similar to: VoIP Users Conference Recordings

Hi, I met Michael Iedema of the Askozia pbx project at ANOOMA (ANOOMA is the Cannes Festival of the Asterisk world) and begged him to take time off from porting Askozia from FreeBSD to linux to tell us about his project and the problems of porting as well. Hope you'll all be able to stop by and lmisten or download or stream the recordings. We are having a lot of people call in on widenabd

On Fri, Feb 13, 2015 at 9:52 PM, Ilia Mirkin <imirkin at alum.mit.edu> wrote: > On Fri, Feb 13, 2015 at 8:59 PM, Andy Lutomirski <luto at amacapital.net> wrote: >> My new GTX 960 (GM206, I think) says: >> >> [ 10.329351] nouveau ![ DEVICE][0000:09:00.0] unknown Maxwell chipset >> [ 10.329355] nouveau E[ DEVICE][0000:09:00.0] unknown chipset, 0x126010a1

On this weeks VUC call we will collectively be our own guests. That is, we'd like to know what was the big issue that impacted YOU in 2010? All opinions welcome. Here are a few things to get you thinking in advance: - Apple's Antenna-gate - Asterisk 1.8 Launches - Amazon EC2 as a DOS platform - Cisco launched UMI video conference device - More HDVoice capable phones - Skype Outage - VoIP

Dear all, I'm writing about setup vdpau firmware with the nouveau driver according to this page: http://nouveau.freedesktop.org/wiki/NVC0_Firmware I have installed the nvidia-313 binary drver to attempt to extract the firmware. I have obtained another mmiotrace using my default kernel I have installed on my Chakra Distrobution. I have used the instructions on

Hi all, You're welcome as always to join the talk on the VoIP Users Conference, VUC for short. VUC began as the "Asterisk Users Conference" but for obvious reasons, we changed the name in the first year, although Digium was our sponsor for three years. We still have plenty of you who are asterisk users and developers on this our fifth year. Come on by, listen, talk ro text on IRC.

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, We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that "Trapezoidal VoIP is Evil". Join us and ask questions, make comments, argue about geeky details... and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an additional handset. Those of us who have these phones like them a lot. All dial in info is here: http://VUC.me -

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

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

The AstLinux Team would like to announce that the 0.7.0 version of AstLinux is available for download. There have been many significant updates in this release including updating to the latest Asterisk Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other system updates. For a complete list of changes, read the changelog available on the download page:

Create the ref_bo without any storage type flags set for now. The issue probably arises from our use of the additional buffer space at the end of the ref_bo. It should probably be split up in the future. Signed-off-by: Ilia Mirkin <imirkin at alum.mit.edu> Tested-by: Martin Peres <martin.peres at labri.fr> Cc: "10.0" <mesa-stable at lists.freedesktop.org> ---

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I remember someone wrote a great document concerning Polycom server provisioning that provided a way to ensure that updates to the firmware did not overwrite customizations. I'll be damned if I can remember where I saw it. It may have been discussed during a VUC session or may have been on this list. Either way, I'm unable to google my way

Several regulars from the VUC will be there, some of us are arriving Tuesday night. Anyone else considering the trip? Post here or contact me off list so we can meet. /r

Integrics is pleased to announce the release of ITSP version 1.6. This version has the following new features: - Comes in 2 editions: * Carrier edition, for 250 to tens of thousands of users on hosted systems. Integrics sells this edition directly and through partners. * Office edition, for 10 to 250 users. This edition is sold only through our partners, for them to sell as PBX systems at

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

Integrics is pleased to announce version 2.0 of Enswitch, the most integrated platform available for offering commercial telephony services such as ITSP, hosted PBX, calling cards, call shops, number translation services, and much more. Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is effectively the same product as ITSP 1.7. The product has been rebranded as, although it

This is a request to the website manager for asterisk.org. The build scripts for our ITSP product include the URLs to download the Asterisk files, such as: wget "http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz" However, if a new version is released, asterisk-1.2.5.tar.gz is moved to the "old" directory. This breaks our scripts until we can update them and send

If you've ever wondered what Karanbir Singh looks like, check out this week's (142) video podcast at http://twit.tv/floss. I'm only partway through the audio version (will finish on the drive home) and haven't heard anything that would be surprising to people already using Centos yet but it's a great presentation and something to recommend to people who aren't

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of