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Hello. My setup: _ one Samba 3.5 domain (XXXXXXXX), with a PDC and a BDC, both running FreeBSD; _ one AD domain (YYYYYYYY) running on two Windows 2003 DCs; _ bidirectional trust between the two domains. Everything used to work until I moved the PDC from Samba 3.5 (EOL'ed) to 3.6; now, users from domain YYYYYYYY cannot access the PDC's shares. I used to have in smb.conf: >

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a "Call Disconnected" 486 error. What's odd is that I

Hello All, I am having a problem with samba's ability to interpret certain files with windows 2k clients. Setup Details: Linux ( RedHat 7.3 ) File Server Win 2k Clients Samba v2.2.3a Kernel 2.4.18 statically compiled. Problem Description: When copying jpegs from a Win2k client which is displaying the JPEGS in thumbnail mode, to the Linux / Samba file server, 3 files are copied as

Dear R-WinEdit users, I got a simple question, but somehow I cannot find the answer, although I have tried a lot! I got an ASCII-file and I want to import it into R, so that every character is defined by [i;j]. The rows are not of the same length. the file looks like the following shortened abstract example: name: xxxxx xxxx age: 9.9.99 record number: 999 title: xxxxx xxxx xxx keywords: xxx xx

Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone

Hi all We have an Asterisk server that?s been running for a few years now without problems. We have IPTables running, as well as fail2ban and have followed all the security recommendations we have found. Every few weeks we get an attack that lasts about a minute or two, resulting in our AGI script being overloaded. What happens is that somebody seems to be trying to connect from our

Hi to all, I and using asterisk with following 1. TDM400p card with four FXS modules, So there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 But when some dials form his number (suppose abc) to my number (suppose xxxx) I get abc number on my analog phone, but now I have purchased more than one numbers

Hi, I'm running our local school's mail server on CentOS 7, Postfix and Dovecot. We get quite a lot of spam, so I have the following sender restrictions in my /etc/postfix/main.cf: --8<------------------------------------------------------ # Restrictions SMTP smtpd_helo_restrictions = reject_unknown_helo_hostname smtpd_sender_restrictions = reject_unknown_sender_domain,

Hi, This is probably a bit OT, but here goes. I've been running our local school's mail server since 2013, with mail addresses for school staff and some teachers. The server is running CentOS 7 with Postfix and Dovecot, and it's a nice no-bullshit configuration with SPF, DKIM and DMARC. The school sends quite a lot of email out to parents, and sometimes, mail gets rejected:

Hi all, I am having a problem with mysql. I set up root password set password for root at localhost=password('xxxxxxxxx'); set password for root@'xxx.yyyy.com'=password('yyyyyyyy'); after that I tried to log in, but mysql says ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: YES) I guess that I had the right password. My

Hi! I have to Set up an asterisk Server with a Diva Server PRI E1-30M. Capi, asterisk, etc. everythink works. my problem is the handling of the MSN's. say, we have the block (without area-code..) 4321-0 to 4321-4999 between this numbers (including em) every MSN is possible. do I have to add all MSN's i need (several hundrets) to the capi.conf? then the routing to SIP-Phones shall be

I'm trying to get two * boxes to talk.... no matter what variation I try I get No Authority Found and connection refused from 192.168.1.5 I've googled, I've site searched.... to no avail. Here is the server a configs (192.168.1.5): iax.conf [general] port=5036 bandwidth=low disallow=all allow=gsm jitterbuffer=yes tos=lowdelay register => pbx:test@192.168.2.2 [pbx] type=peer

On Thu, 8 Oct 2020 at 17:50, Nicolas Kovacs <info at microlinux.fr> wrote: > Hi, > > This is probably a bit OT, but here goes. > > I've been running our local school's mail server since 2013, with mail > addresses for school staff and some teachers. The server is running CentOS > 7 > with Postfix and Dovecot, and it's a nice no-bullshit configuration with

hi, it turns out that the iksemel library (which i installed using an rpm) was returning 0 when the function iks_has_tls() was called. it should return 1 otherwise res_jabber.o thinks gnuTLS is not installed. i confirmed this by running a test program i wrote, that calls iks_has_tls . it returned 0. i downloaded iksemel source, compiled it and now the test program returned 1. now,

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

Hi. I've just bought SIP telephony service from a Swedish telco. I've managed to make and receive calls with kphone. Now I want to set up asterisk to be able to add fancy features like voice mail and recording conversations. But first I have to get the basic setup right. I'm running asterisk and kphone on the same machine, behind at NAT-router. When I make a call (from my regular

We have dropped dovecot in place of UW-IMAP for about 40000 users and have solved our immediate problem: disk I/O overload, mainly from people doing POP3 and "leave mail on server". Thank you! Now i got loads of dovecot error messages in the logs and don't know if there is any significance. Here are a few examples: The most common event is: 2007-12-06 16:00:17.837654500 dovecot:

Forgive the repost, but I've changed the subject line since I haven't seen a reply yet...please help, I don't have much hair left! Hello All, I had been successfully running Samba 2.2.8a on a FreeBSD 4.7 box for a couple years using roaming profiles. The box was functioning as PDC. The hardware was getting old; I needed to migrate to a new server. So, I built a FreeBSD 5.3-RELEASE

Hi, Using a couple of sip phones and using asterisk to connect them to a single sipgate.de account. if I call a mobile I have no problem makeing conversions. If the mobile rejects the call (by pressing hangup while it rings), something strange happens: the following is seen in the logfile, everytime a rejected mobile call happens: ----------------- Mar 20 22:52:29 WARNING[4682]: Forbidden