similar to: can not get t'bird to create folder

Few weeks ago I upgraded dovecot from 1.2 to 2.0.16 and antispam plugin to 2.0_pre20101222. Since the upgrade I'm not able to move messages to my Junk folder. In the maillog I have found this message: dspam[25060]: empty message (no data received) Message is copied from my INBOX to Junk folder, but dspam got an empty message and sent an error return code. So the moving operation is not

It is few months ago I requested help with combination dovecot - dovecot- antispam plugin and dspam. Now I got into troubles with a lot of spam delivering to users inbox. Problem described bellow is now better hidden but stil remains: When moving a message from INBOX to Junk, dspam got an empty message. I made a wrapper about dspamc and there is no input on stdio. The dspam was not trained

Hi, on my small Xen-virtualised server with 48 MiB RAM I use Postfix and Dovecot, because the Debian administrators dislike qmail [1], which is in my opinion despite some maintainability and code quality issues a quite well designed software, because it mostly follows the UNIX principles. Postfix is not able to sort my E-Mail into different Maildir folders and after I looked at procmail's

I can't get sieve to put virus files in the SPAM folder. dovecot -n # 2.0.16: /etc/dovecot/dovecot.conf # OS: Linux 2.6.38.8-32.fc15.i686.PAE i686 Fedora release 15 (Lovelock) auth_debug_passwords = yes info_log_path = /var/log/dovecot-info.log log_path = /var/log/dovecot.log mail_access_groups = mail mail_home = /home/vmail/%d/%n mail_location = maildir:~/mail mail_privileged_group = mail

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ]

Hello, I'm trying to configure the dovecot-antispam plugin. Now I've run into a problem: whenever I try to move/copy a message in/from the spam mailbox, Thunderbird (and also Sylpheed, haven't tried anything else) says: The current command did not succeed. The mail server responded: Failed to read mail beginning. and in the logs I get: dovecot: Aug 07 10:41:23 Error:

Hello Everyone, I'm running the antispam plugin on Dovecot 2.0.19 on Ubuntu Server 14.04 and I can't seem to get it to work. In the IMAP section of dovecot.conf I have the following lines: protocol imap { mail_plugins = $mail_plugins imap_quota imap_acl antispam # mail_plugins = $mail_plugins imap_quota imap_acl imap_client_workarounds = tb-extra-mailbox-sep # Maximum

On 08/29/2018 09:42 AM, Carlos Rojas wrote: > Hi > > Probably somebody is trying to hack your system, you should block that > ip on your firewall. > > Regards > > On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at gmail.com>> wrote: > > I'm getting invites to very high ports every 30 seconds from a

On 08/29/2018 11:59 AM, Telium Support Group wrote: > Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki: > > https://www.voip-info.org/asterisk-security/ > > > > -----Original Message----- > From: asterisk-users [mailto:asterisk-users-bounces at

I'm trying to allow access to the office from home. But the ip provider (cablevision) blocks udp 5060. I can see the register packets leaving on wireshark, but nothing received by office. Changed to port to 6111 and now the packets show up. In the server I've set port=6111 in the device in sip.conf, but * is NOT listening for 6111: netstat -an | grep 5060 tcp 0 0

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

On 12/02/2014 02:46 PM, Jeffrey Ollie wrote: > On Tue, Dec 2, 2014 at 1:22 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >> Or do I >> find a new place to put asterisk.pid? > > Also, if you use the native systemd unit file, you no longer need a > PID file, although you still need /run/asterisk to store the control > socket. > So systemd is taking

My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message. I'm thinking of: same =>n,VoiceMail(vm,u) same =>n,System(ssh myserver "emailVM

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A

On Fedora 20, every time the kernel updates, /var/run/asterisk owner is set to root.root. I'm running asterisk under user asterisk. Is there any way to keep /var/run/asterisk as asterisk.asterisk. Or do I find a new place to put asterisk.pid? sean

I wonder if I could have that patch, maybe I could add it to my fail2ban regexp and if you have the correct regexp, I would apperciate that as well. Thanks. On Wed, 29 Aug 2018 19:18:29 -0400, Telium Support Group wrote: > > Depending on log trolling (Asterisk security log) misses a lot, and also depends on the SIP/PJSIP folks to not change message structure (which has already happened

1.6.2: -- Executing [s at incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]:

OK, Thanks. I have a couple of questions -- the line numbers do not match exactly, so can you tell me a couple of lines before and after the line in question? Also, when will this be logged, if its only during sip debug, I need to change it to log when I can see it more readily. Thanks. On Wed, 29 Aug 2018 20:31:15 -0400, sean darcy wrote: > > On 08/29/2018 08:07 PM, John Covici wrote: