similar to: Disable Console Audio

Hi, I'm now setting up a VoIP conference room using Asterisk. All the clients are SIP phone (to be exact, Xlite), number of clients that should be registered are around 50 and concurrent users are maybe 15 clients at most. So, basically I think I can handle the situation only with Asterisk. I'm wondering however, most of my clients are behind NAT of home router and using SER together

Dear Asterisk Users, I'm a Japanese and now configuring Voicemail. Now I need to modify the way cmd VoicemailMain works to fix language difference and other my conveniences. What I want to do are... 1) Add words used in message retrieving guidance. I need to add different suffixes to numeric words due to Japanese way of mentioning time. (e.g. in English, you can say "Five

Hi, I'm a beginner of voip and just wondering the possibilities of *. Is that possible for * to handle jabber based voice chat IMs, possibly inter-connecting them to different kind of clients -say, H.323 clients- in meetme conference function? If I use SER together with *, is that possible? -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter

Hi, I'm thinking of introducing Asterisk on Linux for IP PBX. Now I'm using ISP that has VoIP service and I have VoIP terminal box for that ISP and a SIP account for SIP server of the ISP. Now, what I would like to do is the following. A. Setup IP PBX on Linux by using Asterisk. B. For IP-PSTN gateway, configure Asterisk to use my ISP's SIP account and connect to my ISP's IP

Hi, I'm thinking of setting up Asterisk for H.323 video phone clients. Now, what is the difference between native H.323 that come with Asterisk and "Open H.323 for Asterisk" ? TIA Kuni -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter mailto:kuni@ej-interpreter.net Macintosh Webcast Specialist

Hi, I'm going to setup Asterisk on my server which have 5 IPs (3 global and 2 local). Now I want to bind Asterisk to 2 IPs (1 blobal and 1 local) Is this possible on config? -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter mailto:kuni@ej-interpreter.net Macintosh Webcast Specialist http://www.macwebcaster.com

Hi, Is there any future possibility that Asterisk will be compatible with connection to 3G video mobile phone such as Nokia 7600, Nokia 6630 and many ohters in Japan, Europe and HongKong? If this become possible, H.323 video clients and 3G mobile phone will be able to share video conversation, which will be huge in those countries. In Japan, more than 3 million 3G video mobile phones are

Hi, I know the following is mostly the issue of SER and I already posted the same content to SER User list. Just for more input, I posted it to this list. Sorry for the cross post for some people. I've set up SER for UA to UA call. I'm thinking of setting up SER to relay to Asterisk PBX to use conference call and voicemail of Asterisk. I will employ this system for client connection

The first beta of Asterisk 1.2.0 has been released! It is available from the ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0-beta1' tag). This version of Asterisk represents a significant improvement in features, stability and compatibility over the 1.0.x releases. Some of the major new (or upgraded) features include: * Asterisk Realtime Architecture

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per

I have some data I would like to model which involves choice of food by dung beetles. There are a number of experiments where in each case, there are five choices. Overall there are more than 5 different foods being compared (including a placebo) and different experiments use different comparisons. The problem is a generalization of Bradley-Terry but it differs from some generalizations in

I'm trying to set up the console as an extension (so I can set up overhead paging), but I can't seem to get it to work. When I call my paging extension, I get an error that it can't open the device: -- Executing Ringing("Zap/9-1", "") in new stack -- Executing Dial("Zap/9-1", "Console/dsp0|18|A(new/whistle)") in new stack << Call

Hi, I'm new to *. I just installed my X101P card with * from the source on Mandrake 10.0 and I test it. Everything seems to work fine. When I call at my home office all the demo ivr seem to work. But I have one question regarding * using /dev/dsp0. I only have one sound card on my system and it has to be use by my personnal PVR called MythTV. I though that * did not need a sound card to work.

Hi! Please help me... I'm really confused.. I have read almost all the books out there but its not working.. I just want my Windows PC to be able to access UNIX PC with a username and password authentication. Below is my smb.conf file.. [global] workgroup = MyWorkgroup netbios name = board_pc server string = %h server (samba %v) log level = 10 syslog = 0 log file =

I use Wine for play World of Warcraft, that support on Windows, 6 channel sounds mixing software. On Ubuntu 8.04 and Wine i get only stereo output. I've tried to do this .asoundrc for "force" use on Wine, of 6channels and I don't get any sound. Code: pcm.!default { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0660 # IPC permissions

Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: cat /proc/assound/cards 0 [V8235 ]: VIA 8233 - VIA 8235 VIA 8235 at )xe400, irq 11 Now I installed the bristuff+asterisk 1.0.9 and always have in my logs

I am using alsa with asterisk. The asound.conf is below. When I start asterisk with /etc/asound.conf present I get errors on the console that: chan_alsa.c:304 alsa_card_init: snd_pcm_open failed: Invalid argument If I remove the asound.conf asterisk starts up and works. However I NEED the asound.conf for another application. What might be the issue here? THanks jerry /etc/asound.conf

Hi I had have trouble to get sound working under freebsd 8.1 amd64 arch. So i decide to dig in wine code and I create a simple "proof of concept" patch to get work my sound card.? ********* SND STAT ********FreeBSD Audio Driver (newpcm: 64bit 2009061500/amd64) Installed devices: pcm0: <HDA Realtek ALC272 PCM #0 Analog> (play/rec) default <- This is my primary snd pcm1: <HDA

I found this information elsewhere.... making it /root/.asoundrc helped. # This is .asoundrc # # this makes legacy OSS apps use alsa software mixing dmix pcm.dsp0 { type plug slave.pcm "dmix" } # mixer0 can stay unchanged, because it isn't used anyway, I guess ? ;) ctl.mixer0 { type hw card 0 } # this makes native ALSA apps default to using dmix pcm.!default {

Hello, On July 22th, 2010, Typo version 5.5 named for famous photographer Richard Avedon was released to the public. Coming with a new admin and setup, Typo 5.5 runs on Rails 2.3.8. Being around since March 2005, Typo is (probably) the oldest blogging platform in Rails. It has a full set of feature you would expect from such an engine, powerful SEO capabilities and full themes and plugin