similar to: Stupid tricks: preventable?

Anyone know if - it is possible to limit 1 agent per extension where the last agent to log in overrides any previous agents or - a Command/application to clear all agents logged in on extension Does this look like it would require a custom mod to do it? J __________________________________ Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out!

Hello, We are running dovecot 1.0.5 on a test server, with FreeBSD 6.2 (though I have noticed the same problem since dovecot versions in the 0.99 range). We don't have very many simultaneous pop/imap users, but we have a proliferation of pop3-login processes. Currently we have 128 such processes. We have 11 imap-login processes, but only a few actual imap processes running. Is this normal?

Hello, We're having a problem in Dovecot 1.1.8 with a failed assertion on certain mbox format mailboxes. It happens both with deliver when it attempts to delier to the mailbox, and with IMAP connections for the affected box (though I'm not sure what they're doing at the time). Mar 3 12:55:26 <snip> dovecot: Panic: IMAP(<snip>): file istream.c: line 81 (i_stream_read):

Our current virtual mailbox configuration is not compatible with one of the assertions in the dovecot quota plugin's assertions in quota-dirsize.c. I believe the assertion is incorrect, but I would also be happy if I could get the same result with a better configuration setting. Here is a sample passdb entry which causes the quota assertion to fail: test at

I posted this 1.2.0-beta1 success story to asterisk-dev, and someone recommended that asterisk-users might benefit from it as well. Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com ---------- Forwarded message ---------- Date: Thu, 22 Sep 2005 17:35:08 -0400 (EDT) Subject: [Asterisk-Dev] Re: Ring requested on channel already in use To: asterisk-dev@lists.digium.com > alan wrote:

Hello. I have a likely unusual request regarding IP based virtual dovecot users. When you specify a passdb passwd-file name containing "%d", then the domain portion is stripped from the login username, before the user is checked in the passwd-file. However, if you specify a passwd-file name containing "%l" (the local IP), the domain portion of the login is not stripped off

Auto Callback on Busy Register on Busy I have implemented it as 1- I store Caller and Called party numbers in database when Called part is busy 2- I retrieve it from database and Caller is called by called party when Called party hangs up It is working fine with all kind of SIP phones I have with me basic configuration for extensions.conf is given and can be accommodated according to

Hello, In our configuration, we are using a "passdb passwd-file", with "user=" directives in each username, and a separate "userdb passwd-file" which contains the target usernames for the "user=" directives. This works fine, for normal logins via POP and IMAP. For customer support testing purposes, we also set up a temporary "master=yes"

Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. One feature the SPA-841 has, which I can't figure out how to implement on the snom 190, is the "make/accept calls without registration" feature. Or more specifically, "produce a dial tone even if I'm not registered." I would like to set our

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

sorry, was only for users list... Hi Kevin, Hi list, you are right, acting now is not needed, when callbacklogin will be removed anywhere in future... But thinking how to realice alternatives can't be so wrong. Callbacklogin gives a very simple way to use more queues for one agent, which only has to logon to only one system. No need to make dbs or tables for saving, where the agent has to be

Hi Does anyone know how I can check if a callerID is more than 2 digits. I am setting up my phones so that if the callerID is 3 digits the phones ring one way if it is more than 3 digits it rings another i.e. internal calls and external calls. exten => 2222,1,GotoIf($["${CALLERIDNUM}" = "1111"]?5) This will tell it to jump to 5 if callerID if 1111 but how do i

Hi, I need to compile Icecast on a kind of ancient AIX machine. I compiled all the required libraries, except libvorbis, and I can't make it compile at all. Is there a way to compile Icecast without Ogg Vorbis support? Thanks! -- Regards, Alek Andreev alek@zvuk.net

Hi, I had my stream running from ezstream 0.1.2 to icecast 2.2.0. It's been going for about a week, but two days ago I changed my icecast.xml. Since then, Icecast dropped my sources and stopped transmission twice. Everytime, it spew out xmlEncodeEntitiesReentrant : char out of range on STDERR and [2005-02-07 01:14:47] WARN source/get_next_buffer Disconnecting source due to socket

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

Hi David, On 28.10.2011 13:05, David Chisnall wrote: > I disagree there. Perl is pretty much guaranteed to be installed on any UNIXish system. Even FreeBSD, which has removed it from the base system, tends to install the Perl package by default. In contrast, a lot of the machines I use don't have Python installed. I need to install it if I'm doing LLVM development because it's

Hi all, I know this issue has been discussed over and over again, but I'd like to voice my opinion while 3.0 is still fairly early-ish in the pipeline. So the issue is... API breakage. I understand and agree with the rationale why, namely faster development. But this principle should mean that for each breakage, the dude who makes the breakage should accompany the final commit (or something

Alek Paunov <alex at declera.com> writes: >> That said, if the information required for the build is going to be >> made explicit, maybe this isn't such a problem, as other tools can be >> written to parse it and run the build. > > Absolutely - once the generators are prototyped and tested in Python, if > current Perl (presence) > Python's, they can be

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the

I have the following in my dialplan, counts the number of loops and when it hits greater then 5, exit. It works, but errors initially with, "syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or tolken; Input: +1". Could somebody tell me why? Thanks: ; **************************************** ; Setup a varriable to count the number of ; times the message has been