<div dir="ltr"><div>hi </div>
<div> </div>
<div>you can use MixMonitor like below</div>
<div> </div>
<div>in my case i use MixMonitor with SIP and i can record all calls inbound and outbound</div>
<div> </div>
<div>Best Regards</div>
<div> </div>
<div> </div>
<div>exten => 57,1,Answer()<br>exten => 57,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))<br>exten => 57,n,Dial(SIP/223)<br>exten => 57,n,Hangup() <br><br></div>
<div class="gmail_quote">2011/6/30 Ishfaq Malik <span dir="ltr"><<a href="mailto:ish@pack-net.co.uk">ish@pack-net.co.uk</a>></span><br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div class="im">On Thu, 2011-06-30 at 10:22 +0200, Peter Gelencser wrote:<br>> Hi,<br>><br>> I'm trying to record the incoming calls with Asterisk 1.6.2.15 and Dahdi<br>> 2.4.0. Without the line Record, everything works fine. If I add the line<br>
> Record, I can hear only a beep sound, the caller and the callee cannot<br>> hear each other and there is no recording file.<br>><br>> The extension in the extensions.conf:<br>><br>><br>> SZERVERSZOBA=DAHDI/19,,rtT<br>
><br>><br>> exten => 57,1,Record(${TIMESTAMP}${CALLERID(num)}-${EXTEN}.wav)<br>> exten => 57,n,Dial(${SZERVERSZOBA})<br>> exten => 57,n,Playback(vm-nobodyavail)<br>> exten => 57,n,Hangup()<br>
><br>> In the log:<br>><br>> -- Accepting call from '0620XXXXXXX to '57' on channel 0/1, span 2<br>> -- Executing [57@default:1] Record("DAHDI/4-1", "0620XXXXXXX-57.wav") in<br>
> new stack<br>> -- <DAHDI/4-1> Playing 'beep.alaw' (language 'en')<br>><br>><br>> What do I wrong? Should I set any other parameter than this? Thanks for<br>> you help in advance.<br>
><br>><br>> Best regards,<br>> Peter Gelencser<br>><br></div>Hi<br><br>You are using the wrong application, you need to use MixMonitor<br><br>Ish<br><font color="#888888">--<br>Ishfaq Malik<br>Software Developer<br>
PackNet Ltd<br><br>Office: 0161 660 3062<br></font>
<div>
<div></div>
<div class="h5"><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></div></div></blockquote></div><br></div>