similar to: Problem in Extension.conf

Hi, If I do this in extensions.conf exten => 411,1,Dial(IAX2/hhandresen@iaxtel/18005558355@iaxtel,40,rS(10)) the line is cut of in 10 sec., thats fine, but in CDR I got dst as T, and not 411. How can I handle this so I still get kicked of after 10 sec., but get 411 as dst in my cdr ? -- mvh. Hans-Henrik Andresen

If '#' is pressed during a call the CDR that is written at the end of the call contains '#' in the dst / destination field rather than the number that was originally called. How do I avoid losing that original number so that I can use the CDR for billing? I've tried not having a '#' target in extensions.conf and I've tried calling ResetCDR(w) in the

Hi all, Is there a way of disabling ALERTING message on a PRI channel? I have a problem .* is sending ALERTING message to the Nortel telco switch of my local provider BEFORE it dial the number it has to .if the number is busy or invalid there is no way we can tell this to the switch because it has already been told that the phone is ringing I am using asterisk Asterisk CVS-04/06/04-10:46:21 with

Hi, I am new on this field and I'm looking for some help I have this conf: [sip] exten => _333.,1,wait,3 exten => _333.,2,Answer exten => _333.,3,AbsoluteTimeout,7 exten => _333.,4,Hangup this works up to answer. then it can not stop after the 7 sec i specified in AbsoluteTimeout. Why? Any help? Regards Ghislain

Is there a way to get the que logs to go to a postgresql database? And/or are they any documents that explain the layout and field meanings? The logs look like they will be very helpful. Especially to show how long calls are held and who is taking them. I would like to make simple web reporting tool to show a daily summary status of queue activity. Also, there is not a manager command to do

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, I just wanted to point out that I have added a patch to Bug #1420 (http://bugs.digium.com/bug_view_page.php?bug_id=0001420), which adds Manager Events and Queue Log records for the AgentCallbackLogin application. Since the Bug Summary starts with "[Request]" and the Severity is "Trivial", both of which I can't change, this patch might be easy to overlook. Would

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

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

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 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

Hello I have contacted my line provider who is saying that in order to get my 0845 or 0870 number to id as the incoming number on a landline that i may call i need the following. User must provide - NPI set to E.163/E.164 User must provide - TON = "national or international I have had a good search around and can't seem to find a good answer to this. Does anyone have any idea where i

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