similar to: Installing * on fedora 3

G'Day All, Greetings and best wishes. Many moons ago I had an Asterisk system running. Steve Totaro helped me quite a bit. Just now I installed Asterisk NOW 1.5 Beta, and am at the command prompt.....I thought there was a GUI with Asterisk NOW. Anyway, where can I find the install/config documentation or how to launch the GUI, as I have look around on the site but cannot locate it.

<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

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

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 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, 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; 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

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 all, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

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)

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

I'm experiencing a strange problem. My inbound calls are hanging up right after Background() message even though response timeout is set to 10 sec. [voicepulseincoming] exten=>_X.,1,Answer exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)

>>>>> 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)

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and