similar to: CDR, RingTo Number, and DST

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.

Hi, Would just like to ask about cdr, i have an asterisk and i would like to bill only outbound calls not extension to extension, when i'm looking at the CDR, i can't figure out which fields i need to filter all outbound calls only. e.g if i dial 00. or 9XXXXXX (for local pstn calls) those are billable, 100 101 or 102 (all local extensions) not billable. *97 for voicemail not billable,

When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hi asterisk-users, We have recently moved to the 13.x branch of Asterisk from 11.x, and we're trying to correlate CDR records from multiple-legs for billing purposes. As part of this change we have added 'linkedid' to our CDR table schema in an attempt to join the multiple records into one billable record. The call path can be simplified as (transport types in brackets): SIP

Hello, I was looking into cdr (w/ mysql) before enabling dialout through CAPI, found out that transfered call are not reported as such. For instance, let's say we have 2 local SIP extensions A and B and C is a GSM number. If A calls B and B transfers to C (automatic or after picking up) the CDR shows A calling C. IMHO, It should record A called B then B transfered to C, so that the cost of

Is there anyway to get CDR to show just the answered calls. Not by exporting to a spreadsheet and editing. We have ring groups and queues and CDR shows everything as calls received. Even if it's multiple extensions ringing it shows them as multiple calls received. This seems kind of goofy.

Is the CDR table the right table for billing? I did some tests and CDR records billing seconds for calls that where never picked up. Is this a bug in my system or is that the way CDR works? I called out on my X100T card. Best regards, Han Test data Duration 12 seconds 8 seconds billing time (never picked up my phone) Duration 111 seconds 108 seconds billing time (5 second but

I am in search of open-source software to track billable time for myself. A quick Google search did not find anything that is open-source but I am hoping that this group might know of such software? The platform is C7.

Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I make a bridge call (using .call files in outgoing/) I always get 'billsec=12' in the cdr, both mysql and Master file even if the call lasted longer, watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked

Here is the setup: SIP/3044 -> SetCallerID(5551212) -> Call out PRI The CDR shows a src of 5551212. That is a lie! The src of that call was not 5551212, the source was 3044! The "translated source" of that call was 5551212. How can I get "real" source of this call and not some faky nonsense? The "reason" behind using the SetCallerID is because if I

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 =>

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'm currently working on a perl script convert csv logs to a http log equiv: LogFormat "%h %l %u %t \"%r\" %>s %b \"%{Referer}i\" \"%{User-agent}i\"" Right now I have output something similar to this: 111 - - [02/Aug/2003:16:39:15 -0500] "GET /300 HTTP/1.0" 200 6144 "sipext" "ANSWERED" INCOMING - -

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

I have a controller that gets a list of employees (which has an "include => [:salaries, :incentives, :billablegoals, :reviews]"). I then need it to iterate through each employee and determine their current active goal based on the "effective date." After playing around with it a bunch, I got the following to work. Only problem is that if I remove the "@employees.each {

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

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

On Sun, Feb 26, 2023 at 2:51?PM Thorsten Glaser <t.glaser at tarent.de> wrote: > > On Fri, 24 Feb 2023, Keine Eile wrote: > > > does any one of you have a best practice on renewing ssh host keys on cloned > > machines? > > Yes: not cloning machines. Good luck with *that*. Building VM's from media is a far, far too lengthy process for production deployment,

Hi list, I have a little problem that I cannot solve myself, being new to Asterisk. Maybe someone can help me out. I'm executing a Dial command in an AGI script launched from the 's' extension. It works great, but the CDR shows 's' as the destination. I need a CDR record with the number passed to the dial command as dst in the CDR. I'm sure there is a proper way to handle