search for: diegoquintana

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

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, 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

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 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 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 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 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'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 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