Displaying 20 results from an estimated 600 matches similar to: "Dovecot: Mails flagged as read get flagged as unread"
2016 Dec 06
3
Dovecot: Mails flagged as read get flagged as unread
Hi Steffen
Thanks for the reply!
- Users are accessing from multiple devices simultaneously
- Each user has his own mailbox, we do not use shared mailboxes
regards
On 12/06/2016 10:14 AM, Steffen Kaiser wrote:
> On Tue, 6 Dec 2016, plataleas wrote:
>
> > We experience some unexpected behavior with dovecot. It happens that
> > emails marked as read get marked as unread (MUA
2016 Dec 06
0
Dovecot: Mails flagged as read get flagged as unread
Hi, There is a plugin "mail_log" that you can use to audit what
users/user agents are doing, probably mail_log_events flag_change is
helpful?
--
peter
Am 2016-12-06 um 10:24 schrieb plataleas:
> Hi Steffen
>
> Thanks for the reply!
>
> - Users are accessing from multiple devices simultaneously
> - Each user has his own mailbox, we do not use shared mailboxes
>
2016 Dec 06
0
Dovecot: Mails flagged as read get flagged as unread
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
On Tue, 6 Dec 2016, plataleas wrote:
> We experience some unexpected behavior with dovecot. It happens that
> emails marked as read get marked as unread (MUA is Thunderbird on port
> 143). Unfortunately this happens randomly, reproducing this issue is
> difficult. We could not find any pattern, it happens rarely.
does your user(s) access
2015 Jun 23
0
Problem with LDAP... again...
Hi list!
I'm always trying to configure Dovecot to ask our LDAP-Server (AD) in
order to authenticate the users.
I really don'know what can I do wrong...
I configured my Dovecot so:
hosts = chimaera.company.local
dn = CN=mailproxy,CN=Users,DC=company,DC=local
dnpass = SECRET
sasl_bind = no
tls = no
debug_level = -1
auth_bind = yes
ldap_version = 3
base = dc=company,dc=local
deref =
2005 Feb 09
0
Asterisk and SIPphone won't cooperate
When attempting to call one of the example numbers, like 17474745000, I
only get "488 Not Acceptable Here". It works fine when I configure the
softphone (Xten X-Lite) to use sipphone's server directly. Am I missing
something? Here's my relevant config sections:
sip.conf:
in [general]:
register => 17472442457:mypassword@proxy01.sipphone.com
[sipphone]
type=friend
2015 Jun 22
0
LDAP authentication
If you allow anonymous search on AD maybe you can try to set auth_bind =
no .
a.
On 22/06/15 17:19, Luca Bertoncello wrote:
> Hi again
>
> I'm trying to authenticate a user against an LDAP Server (well, our
> AD, but it can LDAP).
>
> This is my configuration:
>
> hosts = my.server.local
> auth_bind = yes
> ldap_version = 3
> base =
2015 Jun 22
4
LDAP authentication
Hi again
I'm trying to authenticate a user against an LDAP Server (well, our
AD, but it can LDAP).
This is my configuration:
hosts = my.server.local
auth_bind = yes
ldap_version = 3
base = CN=Person,CN=Schema,CN=Configuration,DC=company,DC=local
scope = subtree
user_attrs = \
=home=/home/imapproxy/%u, \
=mail=maildir:/home/imapproxy/%u
pass_attrs = uid=%u, userPassword=%w
2010 Dec 26
1
Asterisk 1.8 Realtime Queue not working
I have configured my mysql database by following this link
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
The only difference is that I am using ODBC instead of MySQL with Realtime.
Within extensions.conf I have the following for my queue
exten => 9**2**1611,1,Answer
exten => 9**2**1611,2,Queue(irock.com,tT,,,300)
exten => *50,1,Answer
exten =>
2005 Mar 03
2
FWD and SIPPHONE problems after upgrading to CVS HEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls
2009 Feb 17
0
Optimizing this script for calling Skype users from Asterisk
I have written this configuration script which uses OpenSky to make Skype
calls directly from Asterisk devices using my companies SIP to Skype
gateway. Users can dial skype_anyskypeusername or manually add names or
extensions which can get mapped to the correct dialing sequence. The right
sequence is username at opensky.gizmo5.com but that gets mapped to sipphone
address so I set that up to map
2009 Apr 26
1
1.6.1: "DNS error" but ping works
With 1.6.1 svn:
[2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout:
-- Registration for '17470121145 at proxy01.sipphone.com' timed out, trying
again (Attempt #30)
[2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable
to lookup 'proxy01.sipphone.com'
[2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register:
Probably a DNS
2014 Apr 01
2
imap process and indexer-worker crash while creating folders
Hi,
our dovecot processes sometimes crash when we create new folders. The "imap" process and the "indexer worker" process is crashing. We can reproduce this, we have a java program with multiple threads, and sometimes 2 threads try to create the same folder for the same user, and if both collide, dovecot processes crash.
We don't see this happening in the real world if
2004 Feb 03
1
sipphone dialing out problem
Hello
when i dial a toll free no using sipphone i get this error message. How do i solve this?
Any help will be appreciated.
console message:
Starting simple switch on 'Zap/2-1'
-- Executing SetCallerID("Zap/2-1", "17473863282") in new stack
-- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack
-- Executing
2005 Jan 03
0
Re: Asterisk won't register with sipphone.com
Hello All.
I started setting up my Asterisk system yesterday and everything was going
well, i have registered with sipphone.com and set-up my Asterisk system to
register with sipphone per the sip.conf file below.
It was registered perfectly but I could not receive calls so I added in the
line "insecure-very" and I then used the Washington DC access number to test
and the phone
2005 Mar 12
0
Hang on "making progrogress passing" when dialing out
I am getting the following on dial-out via Sipphone to a 1-800 number
(numbers obscured):
-------------------------------------------------
== Spawn extension (macro-sipphone, s, 3) exited non-zero on
'SIP/eric-9546' in macro 'sipphone'
== Spawn extension (default, 1747xxxxxxx, 1) exited non-zero on
'SIP/eric-9546'
-- Executing Macro("SIP/eric-8e80",
2006 Feb 10
0
calling to sip provider
Hello,
I am new user of Asterisk. Yesterday I was trying to call from softphone
to Asterisk, and that Asterisk routes this call to sipphone.com provider.
I have found information on internet about how to register to sipphone
and it seems that I have done. "sip show status" (or similar
command) in CLI was showing me that I was registered.
To call was not working, and on Asterisk's
2003 Jun 18
0
Goups and domains trusted
Hello,
I'm using a proxy squid with authentification NT (Challenge/response) but i
have problem with domains trusted.
I have 2 domains DOMAIN1 and DOMAIN2.
When i use wbinfo i have these results :
wbinfo -t
secret is good
wbinfo -m
DOMAIN2
./wbinfo -a DOMAIN2\\proxy%proxy01
plaintext password authentication succeeded
challenge/response password authentication succeeded
It's very
2018 Dec 18
0
Errors with missing links to files when using Single Instance Storage and zlib (Dovecot 2.3.4)
I'm running Dovecot 2.3.4 with Single Instance Storage (SIS) and zlib
and I am frequently seeing errors where files are missing in the
attachments directory even though the zipped file in the hash directory
is actually there.? This appears not to have been an issue in Dovecot
2.3.1 Requested output of the server below.
The error looks like this in the log
Dec 18 08:58:58 dovecot01
2005 Aug 08
4
DTMF issues with SIPPhone?
Does anyone else have DTMF issues with SIPPhone? When calling into my
DID, and entering, say, 1002. Sometimes it will recognize it properly
(rarely), other times it will receive something different. Such as,
1102 or 1000, etc. Has anyone else been having these issues? I'm
only accepting ulaw and alaw, and my relevant sip.conf information
follows:
[sipphone]
type=peer
2006 Apr 26
1
getting asterisk to reliably answer a voip line
I have a sipphone.com account, with asterisk set to
answer incoming calls, using the following settings
(phone number and password omitted) in the Peer
Details for the SIP Trunk:
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
fromdomain=proxy01.sipphone.com
fromuser=1747xxxxxxx
host=proxy01.sipphone.com
insecure=very
secret=xxxxx
type=peer
username=1747xxxxxxx
The Asterisk machine is