similar to: Help with concept.

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, 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

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.

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

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, 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 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!!-

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

G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Thanks much. BTY: Does anyone have a How-To on getting the 7960 fully configured

G'Day All; I am a little unsure on how to get Music On Hold to work. Please critique my extensions.conf. ????? Thanks ; SIP 5001 exten => 5001,1,Dial(SIP/5001) exten => 5001,2,Voicemail(u${EXTEN}) exten => 5001,3,Hangup exten => 5001,102,Voicemail(b${EXTEN}) exten => 5001,103,Hangup Thanks -------------- next part -------------- An HTML attachment was

Hi all, Looks like I'm going to make the trip over to Astricon next month, but finances being what they are and since I'll be paying for my own flight from the UK, I'm trying to cut down on costs. The problem I've got is that the hotel is about 4 miles from the nearest public transportation (Brookhaven station) and the shuttle bus is only for hotel residents. Hiring a car seems

I'm having some trouble trying to workout how to form functions from the c interface I thought I had it sorted but I guess I'm missing something or haven't understood the requirements, a case of trial and error and not really having a clue to start with! I've got binary ops, cmps, for loops, while loops working but then it hit the wall with a tail cmp loop. looking at the

Hi all, I've been batting my head against a brick wall for the best part of the day and still haven't got any further (apart from getting a big headache, that is). I've searched the Wiki and Googled the hours away, but I still can't find supportive documentation. I've just replaced my ISDN Fritz!/chan_capi setup with a HFC/Zap configuration and had the following problems: 1

Andrew Ferguson wrote: > I'm having some trouble trying to workout how to form functions from > the c interface > I thought I had it sorted but I guess I'm missing something or haven't > understood > the requirements, a case of trial and error and not really having a clue > to start with! > > I've got binary ops, cmps, for loops, while loops working but

One of our FreeBSD boxes has a SCSI controller and disk, which showed problems earlier this week. There was a lot of of chatter from the SCSI driver in /var/log/messages and to the console. However, the console is unattended and we only discovered the problem subsequently because /var/log/console.log didn't show any of the chatter. console.log is otherwise working, and very helpful (e.g. it

Hi all, I have just received the following e-mail from an Asterisk user: "I just made a call via BT to a mobile. Then an incoming call came in and Ann else answered it - it made my call go completely fuzzy and I could hear what the woman on the other line was saying to Ann but I couldn't hear my conversation! When Ann's call finished - mine went even fuzzier and all I could hear

Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) -

Hi, Bit of a puzzle this one - let's see if anybody else can shed some light... I Ghosted my Asterisk box to build one for a colleague. I added one HFC card to make the total two and amended zaptel.conf and zapata.conf accordingly. I tested it with my handsets on my ISDN lines and it worked AOK. I unplugged the box and took it to his house and plugged it in. He doesn't have any ISDN