search for: davec

Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the

I'm just forwarding this verbatim from somebody who works here. Comments ? mm ----- Forwarded message from Dave Chamberlain <davec at mv.mv.com> ----- > Subject: possible dovecot problem > From: "Dave Chamberlain" <davec at mv.mv.com> > Date: Tue, 20 May 2003 21:29:48 -0400 (EDT) > To: mem at mv.mv.com (Mark E. Mallett) > > Ok, I'm kind of confused at what dovecot's imap implement...

I realize the recent ssh exploit rumors appear to be false. However I've not saw any comments on hostgator's "patch" http://67.18.54.2/~davec/ssh_exploit_fix.txt They continue to talk as if they have inside information. Comments?

...a call to every parent plays msg c) coach calls in leaves a msg, kicks off a call to every parent, checks for voicemail, plays msg, if human, press 1 to acknowledge msg, coach gets call with a number of how many didn't acknowledge, maybe? so... has anyone done the proverbial phone tree? daveC -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000

Hello Fellow Asterisk Mailing ListMembers, When I tried to download the latest version of Asterisk this is what I get: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed Where are all the latest Asterisk 1.4.x source files? Thanks in advance, -E

I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated.

As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many

Hi All, On Tue, Sep 18, 2007 at 06:04:59PM -0700, Stephen F. Booth wrote: >> I think that what I actually want is for the operating systems to use >> .flac >> instead. Similar to what Mac OS X does with .aiff. This should probably >> have been posted in the advocacy emailing list instead. > > This is an interesting idea. There already is a FUSE filesystem called

I use qualify in sip.conf and need to setup a trigger when asterisk sees it as unreachable, so that I can either drop a call file, or send an email, or both. How can I do that? Thank you

hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/9003@S2) or visa versa. Both processes

Does anyone know what options you need to use with "sox" to output the audio in the WAV49 format that Asterisk uses.

1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070731/74328b51/attachment.htm

Hello,

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part

...velopment in the gov't, financial, and private sectors. have worked with open source products apache, dns, inn, ntp, majordomo, sugarCRM, joomla, and countless others... please contact me off list for specifics and to discuss potential projects. I am in the philadelphia, US area. thanks, daveC

Hi all, I will a TDM card with FXO modules on it. Below is the dial plan. When someone can 9123456, CLI will show dialing to 123456 and answered. After answered, the call hangup. I would like to know what will cause the case to happen. Anyone can give me some advice to solve it? exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT}) exten => _9X.,n,Hangup zapata.conf

hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run " show applications" at cli> . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta