similar to: Areski CDR

Hi folks, I was wondering if there's a guide on how to configure sugarCRM Integration with Asterisk. I was looking in google and all i found was about Trixbox, which has sugarcrm integrated by default. Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011

Hi all, I've just realized that my asterisk isn't storing cdr inputs in mysql. cdr_mysql.conf is well configured and I don't know what else should i configure. I'm using Xorcom's packages, "cdr status" shows: voip*CLI> cdr status CDR logging: enabled CDR mode: simple CDR registered backend: csv CDR registered backend: cdr_manager CDR registered backend:

Hi all, I've just installed again my Asterisk using Xorcom repositories. I can make extensions, but when using any extension i want to dial anything, I got "404 not found" using Xlite. Any ideas of what can be happening? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext.

Hi all, I want to make a SIP trunk between a Cisco 2811 router and a Asterisk. Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk 2XX). Now I want to configure a trunk so that 2811 users can call * users. I've been reading a lot but I'm still confused. Hope you can help me, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User

Hi all, I was wondering if there is any IAX2 sniffer and decoder. Wireshark can decode and play RTP streams using G.711, and Cain & Abel decodes and plays any kind of RTP stream. But I didn't find anyone that can decode IAX2 streams. Any programs?? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP #

Hi everybody, I've installed "future" packages (asterisk 1.2 and freepbx) from Xorcom's Repository in a debian etch, but when i want to uninstall freepbx-panel, i got this error: dialer:~# apt-get remove --purge freepbx-panel Leyendo lista de paquetes... Hecho Creando ??rbol de dependencias... Hecho Los siguientes paquetes se ELIMINAR?N: freepbx-panel* 0 actualizados, 0 se

Hi all, I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything seems nice, but i'm not able to make calls nor to receive any. When I try to make a call, I keep receiven the "all circuits are busy now" message, and when I receive calls, asterisk doesn't seems to care (don't get anything on the CLI) I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's

I have two seperate systems at two different locations. Each hosts there own voicemail for their phones. I have thought about just having all voicemail on one server. Is the best way to do this just through a dial app? For example, if someone dials 1000 to check voicemail at site A. The dialplan will be something like this on Site A: [context-for-phones-at-one-location] exten =>

Hi, I am developing a VoIP client, and I want to use speex as the encoder. I'm using python as the programming language for the project. However, when I try to decode frames recorded from the microphone and then play them, I got an error, which says that I should give a string as an argument in order to play, not a list (speex.decode returns a list) Any idea how to fix this?? Greetings from

>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

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

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

Is anyboby using the areski CDR reporting tool? I have installed asterisk-stats v1.2 three days ago, but I found a possible bug in it. My calls compare graphic shows most of the on the calls at first hours past midnight, and it never logs anything after lunch time. This is wrong, my calls are made on business hours. The call log lists those calls at the right time. Is there

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

Hi all, I am trying to install Vicidial in an existent FreePBX installation (I'm using Xorcom packages for Debian Etch), but I didn't find any documentation, I found only this guide [0], but is for trixbox only, do you think it will work on FreePBX on Etch? [0] http://iptn.org/vicidial/index.html Regards, Diego Quintana Cruz

Resending again, now that I'm a new member - AQ ________________________________ From: Quintana, Andre (M Tech Ops) Sent: Tuesday, June 22, 2010 4:23 PM To: rsync at samba.org Cc: Quintana, Andre (M Tech Ops) Subject: Installing rsync-2.4.6 on an Intel box running Solaris x86.... Importance: High Greetings all RSYNC Guru's, We are presently using rsync-2.4.6 on our

Hi I have gone thru the steps of installing AreskiCC, I would like to know how to get access to the GUI interface of this application. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051011/db024f52/attachment.htm

I like to think I am not a complete idiot... ...I have googled till my fingers bled. I cannot figure out how to install these apps. I have figured out the database protion as well as editing defines.php but the web portion is killing me. I am running apache and have done no configuration to that. Its a fedora core 3 box with the latest, within the last week, asterisk and zaptel packages. I

Hello I got the similar error while trying a call. -- Executing Answer("SIP/8000104-86ef", "") in new stack -- Executing Wait("SIP/8000104-86ef", "2") in new stack -- Executing AGI("SIP/8000104-86ef", "areskicc.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php areskicc.php:

Cool! I'm still away from the office, but I was starting to work towards syching meetme2 up to the version of meetme in * 1.0.7. It is over a 2000 line diff, ignoring the database integration code, so it was looking like a not too trivial task. One question though, how difficult will it be to extend your latest version with the scheduling features I built on top of you previous version? I