Re: [asterisk-users] Limit outbond calls duration to 1 minute
Hi Thanks everyone for your help and support all works perfectly Best Regards 2011/9/29 salaheddine elharit salah.elharit...@gmail.com ok thanks for your response i will try that and i will update you as soon as i have any result best regards 2011/9/29 A J Stiles asterisk_l...@earthshod.co.uk (top-posting mess fixed the lazy man's way .) On Thursday 29 September 2011, salaheddine elharit 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 067XXX using my sip 222 (x-lite) and the call hangup after 1 min any help please What you have to do is create a new context in extensions.conf, and specify this in sip.conf as the default context from extension 222. Then, use the same KkTtL(6) options to your Dial() command(s) within this context. If there are some numbers that you want to be able to make unlimited-length calls to (other SIP phones that don't require going out via the PSTN, for example), just give them their own extension(s) without the KkTlL(6) . Remember, Asterisk always tries to match hardest first, i.e. fewest wild card characters first, irrespective of the actual order of lines in extensions.conf. -- AJS Answers come *after* questions. -- _ -- 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
Re: [asterisk-users] Limit outbond calls duration to 1 minute
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 067XXX 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/${EXTEN}||KkTtL(6)) Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- Date: Wed, 28 Sep 2011 18:32:28 + From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute sorry but the issue still the same there is no hangup after 1Min regards 2011/9/28 Danny Nicholas da...@debsinc.com As I read this, the following should be correct: exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6)) ** ** *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 + 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
Re: [asterisk-users] Limit outbond calls duration to 1 minute
Replace your phone number in place of ${EXTEN} and send it to your outgoing provider. with same dial argument. On Thu, Sep 29, 2011 at 3:09 PM, salaheddine elharit salah.elharit...@gmail.com wrote: ok thanks it's work fine now i have one question please it's work fine when i call extension 222 but i want to call any number from my sip account 222 and the call hang up after 1 Min for exemple i call my mobile phone 067XXX using my sip 222 (x-lite) and the call hangup after 1 min any help please thanks and regards 2011/9/28 Tarek Sawah tareksa...@hotmail.com one adjustment i would suggest is using (|) instead of (,) exten = 222,n,Dial(SIP/${EXTEN}||KkTtL(6)) 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 + 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/${EXTEN},,KkTtL(6)) ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine elharit *Sent:* Wednesday, September 28, 2011 1:23 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Limit outbond calls duration to 1 minute* *** ** ** but there is no exemple for when i must put X in order to limit the call* *** can you please give me an exemple regards 2011/9/28 Tarek Sawah tareksa...@hotmail.com have a look at the following: *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. source http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- Date: Wed, 28 Sep 2011 17:59:27 + From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Limit outbond calls duration to 1 minute ** ** hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten = 222,1,MixMonitor(sip_${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
Re: [asterisk-users] Limit outbond calls duration to 1 minute
(top-posting mess fixed the lazy man's way .) On Thursday 29 September 2011, salaheddine elharit 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 067XXX using my sip 222 (x-lite) and the call hangup after 1 min any help please What you have to do is create a new context in extensions.conf, and specify this in sip.conf as the default context from extension 222. Then, use the same KkTtL(6) options to your Dial() command(s) within this context. If there are some numbers that you want to be able to make unlimited-length calls to (other SIP phones that don't require going out via the PSTN, for example), just give them their own extension(s) without the KkTlL(6) . Remember, Asterisk always tries to match hardest first, i.e. fewest wild card characters first, irrespective of the actual order of lines in extensions.conf. -- AJS Answers come *after* questions. -- _ -- 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
Re: [asterisk-users] Limit outbond calls duration to 1 minute
ok thanks for your response i will try that and i will update you as soon as i have any result best regards 2011/9/29 A J Stiles asterisk_l...@earthshod.co.uk (top-posting mess fixed the lazy man's way .) On Thursday 29 September 2011, salaheddine elharit 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 067XXX using my sip 222 (x-lite) and the call hangup after 1 min any help please What you have to do is create a new context in extensions.conf, and specify this in sip.conf as the default context from extension 222. Then, use the same KkTtL(6) options to your Dial() command(s) within this context. If there are some numbers that you want to be able to make unlimited-length calls to (other SIP phones that don't require going out via the PSTN, for example), just give them their own extension(s) without the KkTlL(6) . Remember, Asterisk always tries to match hardest first, i.e. fewest wild card characters first, irrespective of the actual order of lines in extensions.conf. -- AJS Answers come *after* questions. -- _ -- 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
[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/${EXTEN},,KkTt) exten = 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit outbond calls duration to 1 minute
On 11-09-28 01:59 PM, salaheddine elharit wrote: hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten = 222,1,MixMonitor(sip_${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 *CLI core show application Dial Look at the 'L' flag -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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
Re: [asterisk-users] Limit outbond calls duration to 1 minute
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 + 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/${EXTEN},,KkTt) exten = 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit outbond calls duration to 1 minute
i have this when L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are left. Repeat the warning every 'z' ms. The following special variables can be used with this option: * LIMIT_PLAYAUDIO_CALLER yes|no (default yes) Play sounds to the caller. * LIMIT_PLAYAUDIO_CALLEE yes|no Play sounds to the callee. * LIMIT_TIMEOUT_FILE File to play when time is up. * LIMIT_CONNECT_FILE File to play when call begins. * LIMIT_WARNING_FILE File to play as warning if 'y' is defined. The default is to say the time remaining. but i don't understand what i can do to solve this thanks 2011/9/28 Paul Belanger pabelan...@digium.com On 11-09-28 01:59 PM, salaheddine elharit wrote: hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten = 222,1,MixMonitor(sip_${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 *CLI core show application Dial Look at the 'L' flag -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- __**__**_ -- 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-**usershttp://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
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 + 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/${EXTEN},,KkTt) exten = 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit outbond calls duration to 1 minute
As I read this, the following should be correct: exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6)) 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 + 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/${EXTEN},,KkTt) exten = 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit outbond calls duration to 1 minute
exten = 222,n,Dial(SIP/${EXTEN},,KkTtLL(6:3:1)) this will call the extension and sets the limit to 6MS which equals 60 seconds.. and will inform the caller of his remaining time when he has only 30 seconds left.. and will repeat the notification every ten seconds (this is an over do and playing such sounds files at this rate will consume the resources!) Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 28 Sep 2011 18:22:57 + From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com 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 + 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/${EXTEN},,KkTt) exten = 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit outbond calls duration to 1 minute
sorry but the issue still the same there is no hangup after 1Min regards 2011/9/28 Danny Nicholas da...@debsinc.com As I read this, the following should be correct: exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6)) ** ** *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 + 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/${EXTEN},,KkTt) exten = 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit outbond calls duration to 1 minute
one adjustment i would suggest is using (|) instead of (,) exten = 222,n,Dial(SIP/${EXTEN}||KkTtL(6)) 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 + 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/${EXTEN},,KkTtL(6)) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Wednesday, September 28, 2011 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute but there is no exemple for when i must put X in order to limit the call can you please give me an exemple regards 2011/9/28 Tarek Sawah tareksa...@hotmail.com have a look at the following: L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. source http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 28 Sep 2011 17:59:27 + From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Limit outbond calls duration to 1 minute hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten = 222,1,MixMonitor(sip_${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