ok thanks it's work fine now i have one question please
it's work fine when i call extension 222 but i want to call any number from my sip account 222 and the call hang up after 1 Min for exemple i call my mobile phone 067XXXXXXX using my sip 222 (x-lite) and the call hangup after 1 min any help please thanks and regards 2011/9/28 Tarek Sawah <tareksa...@hotmail.com> > one adjustment i would suggest is using (|) instead of (,) > > > exten => 222,n,Dial(SIP/${EXTEN}||KkTtL(60000)) > > > > > Tarek Sawah > > Information Technology Adviser > > Integrated Digital Systems > > CCNP, MCSE, RHCE, TELECOM > > USA: +1 386 492 9993 > > > > ------------------------------ > Date: Wed, 28 Sep 2011 18:32:28 +0000 > > From: salah.elharit...@gmail.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute > > sorry but the issue still the same there is no hangup after 1Min > > regards > > 2011/9/28 Danny Nicholas <da...@debsinc.com> > > As I read this, the following should be correct:**** > > exten => 222,n,Dial(SIP/${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 -- 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**** > > ** ** > > -- > _____________________________________________________________________ > -- 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 > > > > -- _____________________________________________________________________ -- > 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 > > -- > _____________________________________________________________________ > -- 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 >
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