similar to: Monitor and Record : audio quality

I have the same problem in a box with 2 HFC-PCI, but i already remove the row in modprobe.conf and load the module manually. Both cards works fine Any idea ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Julian J. M. Inviato: luned? 27 giugno 2005 12.04 A:

Hi, Bristuff works great with HFC card... your compilation problem may come from your kernel configuration... You should check this doc, at least, for the redhat config : http://www.automated.it/guidetoasterisk.htm Then, installation of Bristuff works like as charm ! Bye David Masure -----Message d'origine----- De : vdasilva [mailto:aquamoon@wol.co.za] Envoy? : mardi 28 juin 2005 09:01

Hi, I'm facing something strange but maybe I haven't the right solution. What I want ot do is : Someone from outside call my phone number, I check some informations using an IVR script and then I want to transfer the call to an external phone number. The point is that when I'm doing that using the Dial command, I've got a message telling me no is able to answer the call. But

What do you want to knwo exactly ? The way I use to transfer the call is the same if it was an internal extension, I use the Dial command using the trunk to go outside ... Ask me the info you need ... Thanks David -----Message d'origine----- De : Bob Goddard [mailto:asterisk@bgcomp.co.uk] Envoy? : mardi 14 juin 2005 12:52 ? : Asterisk Users Mailing List - Non-Commercial Discussion Objet :

Hi, I want to use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds.... but when I compile, I've got this message.... Does anyone have the same problem and/or know how to solve it ? Thanks Baste regards David Masure make[1]: Entering directory `/usr/src/asterisk/bristuff-0.2.0-RC7k/asterisk-1.0.6/cdr' gcc -pipe -Wall -Wstrict-prototypes

I've been trying to set up incoming faxes using spandsp with a HFC card. Unfortunately, incoming faxes are of very poor quality, the pages are not transferred wholly (sometimes only a bit of a page is transferred etc.). From what I've read, I may have troubles with correct "timing" (and need to set up ztdummy etc.). On the other hand, it is impossible to set ztdummy if one

Hi! I have just installed Redhat 9 and Asterisk to my computer, and now i have problems with my non-zaptel Card, I don't know how to set it up since all instructions are for digium's hardware. I have searched from the Internet for hours now, can you help me to understand all this HFC-s thing and how it is related to CAPI, ISDN4Linux, bristuff and so on. I have to say that I am not so

We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to

Hi, it took me 2 days to get my asterisk box running, so now I completed and I am disappointed of the sound quality. When I call other people their voices sound somewhat scratchy. First I thought it might be a codec problem, but I also recognized it during the ring tone or even the DISA connect tone. Sometimes it is better quality and sometimes more scratchy. Where might be the problem? I am

I'm still trying to install a HFC-s BRI card onto *@home .6 I'm new to this so I probably am overlooking the obvious. Can I just install BRIstuff onto a fresh *@home install? The BRIstuff installer downloads another * from Digium. Will this interfere with the @home install and must I comment out the Asterisk install in the BRIstuff install.sh file? Any pointers are much appreciated.

Hello. I'm looking for a fast way to group by similarity many (5-10k) long (2-10k points) time series. Using PAM on distance matrix obtained via as.dist(1-abs(cor(data))) produces usable results but it's rather slow and doesn't catch slightly shifted time series. DTW implementation from package 'dtw' is orders of magnitude slower even with global window constraints which

Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure -------------- next part -------------- An HTML attachment was scrubbed... URL:

i've been trying to get samba 3 to join my AD domain, and have gotten stuck. when I wbinfo -t it returns "Could not check secret" log.winbindd says: nsswitch/winbindd.c: nsswitch/winbindd.c:process_loop(722) process_loop: Invalid request size from pid 5368: 1304 bytes sent, should be 1568 here's my smb.conf incase it matters. # Global parameters [global] workgroup =

Hi all, I have an asterisk box set up with a bri card (using zaphfc). I have a bri cellular gateway connected to it beacuse I'd like to route all my cellular calls through that gateway. The probel I encounter is that when trying to dial a phone number, I've the message : unable to create a zap channel. My card is normally well configured because when connected to the NT, It works

----- Forwarded message from Don Armstrong <don@donarmstrong.com> ----- Date: Thu, 30 Sep 2004 16:19:41 -0700 From: Don Armstrong <don@donarmstrong.com> Resent-From: Don Armstrong <don@donarmstrong.com> To: submit@bugs.debian.org Subject: Bug#274301: libflac4 segfaults on corrupt flac files Severity: normal Package: libflac4 Version: 1.1.0-11 Running ogg123 on

Hello all. Since half december we are trying to implement * as our primary PBX. We had a test machine running with 2 sip phones and a single ISDN card. Since this was working ok, we installed * on our main Debian server. Some specs: P4 2.8 Ghz HyperThreading 512 MB RAM Debian Sid updated every week. Linux 2.6.9 vanilla kernel (not the debian package) 1 Intel Pro 100 nic for internal network 2

Hi, all. I'm doing an upgrade from an Asterisk at Home (Asterisk 1.x) system to stock Asterisk 1.4. Everything's working great, except that all the prompts (both stock system prompts on the new system and people's old recorded VM prompts) sound HORRIBLE. Call quality is great, both internal and external. Any idea as to what might have happened? Could I have brought over a config

Dear R People: Here is a really goofy question. I have some objects which have 2 classes: data.frame and ucr. Also, the classes will always be in that order. I have tried all sorts of things, but to no avail. listucrModels <- function(envir=.GlobalEnv, ...) { objects <- ls(envir=envir, ...) if (length(objects) == 0) NULL else objects[sapply(objects, function(.x)

Hi, I have installed Asterisk 1.4 with mISDN with the install-asterisk.tar.gz script from beronet.com. On my system I have two cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to work well with mISDN on my system from a previous installation. Now however, the AVM card works well at first glance, i.e. it "registers" incoming calls and works through the asterisk

Hi We're getting requests coming in for higher quality audio in our call recordings. We currently use MixMonitor and everything is being saved in it's native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a conversion to mp3 when the call ends and the 2 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the