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

Reply via email to