similar to: IMAP STARTTLS Problem

I'm using thunderbird 5.0 with dovecot 1.0.15. My client stuck downloading IMAP header every 10-15' Until the FETCH is done from 1 to 76687 (it's needs a looong time), my client isn't usable Could you help me to fix this issue? Below the log. Thank you 2680[9cc6d00]: 9ccf800:mailserver:A:CreateNewLineFromSocket: * FLAGS (\Answered \Flagged \Deleted \Seen \Draft unknown-1

Hi, Unfortunately it looks like the parameter: "imap_client_workarounds = tb-extra-mailbox-sep" does not work. Neither inside nor outside "protocol imap {}" configuration section. ACLs are set correctly. Manual "DELETE Public/Test" via telnet is working as expected. Is this a known bug? See: - https://help.ubuntu.com/community/Dovecot#Thunderbird -

Hi Anton, Thank you very much for your helpful hint. Thunderbird clearly wants to move Public/Test to the Trash of the user who subscribed the Public folder. Question is how to solve this from a Dovecot point of a view, so a user can also delete folders? 2020200448[10053c700]: proposed url = Public/Test folder for connection INBOX has To Wait = FALSE can run = FALSE 2020200448[10053c700]:

Hi folks. Here's our situation: Migrating from UW-IMAP. Have lots (as in, tens of thousands) of clients set up using '~/mail' as the IMAP root, and using subscriptions. Dovecot mail_location is using '~' for mbox storage. We've disabled full filesystem access in Dovecot in order to use ACLs for shared folders. Hence we have a hidden namespace for backwards

Hello, the error is still present: May 8 19:47:18 opsys dovecot: imap-login: Disconnected (no auth attempts): rip=82.113.119.140, lip=78.46.216.126 Whenever I start a session with openssl to STARTTTL (Server: mail.opsys.de) the handshake is successfull. Also I am able to login to my account via 1 login. In Thunderbird port 993 for SSL/TLS works correct, only STARTTLS on port 143 isn't

hi all, i just inherit a server: Compaq ProLiant ML370 Generation 2 Server with Smart Array 5300 Controller with 6x18.2GB hard drives. i want to install centos 5 on it with hardware raid 1. any idea how? it also has a build-in and add-on ethernet cards. how do i know which one is which to assign ip? thanks. t. hiep

On Mon, 1 Apr 2019 at 23:23, Kevin Brott <kevin.brott at gmail.com> wrote: [...] > port-aix.c:316:1: error: conflicting types for 'record_failed_login' > record_failed_login(const char *user, const char *hostname, const char *ttyname) I think I just fixed that by changing the function to match the new prototype. Unfortunately I can't currently test it myself. Please

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

Hi, I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace... 1) (this bit works fine) I want a queue caller to have access to the basic set of agents initially, with an overflow to additional agents if they are busy - This is done using penalty: queues.conf: member => SIP/dev1,0,Agent1 member => SIP/dev2,0,Agent2 member =>

I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp 0 788 0.0.0.0:5060 0.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

After tinkering with a hand-knitted extensions.conf based on "Asterisk - TFOT", I've now set up a server with Asterisk@Home and am experimenting with it. I'd appreciated any advice from the more experienced list members about which way to proceed. We (my wife and I) have two separate micro-businesses with two POTS lines plus fax. I'd like to have inbound calls on the two

I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great !

http://techworld.com/opsys/news/index.cfm?NewsID=3064 The fact that they talk about Notes (the client) rather than Domino (the server) suggests that they may be running the Notes client (as I and many others do) under Linux, and the only way I know of to do this is to use Wine. David

Or you can use authenticate() and have it take its 'passwords' form a text file on your machine. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > trixter aka Bret McDanel > Sent: Sunday, January 29, 2006 5:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion

>From the queues.com file. ; An announcement may be specified which is played for the member as ; soon as they answer a call, typically to indicate to them which queue ; this call should be answered as, so that agents or members who are ; listening to more than one queue can differentiated how they should ; engage the customer ; ;announce = queue-markq This allows you to have one

I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?" AT&T Local did not, But XO communications said they did. Before I call to complain, is there an setting to turn this on in asterisk? I want to make sure that I have my side covered before I call XO. My current zaptel.conf is: context=from-pstn switchtype=national

I am running Asterisk CVS-04/16/03-18:57:13, and mpg123-0.59r It all sounds great and it plays at the correct pitch and speed. However at the end of the file it simply does nothing. It does not go on the the next step in the extension.conf nor does it hang up. It just sits there. During play I have two processes running for the mp3 stream: root 6300 6299 8 22:32 ?

I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL: