Displaying 20 results from an estimated 4000 matches similar to: "SIP and permit specified ip addresses"
2020 Oct 11
2
Preparing for replication: dsync-local(testaccount): Panic: file mbox-lock.c
I am preparing a bit for setting up replication. However when I manually
try to dsync an account, the first time I execute this[1] command it
seems to be ok. The 2nd time I am getting this error[2].
If I add -1 (one way syncing) the error disappears. Does this mean I
will have problems with setting up replication between these two
servers?
[1]
[@ ~]# doveadm sync -n inbox -u testaccount
2004 Jun 29
3
smbpasswd !!?!
please help me.
Why I cannot create a user with smbpasswd without having this username in /etc/passwd???
###################
bash-2.05# smbpasswd -a testaccount
New SMB password:
Retype new SMB password:
Failed to initialise SAM_ACCOUNT for user testaccount.
Failed to modify password entry for user testaccount
bash-2.05#
##############
my global in smb.conf
[global]
workgroup = J9_C
server
2007 Sep 19
1
quota reporting and mail.app
Hi everyone.
I can't get mail.app to report back correct quota information using
mail.app and dovecot (though it works with courier-imap).
I am looking at migrating away from courier-imap to dovecot but this
issue is something I need to have resolved <somehow>.
Thunderbird is fine and reports back good info on either connection.
I am off to read RFCs (can't I just read the
2003 Nov 07
3
Unable dial out with the new Oh323 0.5.6
Hi all,
i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then
i've installed the new chan_oh323 (0.5.6).
when i try to make a call with "netmeeting" through * ( * dial out with
"Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked.
Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7)
installed, and it worked.
Is here
2003 Aug 29
1
additional digit in front of the dialed extenesion by outgoing pri/E1 call
Hi all,
i have configured incoming voip traffic as follows:
[voipin]
exten => _X.,1,SetCallerID(033283077734)
exten => _X.,2,Dial,Zap/g4/${EXTEN}
exten => _X.,3,Hangup
If the call going out the pri dials with an additional '0' before the dialed
number.
This is for caller number AND called number. But i can't see anything that
says set a '0' more in front of the
2020 Aug 29
1
401 Unauthorized when originating SIP user exists on remote server
Hi list!
I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC
behind NAT.
From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the
internet at 100.100.94.210 with a SIP account "3333" created in sip.conf:
[3333]
type=friend
secret=something
host=dynamic
nat=yes
qualify=no
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
context=voipin
I dial +1234
2014 Aug 28
2
Winbind + sernet Samba4 + CentOS 6.5 + AD
Hi y'all
So I'm having some issues trying to deploy this system and was hoping I
cold get some insights into this from some gurus
This is the situation:
OS: CentOS 6.5 (2.6.32-431.23.3.el6.x86_64)
AD: Windows Servers 2012 R2 running on domain functional level of Windows
Server 2008 R2
Samba4 installed from sernet yum repo. version 4.1.11-9.el6
As of now the centos server has been
2020 Oct 12
0
Preparing for replication: dsync-local(testaccount): Panic: file mbox-lock.c
<!doctype html>
<html>
<head>
<meta charset="UTF-8">
</head>
<body>
<div class="default-style">
Replication is not supported with mbox format. You can only do unidirectional sync out of mbox.
</div>
<div class="default-style">
<br>
</div>
<div
2023 May 05
0
Calls running forever / CDRs inaccurate
Hi list!
Running Asterisk 20.0.0 on CentOS 7, logging CDRs using
cdr_adaptive_odbc to mariadb-server-5.5.68 (via
mariadb-connector-odbc-3.1.7-ga-rhel7)
Using chan_sip.
I'm facing the problem when there is a sudden spike of calls, some of
the calls that are being made during those spikes hang forever
basically. This looks like this:
[root at voip]# asterisk -rx 'core show channels
2020 Oct 17
0
backup of namespace, is still looking at (touching?) other namespace?
When I am doing this:
doveadm backup -f -n inbox -F /root/backup-accounts.txt
tcp:mailxx.local:542
I am getting an error on the distributed storage, which I exactly did
not wanted to be touched.
doveadm(testaccount): Error: remote(mailxx.local:542): User
initialization failed: Namespace 'Archive/':
stat(/home/mail-archive/testaccount/Archive/mailboxes) failed:
Permission denied
2012 Aug 21
1
Asterisk 11 - XMPP and JabberSend()
I'm trying to get my Asterisk 11 test box set up with XMPP, having troubles with JabberSend().
My jabber.conf file is as follows:
[general]
debug=no
autoprune=no
[testaccount]
type=client
serverhost=my.jabber.server
username=myaccount at my.jabber.server
secret=mypassword
port=jabberport
usetls=yes
usesasl=yes
xmpp show connections gives the following output from the console:
2001 Jan 23
10
smbpasswd error
Hallo ,
when I try to change user password on samba server i become this error :
error connecting to 127.0.0.1:139 (Verbindungsaufbau abgelehnt)
unable to connect to SMB server on machine 127.0.0.1. Error was : code
0.
Failed to change password for bukhari
if any someone I have an Idea please send me an E-Mail to
"ufz6@rz.uni-karlsruhe.de"
Thanks to All
Amir Bukhari
2005 Oct 03
0
Hangup not detected on callback
Hi,
I'm trying to set up a call-back system using auto-dialout files. I
want the call to be terminated when a specific timeout (defined in the
.call file) is detected. Both parties should then be hangup.
The problem is that the timeout is never detected... How to solve this?
Thank you,
Pierre
.call file
----------
Channel: IAX2/:@xxx.xxx.xxx.xxx/0111111111
Callerid: 111111111
2016 Jan 28
2
Caller ID Sent in PAI header.
Hi All,
When receiving an invite containing two different caller ID, one in FROM
header and the other in "P-Asserted Identity" Header, Which one will be
used by the callee ? I couldn't find any RFC specifying this detail.
Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2015 Sep 21
2
Call waiting for Queue Agents.
Hi All,
I have a question about the Queues.
I'm using Asterisk 11.13.0 , and I want to configure the following setup :
When there is an incoming call to the queue all agents should ring even
those that are already in call, they should receive a second call.
Is this doable in any Asterisk version ?
Thanks in advance.
-------------- next part --------------
An HTML attachment was
2009 Jul 22
2
sip configuration masking the peers
Hi all,
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090722/30afa9ee/attachment.htm
2012 Aug 31
1
Receiving and processing unsolicited XMPP messages with Asterisk 11
I'm trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I'm only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the "call" hangs up without executing further instructions. Anything
2006 Jan 12
1
iproute problem
Hello,
I''m on Debian Sarge, and try actually to setup iproute this way :
- local network 1: 192.168.12.0/24
- local network 2: 172.20.0.0/16
- one router on both network : 192.168.12.50 & 172.20.201.50
- one router to internet (SLIS) : 172.20.1.1
I want from my 1st local network to access internet...
here are my lines... (taken from The LARTC Howto)
ip route add 172.20.0.0/16 dev
2003 Jun 10
3
s extension don't work on TDM40B
Hi all,
i have read in the * whitepaper the following:
"s: The "start" extension. A call which does not have digits associated with
it (for
example, a loopstart analog line) begins at the "s" extension."
I think this means the s extension will be execute when the phone is picked
up.
But when i pick up the phone the s extension will be never executed.
Whats wrong
2003 Sep 23
4
Dial over IAX ahngs up after 3 rings
Hi all,
can somebody explain this ?
Thanks,
Thomas.
*******************************************
beroNet technologies GmbH
Dipl.- Ing. Thomas H?ger
Potsdamer Str. 18 A
14513 Teltow
FON: +49 (0) 3328 3077731
FAX: +49 (0) 3328 334779
Email: thomas.haeger@beronet.com
*******************************************