similar to: Asterisk's DANGEROUS Transfer CDR's

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

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

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! 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

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as

Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

Greyman-- I'm taking this discussion to the list. Folks, what we are talking about here, is me trying to get a grasp around Greyman's (Aaron's) request for a bare-bones CDR generation that describes just total connect time for channels, stripping out all the details. Who cares about xfer, park, hold, etc.? So in the following is our discussion about what *should* be there, and in

Can any explain the following error: Error in if ((seedCount <= seedNumber) && (valueDiff > sup)) { : missing value where TRUE/FALSE needed which I get upon running this script: seedNumber <- 10 seeds <- array(dim = seedNumber) seedCount <- 1 maxValue <- 100 sup <- maxValue / 2 fcsPar <- array(as.integer(rnorm(100, 50, 10))) while (seedCount <=

Can any explain the following error: Error in if ((seedCount <= seedNumber) && (valueDiff > sup)) { : missing value where TRUE/FALSE needed which I get upon running this script: seedNumber <- 10 seeds <- array(dim = seedNumber) seedCount <- 1 maxValue <- 100 sup <- maxValue / 2 fcsPar <- array(as.integer(rnorm(100, 50, 10))) while (seedCount <=

Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use

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

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip

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 =>

Hi there, How do i setup asterisk, so that in the CDR's is only the time, which the line actually was connected? Not the time, the line was up, but the time the user was able to talk to another user. Thanks in advance, Carsten

Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent.

Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks!

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi guy's I've got an existing application which takes CDR information from a com port from an avaya switch, that we use for billing and accounting. Since the cost up upgrading the Index platform is extortionate, we're looking into alternatives and I've been told to come look at asterisk. First off, asterisk is excellent...

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