search for: randulo2008

Many of you are interested in and have used or recommended fail2ban for your linux boxes. I finally installed it on our FreeBSD server (no asterisk, hence the OT) with the help of a friend from the VoIP Users Conference and Asterisk community. After a lot of new learning about regex, I extended the actions and filters to look at our mail server, plagued by spammers - who isn't? Our server has

Hi, I know some of you are very experienced as to the working of networks. I wondered whether there is some accepted way of determining bandwidth needs based on the network traffic over time. For example, looking at the figures for the network traffic through the server interface, we have hourly, daily and monthly figures. If everything were linear, taking the hourly figure and dividing it by

To celebrate three years of the VoIP Users Conference, we're doing a 24-hour VoIP conference call today. Details are at http://voipathon.org IRC: #vuc on Freenode.net SIP: voipathon at vuc.onsip.com - Enter 22622# and your PIN# if you have no PIN you can listen using 1# iNum - +883 51007 039 9924 PSTN: +1 724 444 7444 again, 22622#1# or PIN# if you have one. Those of you in the

Interesting offering, free from Twilio, this is php you install on your own server to build a brandable "VBX". Worth checking out! Listen to tomorrow for more about this and talk to lead engineer or Twilio CEO if you have any questions; sip:200901 at login.zipdx.com or Skype:vuc.me IRC: #vuc on Freenode.net or http://vuc.me/irc Info about VUC is htp://vuc.me Best, /r

Hi, With my Gigaset C470IP (with latest 02223 firmware), I can't find a way to access SMS settings from web configuration app or using a handset. Has someone been more successful without using auto-configuration mode ? (For instance, manual says an SMS entry is showing on handset screen but as I plugged my base station into a private LAN, I skipped the whole auto-configuration process ).

Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more "managerial" phones than the base phones which will be used for one line only. TIA Julian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Alistair Cunningham of Integrics was our guest yesterday. We talked about Integrics new product Geons, a suite of software for building large-scale distributed enterprise applications. The recorded session is now available here: http://www.voipusersconference.org/2010/geons/ The extremely rare John Todd was sighted (and heard) at this event. If you are developing a product or service

Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get "Push message cannot be displayed" back from the Polycom phone, and all I am sending is the Polycom example : <PolycomIPPhone> <Data priority=?critical?> <h1> Fire Drill at 2pm </h1>

Hi, I'm getting some strange behaviour on Asterisk 1.4 running on Debian Stable (Lenny). I suspect it's something to do with my setup, rather than a bug, but I'm struggling to see it, and would appreciate any input. Setup: PC with two ethernet cards: eth0 goes to local network, including two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes to router and

Hi, I'm looking to build an Asterisk box that can run at a remote location. Here are most of the specs of what I'm looking for: Physical hardware * Small pre-built PC (not buying board, case, all parts separately) * Low power consumption * No fan or very small fan * Hard drive (not flash memory) Capabilities/capacity * No GUI, no X * Register to multiple SIP

Hello All - I've been poking around the past few weeks, trying to familiarize myself with all of this. I am new to Linux, VoIP and Asterisk -- to be complete. This is my first exposure to all of these technologies. I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge 2400 and the install went well. I can log in and poke around in Linux and I even configured the box to be

Looking at my shiny new Google Wave account, I was wondering if anyone else on this list is in the beta AND going to Astricon. Astricon seems like it would be a good test of the kind of collaboration GW is trying for. In any case, I'd love to try to do an Astricon wave so let me know if you're interested and we'll get together. I know at least two other people who'll be there

If you missed @voicegal last time or didn't go to Astricon, join us today on the Voip Users Conference to meet Allison Smith, the voice of Asterisk. Or go listen to the FBI talk about security... http://VoipUsersConference.org for details. /r

Hello, I am working with several SIP projects that use g722, or are trying to do so, with pjsip library. According to pjsip team's interpretation of g722, it works with 14bits PCM for input/output, so pjsip basically 'converts' the audio sample from 16 bits to 14 when encoding and vice-versa. Some implementations don't do 16<->14 bits conversion, so when pjmedia talks to

Hi, We had a last-minute cancellation from Vivox for today's conference. It happens that someone suggested a guest idea, Howler Technologies CTO Jay Fenton, who agreed to join the call from the road. Anything you want to know about transcoding to and from g729 is out topic for the first hour. My pal David Duffet knows this technology well and has kindly signed in to help guide us through this

http://vuc.me Kamailio, Open SER and Asterisk walk into a bar... The bartender is Alex Balashov, someone whose posts I have long admired on this list. Alex has agreed to take us through the following areas: - Relationship of Kamailio to OpenSER project history. - What is Kamailio/OpenSER? - SIP proxy - SIP server (for certain purposes, such as registrar, presence user agent, etc.) -

Hi, Curious, do many of you check out software or projects when they have a live CD or does that make any difference to you? Does anyone know if the general public (not reading this kind of list) is attracted to a Live CD more than an Install one? thx, /r

Thanks to Digium, the company, and to all of the fine people from Digium who participate in the weekly VoIP Users Conference conference! We will be live on Friday January 1, 2010 and there is also a "reel" of recorded greetings from people around the world wishing the VoIP Community a Happy New Year. You can hear this anytime during the year by downloading it from the site starting next

Hello, In about one hour we should be chatting with Tim Behrins of Voxbone about their initiative, iNum. I say "should" because he's the scheduled guest, but I haven't heard from him today :) Next week, we'll be "Hacking VoIP" Feel free to top post your answers, it seems to stimulate conversation. /r http://VoipUsersConference.org for the usual data or jump on

Hi, Our guest this Friday is Himanshu Dwivendi, author of the book Hacking VoIP. You're welcome to come discuss it with us on the conference. Find your local time by going to http://vuc.me/next - the conference begins a little before 12 Noon Eastern Time. VUC has an IRC channel #vuc on Freenode.net and uses Skype for Asterisk to allow connections to listen and/or speak. SIP g722 : 200901 at