<div>Hi,</div>
<div>I was asked for the patch and I sent it. Does anybody have any news about this subject?</div>
<div>I'm willing to try a fix for 1.4 but I'd need any guidelines to do it.</div>
<div>Thanks in advanced<br>Jose<br></div>
<div class="gmail_quote">2009/4/2 Moises Silva <span dir="ltr"><<a href="mailto:moises.silva@gmail.com">moises.silva@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Async AGI was never released for Asterisk 1.4.X, so probably the patch<br>you used has a bug or something, do you still have the patch around?<br>
<br>Moy<br><br>On Thu, Apr 2, 2009 at 5:44 AM, <<a href="mailto:cyr2242@gmail.com">cyr2242@gmail.com</a>> wrote:<br>> Hi Henrik,<br>><br>> I would like to do the same thing you are doing here. I want to implement an external queue functionality so I need to stop a play file launched previously with an async agi command on caller's channel, sending the call to agent's extension.<br>
><br>> I'm redirecting caller's channel with REDIRECT while playing is taking place but I'm always getting a hang up on caller's channel.<br>><br>> I'm using:<br>><br>> asterisk-1.4.18<br>
> asterisk-addons-1.4.7<br>> async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4)<br>><br>> Both caller and agent are using 501 and 500 extensions and the async agi loop is waiting on 800, for example. The caller is dialing 800 where a play file is commanded through and async agi stream file command by the application.<br>
><br>> The relevant part of extensions.conf follows:<br>><br>> exten => _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN});<br>> exten => _5.,n,Wait(1);<br>> exten => _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);<br>
> exten => _5.,n,Hangup();<br>><br>> exten => _8.,1,Noop(every thing starting 8 ${EXTEN});<br>> exten => _8.,n,AGI(agi:async);<br>> exten => _8.,n,Hangup();<br>><br>> And the redirect command the application is sending to is:<br>
><br>> Action: Redirect<br>> Channel: SIP/501-081f0730<br>> Exten: 500<br>> Context: sip_sercom<br>> Priority: 1<br>><br>> Therefore, Henrik, could you show me your related dial plan and the redirect command you are sending? I wasn't able to see what I'm getting wrong.<br>
><br>> thanks in advanced<br>> Jose M Arias<br>><br>> --<br>> This message was sent on behalf of <a href="mailto:cyr2242@gmail.com">cyr2242@gmail.com</a> at openSubscriber.com<br>> <a href="http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10933120.html" target="_blank">http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10933120.html</a><br>
><br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</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><font color="#888888"><br>
<br><br>--<br>"I do not agree with what you have to say, but I’ll defend to the<br>death your right to say it." Voltaire<br></font></blockquote></div><br>