similar to: TDM410E card, 1 FXO module - how to dial Out

On Thu, 2005-01-13 at 12:42 -0500, Jean-Marc Valin wrote: > Le jeudi 13 janvier 2005 ? 10:59 -0500, Jared Whitby a ?crit : > > Interestingly enough.. I started playing around with preprocessing > > options in 1.1.6 and happened upon the denoise filter > > (SPEEX_PREPROCESS_SET_DENOISE). When i run the test tone using that > > option it is completely filtered out and I

Interestingly enough.. I started playing around with preprocessing options in 1.1.6 and happened upon the denoise filter (SPEEX_PREPROCESS_SET_DENOISE). When i run the test tone using that option it is completely filtered out and I just get (complete) silence. When the test tone is intermixed with regular voice I only get the voice. So while i still don't quite understand why the test tone

Hi all! Did the nested xen stuff make it into the xen-unstable (4.1-rc1?) tree as suggested back in January by Tim Deegan? TIA ________________________________ Date: Fri, 7 Jan 2011 16:01:12 +0000 From: Tim Deegan <Tim.Deegan@citrix.com> Subject: Re: [Xen-devel] [PATCH 00/12] Nested Virtualization: Overview To: Christoph Egger <Christoph.Egger@amd.com> Cc: Keir Fraser

On Tue, 01 Oct 2013 06:10:46 -0600 Eric Blake <eblake@redhat.com> wrote: > On 10/01/2013 06:04 AM, James Gibbon wrote: > > > > > > Hello all, > > > > I have a KVM guest VM which is a clone of a production machine > > running on a different physical server, incarnated from an > > image backup. > > Careful. You need to scrub more than

*I have a Ham Radio program, named CQ100, it works fine using WINDOZE, but when I installed the same program on my linux system everything works except there is no-audio I'am using Ubuntu 7.10 linux... The author told me that windoze uses GSM 6.10 codec, plus ACM audio compression manager, these are built-in... So by anychance does anyone know of a program that one can get to use on a

I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco, Span 2 sends to my existing phone system(Nortel). My Span1 gets sent to the context from-pri, detailed here: [from-pri] exten => _49XX,1,Set(CALLERID(all)=${CALLERID(all)}) exten => _49XX,2,Dial(Zap/g2/${EXTEN},,twk) exten => _49XX,3,Congestion() exten => _49XX,4,Set(CALLERID(all)="") exten =>

I'm trying to build R from source on Ubuntu Gutsy Gibbon. I've done apt-get install r-base-dev and apt-get libX11-dev, but R configure is still complaining about X11 headers/libs are not available. What else do I need? Thanks, Paul ==================================================================================== La version fran?aise suit le texte anglais.

?? Hello.? We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).? We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.? If a person starts talking they will cut off others on the call.? Is this normal behavior?? Are there any options I can change to change this? ?? Thanks! James -------------- next

Evening all: Thought I'd announce myself with a question about setting up sources.list to be able to begin download and install of R. Running version 7.10 of Ubuntu. (Tried 8.04 but found it slow and finicky with my hardware. Perhaps will try 8.10 or above, but that's a thought for another time.) Followed the notes on the R site and added the following lines to my sources.list: deb

Hiya everyone. I have been working on a fun little app to watch what's going on in your asterisk box via its manager interface. There's a screenshot up and some info at http://sitkavoip.com/astsee/ -- Sorry it requires allegro, but I was more keen about getting the ideas down than worrying about the framework. Comments/questions welcome, but probably off-list is best unless they

Hi List; I read the following sentence: "The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable" In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad

I am aware that there are provising for redundant database connections Basically i was working on my main db server (which is also a mail sever) I current have this in the dovecot-pgsql.conf ______________________________________________________________________ driver = pgsql connect = host=localhost port=5433 dbname=scom_billing user=pgsql password=xxxxxxxxx default_pass_scheme = PLAIN

Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It

Hello All, I have a very large data set (1.1GB) that I am trying to read into R. The file is tab delimited and contains headers; there are over 800 columns and almost 700,000 rows. I am using the Ubuntu 7.10 Gutsy Gibbon version of R. I am using Kernel Linux 2.6.22-14-generic. I have 3.1GB of RAM with the AMD Athlon(tm) 64 Processor 3200+. I downloaded R using the instructions from cran under

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST

*I have asked this once before, but never got any good info as how I can get this codec using wine or crossover...Hopefully some one may have the answer for I am sure that there are many Ham operators out there that would like to find out also... The program CQ100 uses the GSM 6.10 codec that ships with windows, it also uses the windows ACM system (Audio Compression manager)... This info came

Hi, OS: Ubuntu (Gutsy Gibbon) 7.10 Ver: Wine 1.0.0 I tried to install Wine onto my system, but I get the following error. Code: libaudio2 depends on libc6 (>= 2.7-1); however: Version of libc6 on system is 2.6.1-1ubuntu9. dpkg: error processing libaudio2 (--install): dependency problems - leaving unconfigured Errors were encountered while processing: libaudio2 Please help me

Hi, Is it possible to get the remote channelname that will be bridged when the call is answered, only having the channel that is in the Ring(ing) state? As far as I can see no variable seems to fit when doing the show channel command. I want to be able to redirect/manipulate an incoming call before it gets answered/bridged, but to do that I have to now which channel to use. Is there a way?

Can I do this? I have a x100p card on my PSTN line and I have an incoming context for these calls which uses the "s" extension. I'm wanting to set up a simple IVR and would like to be able to test the dialplan as I go. But having to dial-in on my PSTN line each time is going to cost me money. Can I connect to my zap_incoming context from my locally connected SIP phone? I'm

Hello, I am a newbie when it comes to Dovecot, and in following this ( http://workaround.org/articles/ispmail-etch/ ) document in setting these things up, I have run into an error that I am not quite certain what to do... I am running Ubuntu Gutsy Gibbon, and using 1.0.5 for Dovecot. The message I am getting when I try to 'restart' Dovecot (using '/etc/init.d/dovecot restart') is: