similar to: CDR after SIP blind transfer.

Hi I'm having 2 strange problems (or rather I am doing it wrongly!) ..need some help.I am working on trixbox 2.6 with Asterisk 1.4.22 1. I've a Queue with dynamic agents (ampextensions=deviceanduser mode in amportal.conf), some of agents when they punch in the wrong UserID - they still get logged onto the Queue as a result many of them complain that they don't receive calls on

Greetings list, Whilst running through a routine check of some CDRs, I've noticed that the originating channel's accountcode isn't preserved on creating a local channel. For example, if we start with: exten => 123,1,Set(CDR(accountcode)=foo) exten => 123,n,Queue(bar,nrtw,,,) And the queue 'bar' is defined as follows: [bar] member => Local/456 at outbound member

Hi, as far as I know my problem is not a bug but wanted behaviour. Let's assume the following dialplan: exten => 123,1,Answer exten => 123,n,Dial(DAHDI/g0/00492112233,20,g) [...] exten => 123,n,Hangup I do an dial-in with my SIP-Client (or phone). The Dial-Application starts the outdial and I get connected with my partner. After hangup, only 1 CDR is generated. But I(!) need 2

I have created an asterisk.org blog entry: http://www.asterisk.org/node/48358 to describe what I will shortly be committing to trunk to correct the weaknesses of CDRs, that asterisk users and developers have been complaining about for quite some time. Highlights: Restructuring the code and philosophy of CDRs. Plans to eliminate the ForkCDR() application Plans to create

Hello, Sorry for a bit of a newbie post but we all had to start somewhere right .. I'm wondering if someone can briefly explain the difference between blind and attended transfers and why they would generate two very different CDR entries.? From my own research, it seems that transfers are both ultimately a SIP REFER and thus seeing two different CDR entries just confuses me further.

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

Hello, I have an application that needs to record outgoing calls. It's running on Asterisk 1.4.18, with CDR data stored in MySQL. Outgoing calls are recorded based on their uniqueid. When outgoing calls are placed, there is a line like this on my extensions.conf: exten => _.,n,MixMonitor(/var/spool/asterisk/monitor/${UNIQUEID}.gsm) For regular outgoing calls, this works fine. The

Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to

Trying to figure out if a bug report should be submitted. Can anyone on 1.2.x verify of this has been corrected? I am on CVS 8/2005 If a call comes in to an extension that dials more than one channel (rings at more than one phone) both calls in the CDR show a status of answered when only one is answered, the source channel is bridged to only one of the two destination channels, but both CDRs

When you use a product which version number is 11 or even 12, you might go with the assumption all big bugs are fixed and then you find there is a huge, important, expensive bug still running in the code we are relaying upon... The problem is simple. If you transfer a call, that dialing will be not reported in the CDR, so no billing will happen. This is a simple example: Extension 100 calls

Hi All, I'm having some troubles with using call files. I'm trying to establish the following: - want to use call files to connect two (outside) extensions - want to use the outbound routes set in FreePBX - want to set the outgoing callerid for both calls - want to set a custom CDR field in MySQL ( field name 'azonosito' ) Asterisk is version 1.8.2.3 with freepbx 2.8.1. What

Hi Gang I have not yet managed to find a solution to correctly generate CDRs for this situation: Alice calls Bob. Bob has call forwarding delayed 20s to Charlie. Charlie picks up immediately. exten => bob,1,DBget(cfwdly=CFDLY/${exten}); $cfwdly contains charlie same => n,Set(CDR(src)=${CALLERID(number)}) ; src 'alice' same => n,Set(CDR(dst)=${exten}) same =>

Hi, I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything is under control, I have one issue with the way CDRs are kept for queues. And I don`t mean ?I don`t like it?. I mean it crashes the server. I realize there are multiple CDRs per queue call ? one per ring/per phone, basically. The issue is that whenever the number of CDRs ?to be recorded? for a call exceeds 5000,

Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten => 123,1,PlayTones(ring) exten => 123,n,Wait(5) exten => 123,n,Playback(demo-congrats) exten => 123,n,Hangup() exten => 321,1,Dail(LOCAL/123 at test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice

On Sat, 2009-02-07 at 15:51 -0500, Alexander Lopez wrote: > > > -----Original Message----- > > From: Steve Murphy [mailto:murf at digium.com] > > Sent: Saturday, February 07, 2009 1:59 PM > > To: Alexander Lopez > > Subject: RE: [asterisk-dev] 1.4 and CDRs -- The Breaking Point > > > > On Fri, 2009-02-06 at 12:28 -0500, Alexander Lopez wrote: >

Hello! Most are probably bored seeing another letter about this, but I've put in a fair amount work on a spec for rewriting the CDR system in Asterisk, and I have some questions: First, please look at what I've written so far: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the file "CDRfix2.rfc.txt" in the RFCs dir. The spec SIGNIFICANTLY alters the way

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hello! I have been doing some tests with call transfers and I have been looking at the CDRs that Asterisk generates. Scenario 1: A calls B B answers and does a blind transfer to C (using # key) C answers and talks with A Scenario 2: A calls B B answers and does an attended transfer do C (using the phone's transfer key) C answers, B hangs up, and C talks with A For scenario 1, the CDR shows

Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page..