similar to: Music on Hold from Soundcard

Howdy, I've spent a couple of days writing a new feature for Asterisk that allows to record calls in T1 or E1 PRI lines using Asterisk connected to tapped lines. This means that you don't have to install anything in the PBX's/telco equipment that is going to be monitored, all you need is to install a device like the PN 633 Tap Connection Adapter that is available for example, from

Hola, paso el mini programita q estoy viendo, lo q me llama la atencion es una parte donde se definen las funciones. Probe primero meter adentro del boots la estadistica a estimar usando directamente gini(varible, pesos) pero no me dejo. Vi q en el ej del manual de boots, siempre define antes la funcion, entonces probe definir antes una funcion haciendo grini<-function(x) {gini(variable,

Has anyone had any success getting WinXP 32bit to use 4 CPU's under Centos 5.5 KVM? I have tried everything and the best I can get is 2 CPU's. I tried everything listed here http://www.linux-kvm.com/content/running-windows-smp-guests I am using the VIRTIO drivers. Is this known limitation of KVM-83-164 (Centos5 release) with WinXP 32bit? I think there are reports of it working

Hi all, Running Asterisk 1.2.12 (a bit out dated, but it was fully operational until a few days ago), I'm seeing the following message in my logs, repeated literally millions of times: channel.c: Nobody there, continuing? We've started to see some odd behavior (incoming callers can hear us, we can't hear them, we can't dial out, etc). I read that this error might possibly be

Hello, I am synthesising some sales data over a twelve month period, and then trying to use the "predict" function, firstly to generate a thirteenth month forecast with upper and lower 95% confidence limits. So far so good But what I then want to do is add the upper sales value at the 95th confidence limit to the vector of thirteen months and their respective sales to create a

Hello, >From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS.

hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . try to call them !! i'll offer you some money . You can not Call them for some advices ... It's really a bad product don't waste your time to setup it. this enterprise must

Can a temporary message in Asterisk voicemail be de-activated so that the "regular" unavailable and busy messages are played. I have several users who are stuck with the temporary message. Thanks Mark

Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm

Thanks to a generous donation by VoicePulse (http://www.voicepulse.com/) there are now a wide selection of useful (and not-so-useful) voice prompts available on my website, recorded by Allison Smith (http://www.theivrvoice.com/) There were some apps and dialplan tricks that people were waiting on release because of a lack of voice prompts; hopefully I received and remembered your requests.

I have a Linux box (Pentium III processor, 256M RAM) running kernel version 2.4.3, and smbmount version 2.0.7. The Linux box has a Windows 2000 Server share mounted via "mount -t smbfs ...". I am running into what seems to be a cacheing problem. I am doing software development on this share, editing the files from a Win98 workstation, and compiling on the Linux box. However,

Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP & NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked

Quick reminder before Astricon (from which we will be reporting from live): Tomorrow's guest will be VoIP author Alex Robar. Alex has worked with open source telephony solutions for the past four years, and has collaborated on the development and growth of an international Asterisk-based VoIP peering network. His book is FreePBX 2.5 Powerful Telephony Solutions and we'll be chatting with

Hi List, I wanted to know if anyone knew of a way with asterisk to "switch the voice" of a caller from male to female or vice versa. Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070426/2d483875/attachment-0001.htm

Hi Evgeny, I just realized that if these are compile-time errors I can help investigate on my end. Do you have something I can use to reproduce? Cheers, Graham Yiu LLVM Compiler Development IBM Toronto Software Lab Office: (905) 413-4077 C2-707/8200/Markham Email: gyiu at ca.ibm.com From: Graham Yiu/Toronto/IBM To: Evgeny Astigeevich <Evgeny.Astigeevich at arm.com> Cc:

Hi Graham, I created a bug report with a reproducer for the failures I’ve got: https://bugs.llvm.org/show_bug.cgi?id=35288 I have also found that LTO reverts everything the partial inliner has done. Maybe the partial inliner should not be used at the first LTO phase (compilation). I hope I’ll have a chance to look at the code size regressions this week. Thanks, Evgeny Astigeevich From:

Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Im having a trouble understanding the config files setup even with some documentation ive read such as the handbook, maybe im just illiterate. Anyway do you think some one can point me to some examples of real config files. Such as IAX, Extensions, and Sip. I just cant grasp the concept for some reason. If someone would like to help me out, maybe even explain one on one? Thanks a lot

Hi, I have an Asterisk 1.2.9.1 on a Debian Sarge distro connected to a VoIP provider via internet. I noticed Asterisk gets slow and behaves in strange manner if I unplug my internet cable from the PBX: for example I get incoming calls after seconds or I get no audio during calls. I thought it was something connected to DNS resolution so I put VoIP provider addresses inside /etc/hosts but

Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP