2011 at 10:14 PM, Sherwood McGowan <span dir="ltr"><<a href="mailto:sherwood.mcgowan@gmail.com" target="_blank">sherwood.mcgowan@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
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<div></div>
<div>On Sun, Jan 23, 2011 at 1:04 PM, ABBAS SHAKEEL<br><<a href="mailto:shakeel.abbas.qau@gmail.com" target="_blank">shakeel.abbas.qau@gmail.com</a>> wrote:<br>> Hello all,<br>> I am trying to end a call after a specific time period for that reason i<br>
> have tried various options like using S, L in the dial command. But in vain.<br>> Now i am thinking to end the call using the AMI... but i am unable to get<br>> the current active channel.<br>> . i.e SIP/NT000 when i ask for getchannel it return some thing like<br>
> this Channel=Local/99449070380109@default-f08b,2<br>> As i have to use this channel name for hangup... Could some one let me know<br>> how to get channel name.... or any other way to end call..<br>> Thanks in advance<br>
><br>> --<br>> Best Regards<br>> Shakeel Abbas<br>><br>><br></div></div>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>><br>> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>><br><br><br>Easy solution is to store the channel name in a variable for later<br>
use, use ${CHANNEL} to get the channel name<br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br></div></div>-- <br>Best Regards<br>
Shakeel Abbas<br><br></div></div><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div>
<div><br><br clear="all"><br>-- <br>Thanks & Regards,<br>Godson Gera<br><a href="http://godson.in/">FreeSWITCH Asterisk Consultant</a><br></div>