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Thought I would share this with you guys, it brought a smile to my face this morning... So, I work for a quick American company who I won''t name, and for the past few months I have been pushing to use AJAX in some of our Web applications. This looks like its finally happening, and we are using it in an upcoming project. There was a meeting about it yesterday, which unfortunately I

First, I have what I consider to be decent experience with SSH, and have been running VanDyke's VShell SSH server for considerable time under Windows 2000/XP. Recently, I have been working with Mac OS X and FreeBSD and have been using OpenSSH 3.4p1 The sftp server seems to have major issues when serving files, specifically, if one data stream is used the transfer rates fluxuate between 80 and

Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is

...local vendor has Polycom IP500 phones for $174 shipped to me. IP500 would be comparable to a 7940G I'm assuming. I ran into the same problem with pricing, don't want grandstreams, but can't afford the nice Ciscos. Check out the Polycom's. -Tim -----Original Message----- From: Brad Stockdale [mailto:brad@greenepa.net] Sent: Monday, August 30, 2004 7:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: New to Asterisk and a question Greetings all, I have been watching Asterisk for a while now, but haven't had the nerve to jump in and start playing until no...

Hello all, I am working on setting up a basic Asterisk system (using Digium FXS and FXO cards, Polycom IP phones). I've got a few questions in regards to voicemail and was hoping that someone could give me some ideas... Right now we have three incoming POTS lines. There are times when all three lines are used. When the lines busy out, the incoming call is sent to a voicemail

Greetings all, I have been watching Asterisk for a while now, but haven't had the nerve to jump in and start playing until now... I'm fed up with our phone system (or lack thereof) at my office, so I decided to start seriously looking at Asterisk... Mostly as a plaything to get my hands on it, but with the goal of making our phone system here somewhat bearable... We have four

Indeed this might be the failing point... Unfortunately, because I have no Cisco CCO account anymore, I have no access to firmware... I will try to find a copy of an old firmware for these phones. If I can find one, I hope it fixes my problem. Thanks, Brad > Apologies in advance if this is a stupid comment, but don't you have to > convert to SIP at a much lower version than 8. I

Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up

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Greetings all, I'm new to asterisk... Finally got two IP phones and two TDM400P's (one with four FXO modules and one with four FXS modules)... Began experimenting a couple days ago and have ran into a very aggrevating problem. My system consists of the following hardware: Abit BL7-RAID motherboard Pentium 4 1.6GHz processor 512MB SDRAM 2 TDM400P cards 2 3Com 3c905