similar to: Dropped Calls

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 =>

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,Absolutetimeout(20) exten =>

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

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

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

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

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

Hi, I still cant dial out on Zap and I really have no clue 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 =>

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

I have no idea what this really should be called, so for lack of a better name, I called it "reverse hold". Hopefully someone else can make use of it, or even make it better, as its the first thing of its kind I've made for asterisk. Like most people, I'm very busy, so when I call other companies, sitting on hold really sucks. If you have speaker phone, its not so bad, but then

Hi , I''m getting random hangups on zap channels with long calls. It seems that the hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any other thing I should be configuring? Thanks! PHM

I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 call to 404 2.- from 404 don't answer it. 3.- from 402 press *8

I'm not sure if it's the same problem but your error message likely the same. after i additing pridialplan=local in the zapata.conf i'm able to make outboundcalls (located in germany) marcus -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Gesendet: Dienstag, 11. April 2006 16:33 An:

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, I'm trying to setup a small BRI ISDN <-> voip gateway. The ISDN card is based on Cologne chipset, so I try set it up with zaphfc. The versions i'm running: kernel-2.4.27 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e zaptel modules 1.0.7 zaphfc is from bristuff-0.2.0-RC8e When I'm doing the insmod on zaptel, zaphfc, zaprtc: Zapata Telephony Interface Registered on major 196 PCI:

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

Hi, I'm having a major problem with random calls dropping. After spending weeks trying to figure it out, i've finally spotted the issue but don't know how to resolve it. I run a sip server that's hosted in a data centre. It has a public IP address with no nat involved. My provider also has a public ip with no nat involved. The sip phones are in a remote office behind a nat