search for: anomy

I've been doing research on switching our current e-mail server (qpopper, sendmail) to imap. The decision on which server to use is essentially down to Cyrus and Dovecot -- I like Cyrus' approach to a lot of things, but the "blackbox" nature of it makes some niceties like using spamassassin and procmail difficult, or at least counterintuitive. Dovecot seems to play nicer with

...quot;Dave Shariff Yadallee - System Administrator a.k.a. The Root of the Problem" <root at doctor.nl2k.ab.ca> Subject: Repeat mails. INC000023456674 From: ips at shaw.ca Date: Tue, 9 Aug 2011 19:17:34 +0000 X-Sanitizer: This message has been sanitized! X-Sanitizer-URL: http://mailtools.anomy.net/ X-Sanitizer-Rev: $Id: Sanitizer.pm,v 1.94 2006/01/02 16:43:10 bre Exp $ Dave, According to RIM your server is probably changing the assigned serial number or code to mails frequently and because of that Blackberry server takes the same mail as new one and draws again. Read attached link. In...

Hi! I'm currently running RH 9 with postfix, uw-imap and MailScanner. I want to install Fedora Core 2 (when it comes out next week) with postfix, dovecot and amavisd. I'm unhappy with the problems I'm having with MailScanner, with the difficulty I've had with authentication as I am currently configured, and want to move to Maildir instead of mbox. That being said, is dovecot

Hi List.. I m getting this mesg while trying to dial an extension, both SIP UAs are registered with asterisk, m trying to dial extension 1015 from UA 12321@xyz.com to extension 1016 of UA 77777@xyz.com In extensions.conf I added exten => 1015,1, Dial(SIP/77777,20,tr) Any hint? JF WARNING[16397]: File pbx.c, Line 1153 (pbx_extension_helper): No application ' Dial' for

Hi, I have enabled ztdummy in order to have * compile it. I can modprobe ztdummy with no problems. The sole reason, i need ztdummy is to heve musiconhold and meetme working. However when I start *, it says this and does not start. ---------------------------------------------------------------------------- ---------------------- == Parsing '/etc/asterisk/zapata.conf': Found

Hello Eric, call progress detection is the problem. Asterisk mistakenly recognizes the call to be answered and then still "hears" the ringing that should not be there if the line was really up. To solve the problem you would have to either implement a progress detection matching your country's indication tones or at least adjust the existing one for US or Costa Rica in dsp.c. By

Hi there, I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO line. The reason being is that Voicetronix sends out the DTMF too fast even before the line is fully established with the carrier. Usually when dialing an 8 digit number, only 7 digits are actually successfully heard by the carrier. Currently, my dial plan is: exten => _9.,1,Dial(vpb/1-1/${EXTEN:1}) Daniel

We encouter a situation where we need to use SIP info to convey infomation for one end point to another endpoint. I use asterisk to do the test and find asterisk does not forward the SIP info to another endpoint, but act as UAS and returns a 4xx error message. I think asterisk is not right to handle this SIP info message. In RFC 3261 Page 70 "This protocol is designed to be extended.

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Hi. Is it just me or is there an elephant in here? </tongue-in-cheek> The official Markdown homepage (http://daringfireball.net/projects/markdown) offers the 18-month-old version 1.0.1 as a download, and the Dingus page purports to be run on a version 1.0.2b6 - which seems to offer pretty much the same functionality as the old 1.0.1 version. My question is: Is there any active development