similar to: ForkCdr in 1.4.*

Hello, CDR fans! I'm looking at some issues brought forward over time: 12726/10668: someone wants me to revert the changes I made via bug 10668, last Sept; (that's they are messing him up. And I didn't do the change suggested in ForkCDR, for fear of lousing up folks depending on current behavior. Which probably sparked: 11721 :

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

Hi, I'm tryng to place a call through Asterisk Manager Originate Action. Since I want separate CDR for each of the two legs of the call, I'm forking CDR with ForkCDR as the first Channel has picked up. The problem is that, while the first CDR is fine, in the second one the "answer" field is always empty, "billsec" field is 0 and "disposition" field is

Dear members of the list; I am writing in the hope to get some help with a very peculiar problem with my new asterisk 1.6.0.1 installation. The same code runs on version 1.2 without problems, but it seems the behaviour has changed (also on 1.4.7, which I tried). Please consider the following extension: exten => 1213,1,Answer exten => 1213,n,Set(counter=X) exten =>

Gang, I'm having this error pop up when I do a ForkCDR, and I'm not sure how to get around it. Here are a few log lines: Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing ForkCDR("Zap/49-1", "") in new stack Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a CDR The scenario occurs like this: I use a .call file to generate a call on

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.

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, In my 1.6.1.18-powered system, I've got the following dialplan (in extensions.ael) : Dial(SIP/foo,15); if (${DIALSTATUS}="NOANSWER") Dial(SIP/bar,15); When SIP/baz dials peer SIP/foo which do not answer, I've got a single CDR entry like this: SIP/baz SIP/bar time_when_foo_started_to_ring time_when_bar_ended_talking ANSWERED How can I get two CDR entries :

Hi Guys, The ForkCDR is doing pretty well, except for the call disposition. The time stamps are great, but the disposition always reads answered, even when the call is busy or not answered at all. It should read BUSY and NO ANSWER. I can send a snippet of the CDR record with forked calls for an example if needed. I think the record is not forking properly, but is maintaining the disposition

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

Hi all, i am wondering if i can make two leg cdr in mysql cdr table. 1st Leg : Registrar the ATA which registered to the asterisk and it normally logging in cdr table. 2nd Leg : The CDR of carrier for the example if i send call like exten => _x.,1,Dial(SIP/${EXTEN}@AT&TIP) I this cause i can get the accrue duration of call because currently we are facing some call missing not coming

Hi, I am having problem with my Asterisk installation and find out it has to do with ztdummy. if the ztdummy module is loaded, the asterisk playback() command will not play files. DTMF is still properly received. If the ztdummy module is unloaded, sound playback works again. Here is my version zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I

Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it

Hi, I had Asterisk threads consuming 100% CPU at times since last week. Of course, last week an extra card was installed (we had a 1PRI, a 4PRI was added) so search concentrated on that, but to no avail. Today, I installed DDD on the machine and quickly found out that it was looping because cdr->next->next == cdr in ast_cdr_setapp(). I patched this up with some simple code in

Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

Hi, Using 1.6.2.13. I'd like to know how I can force Asterisk to fork a call. To simplify things, Let's say I have an out context (for outbound calls) and an in (for inbound). If person A wants to call person B, and both are on my servers, I don`t want to send the call out. I want all this to happen internally on my server. The problem is if I use some condition to send

Dear all! W2k, R 2.5.1 I am working with an ongoing malting barley variety evaluation within Sweden. The structure is 25 cultivars tested each year at four sites, in field trials with three replicates and 'lattice' structure (the replicates are divided into five sub blocks in a structured way). As we are normally keeping around 15 varieties from each year to the next, and take in 10 new

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more