similar to: CDR / billsec / originate / local chan

Hi All, I am in the process of migrating from 1.4.20 to 1.6.2.x and have stumbled across a number of "differences" between the 2 versions that are forcing me to use local channels in my dialplan (mainly centered around the different behavior of CDR fields in the 2 versions) . Previously, I would place a call via an AMI Originate action similar to: action:.Originate..

I am having a bit of a problem getting AMD to work on a new server. On my regular office server it works like a charm. I am running Asterisk 1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and I am using a SIP trunk to send out calls (the same one on both servers). Here is the output of a call on my office server: -- Attempting call on Local/0445540881644 at CC2 for

If i use AMD() or the code below, now the problem is the fax machine/modem detection and answer machine detection get detected as the same. If i need to seperate the two how do i do that? For example, if i use AMD() to detect an answer machine by saying any greeting exceeding 2.5 seconds is a machine, how do i distinguish between a fax/modem and a long greeting? ---- dave cantera

Hi All, I seem to have stumbled on a bit of a problem. When trying to send a fax with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the current svn version, with FFA 1.2 I get a core dump each time. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use)

Hi, I am pretty new to Asterisk. I am trying to make sure some human being answers the phone not the voice mail machine. How can I programmatically identify that? Here is my Sub: sub DialPhysician { my ($self, $con, $PhysicianPhone, $call_id, $conv_id) = (@_); to_log($self, "Inside Dial Physician", 2); my $DocPhone = "1".

it''s probably the iPhone. I mean, would you *really* want your phone to be connected to the internet all the time? On 8/28/07, richieright-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org <richieright-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote: > > > Hi, > > I met a strange problems with PeriodicalUpdater in iPhone. This timer > event will last only four or five

Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I'm using a Digium TE411P connected to a UK switch (EuroISDN). Everything is working, but if I dial a busy number (from SIP) is seems to stay busy until I hang up, even though the dial-plan drops through some other stuff using CALLSTATUS variable (i.e. S-BUSY), none of the timeouts come into play. Any ideas? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US

Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. #!/usr/bin/perl use strict; use

Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel? I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform APDU and drops the

Hello, In my asterisk box, i have a zaptel card connected to my analogic pstn line. I'm using a IAX2 client to call outside : IAX2 client <--> Asterisk <--> Zaptel card <----> France telecom line When checking cdr logs file, i always have an "ANSWER" on call status when call on this trunk, even if the final destination does not answer. Is "ANSWER" the

Dear all, I have a function that recognizes the following format for timestamps "%Y-%m-%d %H:%M:%S" my function takes two input arguments the TimeStart and TimeEnd I would like to help me create the right list with pairs of TimeStart and TimeEnd which I can feed to lapply (I am using mclapply actually).  For every lapply I want two inputs to be fed to my function. I only know how to

Hello I have this little dataset, my goal is create one column in the data.frame with between the diference DataTime and Duration. I'm using the next code to make do this TIME_STAMP SESSIONTIME Time TimeStart 1162343932 8320 2006-10-31 19:01:34 2006-10-31 16:42:54 1162343215 592 2006-10-31 19:02:04 2006-10-31 18:52:12 1162341465 11875

Hi, I've got a dialplan that transfers all outgoing calls to a Local channel before dialling out via SIP. I did this because sometimes i'm dialling two numbers at the same time and need to know which call is answered for billing purposes. However, I've just noticed that billsec is always equal to 0 even though i know the calls were answered. I now have to take the cdrs from my

Hi, Here is the scenario: Originating local channel using AMI. On answering the channel, it will goes to a context. Which start to playback a file. & after hangup at h extension I am caliing an agi script which insert CDR into DB. Now the problem is when I script hangsup during payback CDR(billsec) returns currect result. But when it hangsup after playback cdr(billsec) returns 0 . Please

Hi I am using Asterisk-1.6.1.6. Recently I had converted billsec and duration fields in my postgres 8.4 database from integer to numeric. I wanted to have better accuracy in calculating duration of the call. When the call is answered I can see better precision (for example 298.758421 sec), The problem is when the call is not answered (disposition is no answer) the duration is close to expected

We recently installed Asterisk 1.4.2 Tried to make calls using the Originate command (Asterisk Manager Interface) All of the calls have zero billsec in the CDR. Stumbled upon this: http://bugs.digium.com/view.php?id=8680 so I guess the fix is not yet in 1.4.2. Is this fixed in 1.4.3/1.4.4? -------------- next part -------------- An HTML attachment was scrubbed... URL:

When a call comes in and is parked using the Park() command it appears that the billing seconds do not include the time while the caller was parked. Is there a simple way to correct this with an option or setting in asterisk. This is not acceptable as we are loosing min when callers park calls. I am using 1.8.x release currently. Thanks zktech -------------- next part -------------- An HTML

Hello, I want to create php rate script and I'm using Deadagi. But I allways get billsec 0 , or nothing. Can you help me to solve this problem... My extension.conf: exten => _123,1,DeadAgi(rate.php) exten => _123,2,hangup And my simple test php script rate.php #!/usr/local/bin/php -q <?php include_once (dirname(__FILE__)."/phpagi.php"); $AGI = new AGI();

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: