similar to: redirect actions exceeds policy limit

I'm having difficulty with the doveadm who command on a multi-instance setup of dovecot. When I run the who command on the non-standard instance with the -m flag (to see their mail location), this happens: [root at wardentest3 dovecot]# doveadm -i mailtest user -m warden doveadm(root): Error: user warden: Initialization failed: Namespace 'INBOX.': Ambiguous mail location setting,

I apologize if this is a stupid question but I just compiled 2.0.8 and am trying to use doveadm to get the quota for a user using the -S <socket_path> option because I have different options for the various local IPs that dovecot listens on. For example, my default quota is filesystem (NFS rquota) and I can get that by doing: doveadm quota get -u warden which tells me: Quota name

On RHEL 6.2 + dovecot 2.1.5 with the stats and fts (squat) plugins enabled and each user having their own uid on the system I get the following error in the logs when the indexer-worker process indexes some messages: Apr 25 09:56:19 wardentest3 dovecot: imap-login: Login: user=warden, method=PLAIN, rip=137.238.60.164, lip=137.238.2.240, lport=1993, mpid=19464, encryption=TLS Apr 25 09:56:26

Greetings, after testing a lot with my system I have to write to this list to ask for help from the cummunity. My system: * Ubuntu 12.04.5 LTS * Dovecot 2.0.19 * Dovecot config (extract) ????? 8< ????? plugin { sieve = ~/.dovecot.sieve sieve_dir = ~/sieve sieve_max_redirects = 10 } ????? >8?????? * Sieve filter rules ????? 8< ????? require ["copy"]; # rule:[Alles

This is somewhat related to my other query on the list regarding NAT traversal. I have heard many times that IAX is "NAT-transperant". I am unsure how it accomplishes this. I do know that SIP works like this: your SIP device send a request to the SIP server (usually on port 5060) with whatever command. The SIP server respends to your device's "apparent" IP and port (this

I have narrowed my problem in a prior topic (I''m having trouble with Routes) to the fact that I get the problem, below only when I reload my page. Clicking on links to go to the pages I want ... I seem not to have problems. Given that it seems to be a reload that is generating an "/undefined" path somewhere ... can anyone suggest what I should be looking for to fix the

I was wondering if there was a way of setting up the dialplan in a way that if you dial an extension that is NOT in the dialplan then it would play a not-in-service gsm file and then play congestion tones. I would rather like this better than just hearing a busy signal on my phones.. I DID search around on the wiki and using google and could not find anything. Thanks. -- Stephen Rosebush,

We need help.... We are doing a project for a statistical class in and we are looking at world record times in different running events over time. We are trying to fit the data with a negative exponential but we just cant seem to get a function that works properly. we have on our x-axis the date and on the y-axis the time(in seconds). So as you can imagine, the times have decreased and appear to

Hello, I have configured iRedMail ( 0.9.1) on Ubuntu (14.04 LTS). In my dovecot.conf file, I have set sieve_max_redirects as 500. In dovecot (2.2.9) configuration file, I see sieve_max_redirect as 100 but when I add email addresses more than 32, I see that redirect fails. Can you please help me to see this issue and fix it through? If you need anything from my side, please let me know.

Hello. I am looking for a simple Asterisk billing solution. I expect about 50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all IAX). I need something that can handle monthly fees and per call charges (depending on destination, obviously), and should provide a web interface for customers and administrators. Something that can tie in to one of the existing management GUIs

@Steffan: This is the error I see in log file: sieve: info: started log at Nov 27 14:30:15. main script: line 42: error: total number of actions exceeds policy limit (33 > 32). Regards... Shakeel On 11/27/2015 06:09 PM, Steffen Kaiser wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Fri, 27 Nov 2015, Shakeel Shahzad wrote: > >> I have configured

Puppet is failing to restart lighttpd using the Debian init script. Both the default action of stop/start and using the reload action, which basically does the same thing, fail. It seems to be a filehandle problem. Changing the execute method in service.rb to redirect stdout to /dev/null allows the daemon to restart. Otherwise, I end up with a zombie process and according to strace, ruby

That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aaron J. Angel Sent: Wednesday, 23

I changed fairly much in my app without constant testing ( my bad ) now i get an error which I cannot seem to debug point my finger on. I have a home controller which just displays a welcome page this had a index only action. Whatever controller I try to open i always get above error message ActionView::TemplateError (undefined method `show'' for

Dear all, I'm currently running R on my laptop -- a Lenovo Thinkpad X201 (Intel Core i7 CPU, M620, 2.67 Ghz, 8 GB RAM). The problem is that some of my calculations run for several days sometimes even weeks (mainly simulations over a large parameter space). Depending on the external conditions, my laptop sometimes shuts down due to overheating. I'm now thinking about buying a more

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

I just upgraded to 0.22.0 from 0.18.4 and I''m running into what I think are problems due to changes in scope rules. In my site.pp manifest, I have the following: import "classes/*" node nfsserver { include nfsserver } node webserver { include webserver } class cluster1_sites { website { "example.com": ...; "example2.com": ...; } } node

Just upgraded to the current stable ver. when I start asterisk with -vvvvvcg I get the following error [pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead Dec 23 19:25:33 WARNING[1633]: loader.c:440 load_modules: Loading module pbx_loopback.so failed! Asterisk

Thank you for the response Axel. I will look into that. I did attempt to switch the PAM/Kerberos authentication to Dovecot LDAP authentication, but now performance is unbelievably slow. For example, with PAM/Kerberos, a user can log into webmail and have all of their emails/folders showing almost immediately. When using Dovecot LDAP, it takes literally 8-10 seconds to see the same thing. I