similar to: CDR on MySQL

Hey gang, Currently I have this dialplan: exten => _9.,1,Dial(blah/blah) exten => h,1,ResetCDR(w) exten => h,2,NoCDR() exten => h,3,DeadAGI(rate_call.php) The AGI script takes the completed call, determines all that NPNAXX crap, finds the cost and then updates the CDR with the cost. Problem is, I keep getting these messages: Jan 4 13:25:36 WARNING[13689]: cdr.c:114

I am using a2billing as billing software ,and I make an 800 call service which means that the destination extension should be build I put this code at extensions.conf exten => 99909994,1,SetAccount(2704714849) exten => 99909994,2,Wait,2 exten => 99909994,3,DeadAGI(a2billingp.php) exten => 99909994,4,Wait,2 exten => 99909994,5,Hangup its not stable ,its works for 3 times

Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone

For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to

Hi, I have some problem to get this setup working. I have a CAC Channel Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2) and I have a TE110p installed in this box. What I want to do is, just to be able to dial one of those lines that already are connected to the channel bank, and transfer that call through TE110p and Asterisk to a user agent somewhere through Internet.

Hello, Asterisk party, If block the call before dialing (Hangup()), CDR's don't write to MySQL or CSV. Otherwise, if Hangup() is after Dial(), CDRs write normally. Here is the dialplan: ; we skipped dial, because the number is "blocked" exten => _X.,n(Finish),Hangup() exten => h,1,NoOP("hangup") exten => h,2,ResetCDR(w) exten => h,n,NoCDR() exten =>

(Both on Asterisk 1.2 and 1.4) I was struggling to find out why my CDR was recording dst = h after a call hangup. It was working fine until I added a GotoIf statement before ResetCDR to calculate some value for userfield column. Today I tested and found out that if ResetCDR is put after GotoIf (or after if in AEL), it doesn't record correct value in dst column, and isntead puts 'h'

Hi all, I'm new in asterisk and I got to set up a dundi config for my work. I have 2 PBX for the test, the two PBX are in the same local network PBX A : 192.168.199.23 PBX B : 192.168.199.21 my config files : (on PBX B , the config files on PBX A looks like it) /etc/asterisk/dundi.conf [general] bind=192.168.199.21 port=4520 cachetime=5 ttl=32 autokill=yes entityid=00:30:18:4C:33:53

Hi, I want to count the number of open Zap channels on my server. [outgoingzap] exten => _0NXXXXXX,1,NoOp(Outgoing Local - 7 digs - ${EXTEN:1}) exten => _0NXXXXXX,2,Set(ZAP01=$[${ZAP01} + 1]|g) exten => _0NXXXXXX,3,Set(UPDATED=true) exten => _0NXXXXXX,4,Dial(${TRUNK}/${EXTEN},60) exten => _0NXXXXXX,6,Busy exten => _0NXXXXXX,7,Playback(thank-you) include => hangupcontext

Hi all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

Hi, I'm a newbie on * trying to setup an HFC card. I'm locked for many days getting the all-circuits-busy. And no idea what else to look for/how to diagnose. I'm in Spain, I've tried changing many parameters on zapata/zaptelcong with no luck, also NT & TE modes (honsetly, I've no idea what is). Any clue will be very much appreciated! I've installed *@home on my RH9,

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

Hi, I'm using the AbsoluteTimeout and Dial (with L() option) commands to set a timeout for my calls, but when the calling user transfers a call the timeout doesn't work and the call last until hanging-up. If the call is not transfered the limit works just fine. ?How can I make this work? Thanks in advance. My asterisk version is 1-0-9-07 and here's an example of one of the macros

Hi, I am trying to dial # or *411, in order to understand what the * box should answer me. In both cases, I only ear "Good-Bye" (italian , "arrivederci") dialing # -- Executing Wait("SIP/555-a2e5", "1") in new stack -- Executing AGI("SIP/555-a2e5", "directory||ext-local|lo") in new stack -- Launched AGI quitScript

Hi, I apologise if this is not the correct place to post such a message. I use Asterisk@Home package and all seems to be going well. I have identified one problem and have not managed to find anyway to fix(modify) it. We have a menu option that diverts to a mobile. If the mobile is off the network sends back a message to that effect. Now, this mobile does not have voicemail and asterisk is

Hello, This is my first asterisk installation, and having read up on the documentation, and trying several tutorials i'm unable to get my outbound route working. I'm certain it's an issue with my configuration and my inexperience with asterisk. So i have my POTS phone connected to my digium card, and when i make a call, I receive the "cannot be completed as dialed" message.

I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

Hi, I was trying to use 2 of asterisk servers and interconnected, one of them as a peer to other sever (configured in sip.conf), so all the calls to server 1 will just be passed to server 2 (has PRI Card, TE 412P, only one PRI connected), i was sending calls to server 1 and that would send to server 2 and then dial out using Dahdi, but the problem that i got was the hangup cause codes, i was not