similar to: Master user with "user="?

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):

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?

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

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

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:

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

I just experienced something I'd rather not experience again. Using a SPA-841 SIP phone connected to our Asterisk server, someone dialed their own extension, answered, and then transferred the call using the phone's "XFER" soft key. This does a SIP REFER. Now, the phone has dropped out of the loop, and Asterisk has connected the two call legs into a loop, as far as I can tell.

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

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

Anyone have any ideas why DTMF would not work on only one number? Looking through the logs, anytime a button is pressed, this is what shows up: 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Echo cancellation already on We

I finally added beginnings of unit tests to Dovecot v1.1 code tree: http://hg.dovecot.org/dovecot/rev/4e4a5d6bb2cb There are currently just a couple of tests implemented in test-lib (lib/ functions) and test-mail (lib-mail/ functions). I'm probably going to be pretty lazy at adding more of those tests, so if you've ever wanted to get your own code to Dovecot, now would be a great time to

Hi there, I have bought 70 sipura 841 phones for a customer of mine. When following the mass provisioning guide in the admin manual for the sipura, I see it download the spa841.cfg file from my tftp server Sometimes the phone also downloads is phone specific file via tftp, and it works okay then. But, after a reboot of the phone, it is very very likely that it won't startup

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

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

Has anyone got any information on bulk provisioning of Linksys SPA-941/94s? There is an overview in the admin guide but it refers to a different provisioning guide that I haven't found anywhere. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com

I was hoping to finally get shared libdovecot.so and libdovecot-storage.so libraries for v1.3, so Sieve (and maybe others) could link against them. But I'm running into trouble getting it to compile in Solaris 10. Could you non-Linux users test if this compiles with you? http://dovecot.org/tmp/dovecot-1.3.UNSTABLE.tar.gz Or if anyone has ideas why this happens, I'd like to know:

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, I'm preparing to convert from qpopper + UW-IMAP to dovecot. So far testing has gone very well. One problem we haven't figured out is that long-running POP sessions keep the mailbox locked, so that the MDA times out while trying to deliver. We're using "maildrop" as our MDA if that matters. We don't see this issue for dovecot IMAP sessions. During long-running

Good morning, I'm running Samba 3.6.5 on a Fedora 16 box, where I configured Samba to act as a PDC. A few weeks ago everything went smoothly; I then had to change the -physical- box running linux + samba, and everything went weird since then. First, I'm using the same smb.conf file from the previous box. The OS hasn't changed, Samba packages might have been updated, but since I

Greetings list, I'm trying to establish if there's an issue whereby certain telcos in certain countries have not updated the London, UK numbering plan to include some parts of the 020 3 range, despite it being in operation for some two years now. To help with this, I'd be most grateful anyone outside the UK could make a test call to +44 203 3393 7389. This is a simple test number