similar to: the correct way to stop a CDR?

Hi All, I'm using asterisk 1.2.15 and call forwarding doesnt work for me. from my extensions.conf: ; Unconditional Call Forward exten => _*21*X.,1,NoCDR exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) exten => _*21*X.,3,Playback(vm-saved) exten => _*21*X.,4,Hangup exten => #21#,1,NoCDR exten => #21#,2,DBdel(CFIM/${CALLERID(NUM)}) exten =>

What would it do if you exten => h,1,ResetCDR(w) exten => h,2,NoCDR() exten => h,3,DEADAGI(get-unqiueid.php) I have not tried it but in theory it should write the first CDR and then kill the write of the second NO ANSWER CDR. Let me know if it works for you as I may need to do it on some of my h exten code as well. Bryant ---------------------------------------- From:

Hi, When I use NoCDR application I obtain this warning in console log: Mar 1 11:16:08 WARNING[3513]: cdr.c:114 ast_cdr_free: CDR on channel 'SIP/492-7371' not posted Mar 1 11:16:08 WARNING[3513]: cdr.c:116 ast_cdr_free: CDR on channel 'SIP/492-7371' lacks end Can someone explain to me what is due? Thanks.

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

I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though

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: >---

I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From

There is an environment Setup uses Asterisk 1.2 and doesn't want to upgrade. There is an issue while a call goes to any queue we create, the call is being disconnected after 20 seconds and it is hangup. The following is the configuration: - vi /etc/asterisk/queues_additional.conf [8] wrapuptime=0 timeout=30 strategy=ringall servicelevel=5 retry=4 reportholdtime=No queue-youarenext=

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

Hi, We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity recorded in our full log. Could you help us to explain what had happened. Thanks. === my friend, 801, from his room did a test by dialing echo test in freepbx, *43: [Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing [*43 at from-internal:1] Answer("SIP/801-000003f5", "") in new stack [Aug 20

Hello, I am having an issue whereby calls are being dropped randomly. I have an ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk install is based on Trixbox 2.0. However, I have updated the source code to the following. The Asterisk release is asterisk-1.2.20. Zaptel release is zaptel-1.2.18. And libpri release is libpri-1.2.4. I have include an extract from the Asterisk log