similar to: Asterisk + CDRTool

This post cannot be left without comment. People who don't know you or Adrian might get a wrong impression. I know Adrian quite well and know that he is one of the real experts in this industry and he and his stuff does not deserve such a treatment. I would recommend that you change your attitude. It seems like you did not get what you want (for free) and you complain like a small child. An

hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . try to call them !! i'll offer you some money . You can not Call them for some advices ... It's really a bad product don't waste your time to setup it. this enterprise must

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Robert, Ser, Asterisk, Mysql and Freeradius are working fine , I'm feeling tired with CDRRTool . I will use an other billing system. Thanks for your answer. Regards Harry --- Robert Zorop <rzorop@gmail.com> a ?crit : > HI, i've got a working config of ser 0.9.6, > freeradius, MySQL, and CDRTool > 4.5.3. I can't get the quotaCheck script working, i > think

Does anyone have the CDRTool working well with Asterisk? I have been trying to figure out the tricks to getting pricing to show up when searching on CDR's in the tool. I have searched on Google and the mailing archives, but have not found anything too helpful. If anyone has any insight that would be great! Thanks, dj

Hello, I read ag-projects it does'not provide support for its cdrtool pseudo gpl licensed . It's one of the bad open source project even seen . No documentation ... A real bad project !! Don't waste time with that Anybody could advise me a billing system based on cdr records with realtime support cdr_obdc.so? Regards Harry

Hello, I try to configure CDRTool + Asterisk . After having fixed a lot of files I get too many problems with this software. First I get this error why ? Error: Table 'asterisk_cdr' was not locked with LOCK TABLES The basic soap client does not connect to the soap server to get the version . harry

While researching Areski's new Web-MeetMe management gui, I found some odd (from what I expected) behaviour). Using the CLI to set un/mute status works but does not update the flags, or so it appears. Starting with a fresh conference (1 user) *CLI> meetme list 3456 User #: 1 Channel: OH323/R61 Using the CLI to mute the caller (no change in the user status0 *CLI> meetme mute 3456 1

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

Does anyone knows a working FOSS rating application for Asterisk? I tried CDRTool, but since it's free for non commercial use, it won't suit me. Also I gave a shot at Trabas (the one everybody says it sucks), but I couldn't understand how does it pull records from Asterisk's CDR table. The rate-engine addon (http://www.voip-info.org/tiki-index.php?page=Asterisk+addon+rate-engine)

25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: >> I want to change call files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!

Hi all, I have done some extensive searching, and drawn a blank so far... Nothing odd is reported in samba logs, or in the syslog file. However, if I try to play an avi straight off the samba server, on an XP client with MP10, it brings the whole deal to its knees after a few mins at the most. I have to hard reset the server. Other than this, all my other uses are flawless (game server,

Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers Steve

Does anyone know if there are any plans to update the dahdi-linux kernel module code? It no longer compiles with recent kernels, and the last release of dahdi-linux appears to have been around March of 2016. I am currently running 4.6.3-300.fc24.x86_64 (on a Fedora system obviously) and the dahdi-linux-complete-2.11.1+2.11.1 release builds and runs under this kernel, but if I try to build it under

Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far: [incoming-fax] exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM}) exten => s,2,ReceiveFAX(${FAXFILE}.tif) exten => s,3,Hangup() exten=>h,1,System(/usr/local/bin/fax2mail --cid-number "0${CALLERIDNUM}" --cid-name "home fax"

Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road,

Is there a variable for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Hi All, Which echo cancellation is good between OSLEC and MG2. Dahdi by default use MG2 echo cancellation on channel. If i want to use OSLEC then what should i need to do ? Do i need to recompile dahdi with OSLEC ? -S -------------- next part -------------- An HTML attachment was scrubbed... URL: