similar to: using dovecot in Asterisk imap storage

Hi,I have a problem, i can't get the authentication for the master user using dovecot - ldap authentication, in combination with sogo.There is a script that is used by Sogo to check and disable the vacation messagge: "sogo /usr/sbin/sogo-tool expire-autoreply -p /etc/dovecot/master-users" Insidie this file there is: sogo:{SHA}gkodfgdk6yujkgdklkkdfkgjrg9ugr9p= if I run the script

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Thu, 24 Jul 2014, Franck CALLEJA wrote: > Date: Thu, 24 Jul 2014 10:50:21 +0200 > From: Franck CALLEJA <fc193 at hotmail.com> > Reply-To: Dovecot Mailing List <dovecot at dovecot.org> > To: "dovecot at dovecot.org" <dovecot at dovecot.org> > Subject: Sieve plugin (fc17 + sendmail + roundcube) > >

Hello, I'm trying to configure sieve plugin on my mail server so that people can do mail filtering (and mostly vacation notification) on their account. The mail server is running fine since 5 years and is based on :- sendmail- dovecot- roundcube It serves 2 differents domains. I've recently installed pigeonhole for fc17, and managed to configure managesieve plugin in roundcube

Hello Aki, I have this problem just with 2.3.8, my self-compiled 2.3.3 works fine. I have previously tried to update from 2.3.3 to higher versions (possibly 2.3.5 or so), but always had this error, which is why I am always back to 2.3.3. Configuration is exactly the same. Here my output from "doveconf -n": # 2.3.8 (9df20d2db): /etc/dovecot/dovecot.conf# Pigeonhole version 0.5.8

Hi, I just set up my mail server using postfix / dovecot / mysql and it works well. I just have a problem : I'm a trying to have a overquota notification and the server sends nothing (mail server rejects incoming mail). Could you tell me if there is a problem with dovecot or postfix configuration? Kind regards,Long My conf : # 2.1.7: /etc/dovecot/dovecot.conf# OS: Linux 2.6.32-26-pve x86_64

Hello, I'm using dovecot v2.0.21. According to http://wiki2.dovecot.org/SSL/DovecotConfiguration,dovecot 2.x supports different SSL certificate for different virtual hosts by using "local_name" directive, but I can't get it to work. When testing the certificate using "openssl s_client -connect domain.com:pop3s" I get the default certificate instead of

Hey there, Just wanted to drop a line and let people know that I'll be heading to San Francisco for O'Reilly's Etel. If you are interested in attending, there are some free passes floating around. If anyone is interested in getting together for a beer, let me know! Info on the conference can be found here: http://conferences.oreillynet.com/etel2006/ I'm looking forward to an

hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another

Hello everyone, As I promised at eTel last week, I have finished up work on my "Asterisk Native Sounds" project. Here's a little diddy from astlinux.org: ----------------------------------- Asterisk Native Sounds are a collection of audio prompts for Asterisk. They will improve quality, reduce CPU usage, reduce latency, and (in some cases) eliminate the need for G729

Hi, I tried to use Monitor(wav,filename) function in dialplan to record Meetme conference. When I monitored on IAX2 or SIP channels in that conference It recorded all audio (in and out) but when I monitored on ZAP channels I could hear only IN audio and piece of OUT audio (announcement get pin and than nothing). Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package)

Hi. I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy server and my problem is that only the first user account get logged in and only that user is able to make call correctly. It seems to be a problem with authorization - I have noticed no "Proxy-Authorization" information in SIP INVITE, ACK, CANCEL messages. I have also noticed that when I remove

http://etel.wiki.oreilly.com/wiki/index.php/Main_Page -- ==================================================== J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' "Wise men talk because they have something to say; fools, because they have to say something." -- Plato -------------- next part -------------- A

Hi, I tried to use Monitor(wav,filename) function in dialplan to record Meetme conference. When I monitored on IAX2 or SIP channels in that conference It recorded all audio (in and out) but when I monitored on ZAP channels I could hear only IN audio and piece of OUT audio (announcement get pin and than nothing). Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package) and

Mondada Gabriele <g.mondada at etel.ch> writes: > I just moved to the CMake solution. By the way, the generated libs > haven't the same names. Which ones? The only difference is that we now generate .lib files where .obj were generated on the past, and require a parameter to be passed to the linker for including them on the final executable. > In my opinion, we have to choose

I have RoR running and have configured RAGI (per the new tutorial provided at ETel http://www.snapvine.com/code/ragi) but I get this: --- user# script/server => Booting WEBrick... [2006-01-30 14:20:25] INFO RAGI::CallServer: default-handler= port=4573 [2006-01-30 14:20:25] INFO WEBrick 1.3.1 [2006-01-30 14:20:25] INFO ruby 1.8.2 (2004-12-25) [i586-linux] [2006-01-30 14:20:25] WARN TCPServer

Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the help in advance. Regards Bilal

Hi, I have some problems and doubts connecting two asterisk servers. I have one asterisk (serverA), with 1 sip client registered (clientA). I have another asterisk (sever B), with another client (clientB). Now I want to call from client A to B and from B to A. Searching in google i find many configuration examples. For instance:

First of all sorry for the little offtopic post :-) For one of our customers i need to make a vpn between the Netherlands and Hungary. Over this vpn two * machines are gonna talk IAX and employees in Hungary are gonna use the Exchange server located in the Netherlands. So far no problem... The problem: I'm from in the Netherlands and don't understand the .hu websites :-( First question:

This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models

hi all whats there in store of ruby/rails to support voice recording feature ? thanks rohit --------------------------------- Yahoo! Mail Use Photomail to share photos without annoying attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060303/44230a4b/attachment.html