similar to: Areski CDR graph incorrect

Hi folks, I was wondering what happened to Areski CDR viewer that came before with Freepbx. I've noticed that the live-CD contains Areski but the repositories don't have it. Will you fix that? or shall I install Areski from sources? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676

Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd

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..

>Message: 3 >Date: Thu, 26 May 2005 23:01:28 -0300 >From: "miguel" <miguel@amplanet.com.br> >Subject: [Asterisk-Users] AreskiCC >To: <asterisk-users@lists.digium.com> >Message-ID: <20050527020659.6171A2C001F@postfix.local> >Content-Type: text/plain; charset="us-ascii" >I'm tring to dowload the AreskiCC but the >www.areski.net is

Folks! Let me clarify something to the Asterisk community about the CDR tool. 1) This is *not* my code to start with. I picked the original code from this forum here... http://www.voip-info.org/tiki-index.php?page=Asterisk%20CDR%20Areski%20GUI 2) The original code was not working (for most part, as the MySql portion has bugs) and I fixed this and added a few bells and whistles. 3) The

I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten => _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it?s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on

Dear All, Here the version 2.2 a new version of your dear CallingCard Software !!! http://www.areski.net/areskicc-doc-v2/ Many new features have been added and several enhancements made! Newest features : - A new re-build rate-engine - LCR & LCD management (OOOOHHH YESSSSS) - Billing Increment - Progressive Rate - Scheduled Rates (days of the weeks) - Expiration rates - Buy rates

Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as

Cher Areski, Je m'excuse si je me suis tromp? sur vos intentions. Cependant j'aimerais souligner que je ne suis pas le seul a avoir trouv? l'installation de votre application un peu trop compliqu?e et les instructions du "Idiots Guide" impr?cises. Il faut dire que Linux n'est dej? pas facile pour un ancien de Microsoft comme moi. Par cons?quent lorsque les instructions

Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since.... -- Chris ----- Original Message ----- From: "Chris Earle

Everybody, I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a dozen years, it was easy to configure and didn't requite installing 'connectors' on anything or adding tables on the DB server. It's based off of PHP5 and the only reason I still keep around a Debian 7 system, since it won't work with the newer PHP7. A friend of mine is learning PHP7

Hi, I'm trying to install Areskicc but with no much sucess. After installing the UI I login as root/mypass, but in the left menu I only have 3 options: Main Template, disconnect and logout. When I click on Mail Template> Show mail template it shows an error page. First, is there a way to increase the debugging level so I can know what's going on behind the scenes? Second, I've

I need decide on if our site will go with Java or Ruby on Rails. The major factor is that does Farret support Lucene''s ChineseAnalyzer or CJKAnalyzer or not. Can anyboby shine some lights on Farret''s Chinese search support? Really appreciate. -- Posted via http://www.ruby-forum.com/.

Hello, I want to analyze the asterisk logs files, looking for all kind of errors, ?Anyboby knows any asterisk logs analyzer? Thanks all, Voipcrazy

I replaced my X100P cards with two TDM04B fully populated (8 FXO modules). They are working fine, I can make and receive calls, but I noticed all modules are always in GREEN state, even if I disconnect the line. Both zttools and a cat /proc/zaptel/<device> shows no RED alarm. Is there a workaround for this? Gelson

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Hi all, Just wondering if someone have already done something like that : SIP Client_A ---> 1)call ---> ASTERISK ---> 2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference (point 1 and 2), it's definitely easy, but call

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Hi there, i called one asterisk server from another asterisk server. The calling server played back a audio data und the answering server recorded the audio sample using record() function. I tried both ISDN, VoIP connections. Camparing with the original audio data, the recorded samples from both connections were amplified by asterisk, so that the recording were much louder. But I didn't