Replace your phone number in place of ${EXTEN} and send it to your outgoing provider.
with same dial argument. On Thu, Sep 29, 2011 at 3:09 PM, salaheddine elharit < salah.elharit...@gmail.com> wrote: > ok thanks it's work fine > > now i have one question please > > it's work fine when i call extension 222 but i want to call any number > from my sip account 222 and the call hang up after 1 Min > > for exemple i call my mobile phone 067XXXXXXX using my sip 222 (x-lite) and > the call hangup after 1 min > > any help please > > thanks and regards > > > > 2011/9/28 Tarek Sawah <tareksa...@hotmail.com> > >> one adjustment i would suggest is using (|) instead of (,) >> >> >> exten => 222,n,Dial(SIP/${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 >> > > > -- > _____________________________________________________________________ > -- 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