similar to: CDR with unix time.

> > Sorry, of course cdr.conf not queues.conf. marcus > > Am 01.12.2010 19:16 schrieb "marcus rothe" <synco16 at googlemail.com>: > > > Hi Rodrigo, have you got enabled the appropriate line in queues. Conf? > Regards Marcus > > Thanks very much, I include the line "unansweredy=yes" in the cdr.conf and solve the problem. Thanks again! --

Hi to all! I'm trying to create a context for integration with extensions.lua and libsql.mysql, but I'm not getting to run. When I reload the module pbx_lua.so the following error appears: [Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua extension: error loading module 'luasql.mysql' from file '/usr/lib/lua/5.1/luasql/mysql.so':

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

Hi everyone Anyone please explain me How Account code is use for billing., Thanks Nikhil

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

Hi, In my application I want to have the sysdate + 5 minutes. I know that the sysdate is in the variable ${DATTIME} But now I want to now how I get the sysdate + 5 minutes into a variable? Doe's anybody knows the answer? Kind Regards Arjan Kroon -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good afternoon list. I am facing a problem with the CDR and Queue_log tables (MySQL). The ABANDON events is being saved correctly in queue_log, but in the table CDR is not saving the registry of such abandoned calls. Apparently the CDR table is functioning normally, I have several records of links in it. From what I noticed, is only the events abandonment that are malfunctioning. With this

Hello, We have finally upgraded to Asterisk 1.4, however we've run into two issues that weren't occurring before the upgrade. Issue #1: We're an outgoing call center and need to record all calls. We use the uniqueid field in the CDR to match with the recording, which we labeled with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct in the CDR. Here is the

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

Hi All; I used A2Billing, basically it is nice and fine, but management possibilities is not that rich, so a lot of staff are need to be repeated that let the admin facing a problem of the needed time to do the task. Anyone advise for another open source prepaid billing that is rich by the management features? Also, I hope to find an open source Billing (prepaid and postpaid) that can work with

Hi, does anyone know how to plot 3D parametric surface,e.g ellipsoid: x=a*cos(s)*cos(t), y=b*sin(s)*cos(t), z=sin(t); s in [0,2*pi], t in [-pi,pi]. Vytautas Maniusis, Vilnius University, Lithuania

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch Thanks

Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunking. Cateris paribus, what is the preferred solution to choose? What points to consider? I can name the provider if this is not against this list policy--is it? Thanks, -kkm

Hello, We would like to share our webinar about Auto-Dialer Business Model (Telemarketing). It is educational video which we made for our clients and now we are sharing it with you. http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/ Enjoy. NOTE: This is not attempt to sell you anything. No product or service is sold/marketed in the video. Regards, Kolmisoft Team

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still

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

I have sent an e-mail to this list (awaiting moderator approval by the size) talking about some difficult to make calls with a SIP Provider in Brazil. I'm new at this list and have no sure if I have posted my question in the right place. If this is not the channel to make this kind of question about this issue, I'm sorry but want to ask if anyone can indicate the correct place . Thanks