search for: absolutetimeout

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten => s,2,AGI(attended-extension,${ARG1},${A...

Hi All, I've been having a hard time getting the AbsoluteTimeout function to work. Is this Function working in for SIP? I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf. I've also been running these test on ver 0.5.0 exten => _X.,1,Absolutetime...

Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=>_.,1,Dial,Zap/2|1 exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup the extension.conf which does not is like this; [incomingSIP]

...Brazil that was 60 seconds of an employee calling a customer, then 18 hours and 47 minutes of background noise in their office. The Cisco 7960's have an issue where you sometime don't realize the phone is still off hook as was the case for this call. I'd like to use some options like AbsoluteTimeout, which would jump in the middle of a call every 60 minutes, and ask the calling party (for outbound calls only) to press 1 to continue. The problem I find with AbsoluteTimeout is that the call is disconnected BEFORE it reaches the T extension. Is there anyway to temporarily interrupt the call then...

Is AbsoluteTimeout designed to work with canreinvite=yes? If not, are the any other options for disconnecting a call after a predefined duration when using canreinvite=yes? Thanks! David

Hi, Have a problem in my extension.conf: I have: [sip] exten => _333.,1,wait,3 exten => _333.,2,Answer exten => _333.,3,AbsoluteTimeout,7 exten => _333.,4,Hangup I wanted to test if * is executing this dial plan by calling 3335254255 for example. The problem is as follow: It waits, it answers but it does not seems to see the Absolutetimeout: call goes forever. What's wrong? Am I missing something? (I am new is this field!...

Does anyone know if AbsoluteTimeout is working completely? As far as I can see on my systems, I'm still getting occasional hung SIP channels, even though there should be nothing over my setting... cheers Sherwood -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/...

I'm using the M() option to Dial() and having problems. In the following dialplan example ANY digit exits the macro. When the callee presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run. Does anyone have any ideas as to what I'm doing wrong? Asterisk 1.2.x [extensions] exten => 2998,1,Dial(Zap/1/5551212,,wM(answer-confirmation^20)) [macro-answer-confirmation] exten => s,1,Noop(Set AbsoluteTimeout(${ARG1}) exten => s,n,Background(/etc/asterisk...

Greetings, I am running Asterisk with a T100P and a Zhone channel bank for over a month now. For the most part it works fine but from time to time (about once a week) the system will not let go of a line and will play the greeting over and over. Anyone calling gets a busy signal. If I reset Asterisk everything works fine. Has anyone seen this problem before and fixed it? If so what did you do?

...tered action Hangup == Manager registered action Status == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 10000 -> 20000 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Re...

Hello Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030821/c1ca1383/attachment.htm

Hi, I am trying to get priority + 101 to work with Dial application. My dial plan is like this: [dial-mobile-peak] exten => s,1,AbsoluteTimeout(${ABSOLUTETIMEOUT}) exten => s,2,Dial(${TRUNKONE}${CALLEDNO:1}) exten => s,103,AbsoluteTimeout(${ABSOLUTETIMEOUT}) exten => s,104,Dial(${TRUNKTWO}${CALLEDNO:1}) I have changed password for first trunk to simulate trunk failure. Trunk one tries to dial, it gets rejected by remote peer and...

Ok I tried but it does not work: now the settings are as follow exten => _123.,1,Answer exten => _123.,2,AGI(test.agi) exten => T,1,hangup the AbsoluteTimeout(5) is in test.agi (PHP) I put "AbsoluteTimeout" before "Answer": when i call for e.g 123456 it tries upto timing out. So I put again "Answer" and then "AbsoluteTimeout" then the last AGI command runs ok (result 200 ok) but it does not stops in the specified...

Hello everybody. Recently I've been trying to limit the duration of some calls for a simple application I'm writing. Unfortunately all of the documented methods are failing and I'm not sure what else to try. Here are some samples of what I've done: 1) The AbsoluteTimeout application. - exten => 1,1,AbsoluteTimeout (30) 2) The new version of AbsoluteTimeout. - exten => 1,1,Set(TIMEOUT(absolute)=30) 3) The S option to the Dial application. - exten => 1,1,Dial(${PROVIDER}/${NUMBER}|30|S(30)) 4) The L option to the Dial application. - exten =&gt...

This probably came up before, but I have a faxing question for everyone. I have a simple extension setup to use rxfax to receive faxes sent to asterisk. It is: exten => s,1,Answer() exten => s,n,AbsoluteTimeout(300) exten => s,n,Set(FAXFILE=/var/spool/asterisk/fax/${ARG1}_${CALLERIDNUM}_${UNIQUEID}.tif) exten => s,n,rxfax(${FAXFILE}) exten => s,n,System(/usr/bin/mailfax "${ARG1}" "${FAXFILE}" "${ARG2}") exten => s,n,Hangup() exten => T,1,Hangup() I read that y...

...lue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten => 111,n,Dial(Zap/G1/002212601574) exten => 111,n,NoOp(${DIALSTATUS}) exten => 111,n,Busy() exten => 111,n,Hangup() My zapata.conf is like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no echocancel=yes echocancelwhenbridged=yes ;...

...tered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Not found (No such file or directory) Unable to open management configuration manager.conf. Call management disabled. == Parsing '/etc/asterisk/rtp.conf&...

...vices is the ability to set a time limit >on duration of calls (esp. outgoing, for each station) and usually a warning >beep is played few seconds before time runs out. as far as i could >understand it's possible to set a time limit on calls using something like: >exten => 2000,1,AbsoluteTimeout(20) >but that's only on *incoming* calls and even so, the call is disconnected >without any warning beeps or anything. so i was wondering, am i missing >something or is this feature really not availabe in asterisk? [moved to asterisk-users, where this belongs] You are correct about...

In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/6eb392cc/attachment.htm