similar to: Upgrade CISCO 7960 Question

Hi all, As promise, the new prerelease (0.9.2) is now available for download from the followiing locations: http://www.laser.com/dante or http://www.geocities.com/tdanro A detailed help file is available online and in the application package as a chm file, accessible from the app help menu too. Unfortunately the IAX2 support is not ready yet, but I work on it now (next on my list). The DLL used

I have read on this list that the config files might be available to make them work on Asterisk? If that is so, could someone please email them to me? We have the Polycom Soundpoint IP 500 phones. Thanks a bunch, my goal is to make this phone and asterisk my business system. Thanks Sean sean@siskiyoutech.com

I am trying to figure out the correct syntax for the CLI command "SIP SHOW CHANNELS". I have tried SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected such as: -- Zap/15-1 is ringing -- Zap/15-1 answered SIP/206-4299 asterisk*CLI> sip show channel SIP/206-4299 No such SIP Call ID 'SIP/206-4299' I always get the "No such SIP

hi, i saw the cdr file called Master.csv and i want to know what these represent. examples "","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u8888","2003-11-07 17:43:04","2003-11-07 17:43:04","2003-11-07 17:43:22","ANSWERED","DOCUMENTATION"

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Hi all, A first basic version of DIAX as an ActiveX can be downloaded from: http://www.laser.com/dante/diax/activediax.zip There is only one small file (diax.ocx) and a readme.txt with the usage instructions. For the moment you can only place authenticated (or not) calls and there is no feedback (ring, messages, etc) Put this simple thing on your web page and you will be able to dial from any

I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following: exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70) There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the

We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with

To recap: T100P card wouldn't sync with the telco using line side clocking ( span=1,1,0.........) Had to use internal clocking (span=1,0,0.......) zttool showed Tx/Rx Levels as 0/ 1 For the grins of it I replaced the T100P card with another newer card from inventory. This newer card has the same rev on the ASIC / FPGA but doesn't have any of the various jumper headers installed

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e9dceeb4/attachment.htm -------------- next part -------------- Hello One of the option in VoicemailMain() is "Adavance Options". Could anyone explain what are these ?. Because whenever I select Advance Options, it repeats the same process of asking "Change Folders,Advance

Should FAX transmission generally work through Asterisk and a TDM400P connected through a PSTN gateway? At first blush I'd think that if they're all g.711uLaw encoded that it would work. But experience shows otherwise. Is there a better way to do FAX? -brian

Does anyone know how to send a message to a Cisco 7940/7960 phone running SIP images 6.3 telling it to light up one of its LED's when new voice mail arrives? I found alot of web based solutions http://www.voip-info.org/wiki-Asterisk+GUI and easy ways of getting email or getting paged of a new voice mail - but nothing where you can just look at the phone and see a blinking light or

Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb

Hi. Can anyone help me with the following: [globals] OFFICEHOURS .................................... [internal] exten => *80,2,SetGlobalVar(OFFICEHOURS=100) exten => *80,2,SetGlobalVar(OFFICEHOURS=200) .................................... [incoming] exten => s,1,GotoIf($[${OFFICEHOURS} = 100}]?incoming-officehours:incoming-officehours-off 1. Am I using the right sytanx when

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/c9e420c5/attachment.htm -------------- next part -------------- Hello all, I have 1 IP 7940 with the following Firmware versions App Load ID: P00303011201 Boot Load ID: PCO303010001 Version 3.1(12.1) Could you please confirm, if my IP phone has the correct SIP image. My asterisk

What happened to "sip show peers" in 1.4.13? Jerry

Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/aa4eda3c/attachment.htm