similar to: Any way to get out of a remote console without stopping *

The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies

Hi, Please let me know if you have any ideas - I am taking wild guesses now.... Here is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12 Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a chanel bank although it only has 2 outputs) in a CAC unit. The unit also has a router card that runs the data side. My extensions are all SIP

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

Reid Spencer wrote: > On Mon, 2006-02-27 at 20:43 -0800, Wink Saville wrote: > >>> Try passing "-L/opt/llvm-1.6/llvm-gcc/lib/ -lcrtend" to llvm-ld. >>> >> This didn't work for me: >> llvm-ld -o t1.app t1.bc t1.a -L/opt/llvm-1.6/llvm-gcc/lib/ -lcrtend.a >> >> I get two warning's about library 'crtend' missing

On Mon, 2006-02-27 at 20:43 -0800, Wink Saville wrote: > > Try passing "-L/opt/llvm-1.6/llvm-gcc/lib/ -lcrtend" to llvm-ld. > This didn't work for me: > llvm-ld -o t1.app t1.bc t1.a -L/opt/llvm-1.6/llvm-gcc/lib/ -lcrtend.a > > I get two warning's about library 'crtend' missing > You didn't get the instructions quite right. The option is

Hello again, I have a little problem, every time I switch on the Asterisk server I must load two modules: modprobe zaptel and modprobe wctdm Is there any way to load there automatically when the server start? I have a Debian Etch. One more cuestion, it's posible to start Asterisk (asterisk -vvvc)as well? What metod do you prefer? "asterisk" or "asterisk -vvvc"? Thanks

Chris Lattner wrote: > On Sun, 26 Feb 2006, Wink Saville wrote: >> I then do the following in my makefile: >> >> llvm-gcc -c t1.c -o t1.bc >> llvm-gcc -c t1sub1.c -o t1sub1.bc >> llvm-gcc -c t1sub2.c -o t1sub2.bc >> llvm-ar r t1.a t1sub1.bc t1sub2.bc >> llvm-ar: creating t1.a >> llvm-ld -o t1.app t1.bc t1.a /opt/llvm-1.6/llvm-gcc/lib/libcrtend.a

Hi, I have resources available to host a portal specifically for the Asterisk system, to help correlate documentation, FAQ's and How To I am new to Asterisk, and my hardest work is in locating information on using or configuring the software. Would Mark, John or any of you feel this would be of benefit to host? I am offering a PHP-NUKE portal for this, (Unless you know of a better

Has anyone been able to get conference calls to work on the Pingtel Phones? I assume this feature works with their implementation, but connected to my asterisk box it doesn't work. The Pingtel phone thinks it is making a second call, but asterisk never sees anything about a second call. Any help would be appreciated. Sincerely, Andy Hester Consero

Hi All, I am considering testing out some analog sets with * for a customer and thought I would ask what analog phones are in use? The customer would require the usual business functionality ie hold, conference calling, and preferably a soft key to vm and line apearances(correct terminology?) in order for secretary to see if their boss is on the phone before transferring a call, etc. I would

Please point me in the right direction as to where to increase the length of time I am sending DTMF tones on my T100P. I can't remember where/how I set it last time, and I need to do a fresh install. I'd sure appreciate it :) Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977

I came across this hack while looking through chan_zap.c. Is this still applicable for those w/ zplex cb's? Is it a maybe need or an absolutely need? Any background info would be appreciated. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977

I've wine installed as part of SuSE Linux 9.1 Pro and it works fine for several windows prgs. I try to install the Atari emulator Magic PC (because it is better than STonX). Wine starts the installer of MagiC, Setup.exe, which gives the message "can't open ... Setup.ex_". Setup.exe works under win98 but under wine it looks for itself with the wrong extension .ex_ MagiC PC test

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

#this is my stab at - I am sure that I am missing something. If this doesn't make sense then please ask for more details. #This may show my low level of programing knowledge hester. <- c(1,1,1,1,2,2,2,2,3,3,3,3,4,4,4,4) value <- rnorm(16) x <- data.frame(value, hester.) z <- (if(x[,"hester."]==1){ x[,"value"]*6.250} else if(x[,"hester."]==2){

Hello, Need some help using the speex_echo_cancel. I've read the documentation about the speex_echo_cancellation function: speex_echo_cancellation(echo_state, input_frame, echo_frame, output_frame); (in) echo_state => speex internal state. (in) input_frame => audio captured by mic. (in) echo_frame => the signal that was played in the speaker. (out) output_frame => the

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

On Monday, 10 January 2005 at 21:43, Greg J. Ogonowski wrote: > Orban Opticodec-PC LE/PE AAC/aacPlus Streaming Audio Encoder > > Low Cost Version now available. > > http://www.orban.com/orban/products/stream/1010_overview.html This list exists to help people to use icecast and related streaming software. It does not exist to provide free advertising. Your product is admittedly

With the increased traffic as of late, I'm wondering if it is time to split the list again. Specifically I am wondering if it should be split along the various VoIP protocols and zap hardware, then leave a general list that does configuration other than VoIP related? The hope is that those asking SIP or H323 questions could get help from the various supporters while the main list can deal