As I read this, the following should be correct: exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(60000))
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Wednesday, September 28, 2011 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute but there is no exemple for when i must put X in order to limit the call can you please give me an exemple regards 2011/9/28 Tarek Sawah <tareksa...@hotmail.com> have a look at the following: "L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional." source http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 _____ Date: Wed, 28 Sep 2011 17:59:27 +0000 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Limit outbond calls duration to 1 minute hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => 222,n,AbsoluteTimeout(60) exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => 222,n,Dial(SIP/${EXTEN},,KkTt) exten => 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com <http://www.api-digital.com/> -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com <http://www.api-digital.com/> -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users