similar to: Asterisk NOW - Where to start

G'Day List, Can someone help me out a bit please. I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying to install * I am following http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation After running: cd /usr/src/asterisk make clean make make install make samples The instructions says: Configuring Asterisk - Login to your server as user

<P>I am running on Core 3 also with a voicepulse account.</P> <P>I found this document quite helpful....www.voip-info.org/tiki-print.php?page=Asterisk+Fedora+Core+3</P> <P>I did deviate in that I ran my make of Asterisk itself as follows</P> <P>cd /usr/src/asterisk</P> <P>make clean <BR>make linux26 <BR>make install <BR>make

G'Day All, Can anyone give me some direction in setting up the TFTP server on my RadHat ES3 box? I did quite a bit of reading, but I think I am more unsure now than before. I found the information nebulous. TFTP is already installed. I am trying to determine where the root directory for the tftp services is located so I can copy the CISCO 7960 firmware files onto it. Thanks.... Ferg

Thanks everyone for your feedback, especially Mark. I now have the ALL the files I need. My order still stands for the $8.00 product from CISCO but the CP7960 dealer sent me all the files. Now I will move on to completeing the setup of the TFTP server. Thanks again -----Original Message----- From: Michael Loftis [mailto:mloftis@wgops.com] Sent: Friday, February 18, 2005 7:51 PM To:

G'Day All, Newbie here. How can I go about troubleshooting a fast busy when I dial my the phone number on my * server? Thanks. Ferg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041012/a669d795/attachment.htm

G'Day All What do I type at the command line to stop and start * on a RedHat ES3 box? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041202/f9c92727/attachment.htm

G'Day All; So I purchased a Cisco 7960 and am now trying to get it configured for *. No can do without the variuos files/images through a FTPF server. I configured the TFTP server on my RHES 3 box, now to get the required CISCO files. So I contacted CISCO to purchase the required maintenance contract so as to gain access to the download area for the files/images. -WHAT A FRUSTRATION!!-

G'Day All. So I got the TFTP server all set up -thanks to much help from this list- the 7960 found it and updated to SIP the first firmware P0S30200. What I am now trying to do is upgrate through all the versions, as recommended, to the latest version, P003-07-3-00. I thought this would be accomplished by simply changing the sole line in the OS79XX.TXT file to P0S30203 and reboot the phone.

G'Day All, How do I fix this: I receive a call at the extension. Press the hold button. Music on hold starts. When I place the handset back on the cradle, the call gets hung up/disconnected. The Phone is A GrandStream Budge Tone 100. Thanks

I have read everything I can find in the docs and faqs about this, and I feel there must just be some simple thing I''m not doing, but I''m stumped. Two interfaces, eth0 and eth1.  eth1 is the WAN connection to the upstream provider, and has a single IP and the default gateway.  Connection uses bgp. eth0 is the LAN interface, and has multiple IP addresses, private (ie., 10.0.2.x)

G'Day All, I have read a lot but still have not got the concept in my head. My ultimate goal is to setup asterisk with VOIP at my job but am working on setting one up at home first. I have a 10/100 network at home with Cable broadband and would like to setup a server with RedHat and Asterisk. Do I MUST HAVE a VOIP provider so as to be able to make and receive calls VOIP? What should be my

Same here. I called them yesterday plus email and still no reply. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of brett-asterisk@worldcall.net Sent: Friday, January 21, 2005 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and

Hi -- I was happily running Shorewall 1.4 for quite a while, then upgraded recently to 2.0.13 (Debian Linux, Exim4, Squid, no DMZ) and can''t get Shorewall working. When I start it, my ssh and pop3 access is immediately blocked locally. This is a very simple setup. No dmz, just ETH1 to the Internet and ETH0 local, with IP Masq. Turned ON. I would appreciate it if someone who has a

In R 3.5.0 using the `encoding' argument of source() prevents loading files from the internet; without the `encoding' argument files can be loaded from the internet, but if they contain non-ascii characters, these are not correctly displayed under MS-Windows (but they are correctly displayed under GNU/Linux). With R 3.4.{2,3,4} there is no such problem: using `encoding' the files are

>>>>> peter dalgaard >>>>> on Sun, 3 Jun 2018 23:51:24 +0200 writes: > Looks like this actually comes from readLines(), nothing > to do with source() as such: In current R-devel (still): >> f <- file("http://home.versanet.de/~s-berman/source2.R", encoding="UTF-8") >> readLines(f) > character(0)

I just built the latest R-patched from source (SVN-Revision: 73548, Last Changed Date: 2017-10-12) and the build completed without a problem but `make check' errored out: running code in 'reg-tests-1d.R' ...make[3]: *** [Makefile.common:100: reg-tests-1d.Rout] Error 1 make[3]: Leaving directory '/sources/R-patched/tests' make[2]: *** [Makefile.common:275: test-Reg] Error

On R 3.5.0 (Mac) The issue appears when using the default (libcurl) method and specifying the encoding Note that using method='internal' causes a segfault if used in conjunction with encoding. (and works when encoding is not set) urlR <- "http://home.versanet.de/~s-berman/source2.R" # works url_default <- url(urlR) scan(url_default, "") # Read 7 items # [1]

On Fri, 13 Oct 2017 22:36:48 +0200 Stephen Berman <stephen.berman at gmx.net> wrote: > I just built the latest R-patched from source (SVN-Revision: 73548, Last > Changed Date: 2017-10-12) and the build completed without a problem but > `make check' errored out: > > running code in 'reg-tests-1d.R' ...make[3]: *** [Makefile.common:100: > reg-tests-1d.Rout]

I am trying to get two servers to copy each others data to the other server. I need it to be done real time a not use a cron. Can rsync running as a daemon monitor the files system to trigger a transmission? If so how do I configure it? I am running Solaris 9. Thank you. ______________________ Charles Berman Senior Unix Administrator "Think Globally" The contents of this email

Has anyone had any success with the mongrel_secure_download gem? I keep getting "connection reset" errors. -- Cheers, Kevin