Displaying 20 results from an estimated 100 matches similar to: "Re: Asterisk-Users Digest, Vol 21, Issue 113"
2003 Sep 19
1
SIP registration between *'s
Hi everybody,
I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae
In * one sip.conf
register =>usuario1:pass1@<public_ip_2>
In * two sip.conf
[usuario1]
type=friend
username=usuario1
secret=pass1
host=<public_ip_1>
dtmfmode=inband
Logs in * are the followings
In * one logs:
Sip
2012 Nov 20
1
FOOBAR\usuario1 windows explorer hungs forever while accessing shared dirs in LAPAZ\comp1 (interdomain trust relationships)
Hi all
I have two samba PDC installed according to these specifications:
domain FOOBAR with pdc server name: BAR (ip 192.168.1.1)
opensuse 11.1
samba-3.5.6-15.1
openldap2-2.4.12-5.6.1
smbldap-tools-0.9.5-25.1
A winxp called USUARIO1 joined to the FOOBAR domain (ip 192.168.1.100)
domain LAPAZ with pdc server name: SERVERLPZ (ip 192.168.10.4)
openSUSE 12.2
samba-3.6.7-48.12.1.i586
2005 Apr 15
1
The conflicting domain portions are not supported
Hi, maybe I didn't explained myself well.
What i meant is that the user can't have the SID
S-1-5-21-528226156-890416033-2029241632 but MUST have a sid like
S-1-5-21-528226156-890416033-2029241632-xxxx ( where x is usually assigned
automatically by the add user's script)
Best Regards,
Bruno Guerreiro
-----Original Message-----
From: Jos? M. Fandi?o [mailto:samba@fadesa.es]
Sent:
2008 May 20
5
Server recommendation help
I am having a issues with adding a analog card to my dell 2800. I
already have a t1 card installed and running fine but when I install the
analog card asterisk will not start (ztcfg fails). I have determined it
is because of a IRQ problem and have decided to get a new server. Can
anyone suggest a server grade setup that supports this? I would rather
not buy a machine that will be unstable. I
2006 Apr 28
1
Warning: No path to translate with SJPhone
Hi list!
I'm making tests for Asterisk. I've tested with 2 users installing SJphone
and it worked fine, but when I install it over a third user with the
softphone, the phone dial for 2 seconds and a window alert goes out on the
softphone:
Busy
Call rejected: 486 Busy Here
And on my Asterisk server this message:
Apr 28 09:05:37 WARNING[8140]: channel.c:2685 ast_channel_make_compatible:
2008 Jul 25
1
One LDAP attribute for many variables
Hi
Using Dovecot 1.1.2 with LDAP. If i have this:
user_attrs = uid=home=/var/vmail/%$,mailuserquota=quota_rule=*:storage=%$,=mail=maildir:/var/vmail/%n/Maildir
OK. But if i have this:
user_attrs = uid=home=/var/vmail/%$,mailuserquota=quota_rule=*:storage=%$,uid=mail=maildir:/var/vmail/%$/Maildir
I get this error:
deliver(usuario1): Jul 25 12:59:08 Error: Per-user script path is
unknown. See
2005 Apr 15
1
The conflicting domain portions are not supported for NETLOGON calls
Hello list,
When I try to log in a samba 3.0.13 server from a XP Pro
machine, I get this error:
[2005/04/15 10:57:00, 1] rpc_server/srv_netlog_nt.c:_net_sam_logon(766)
_net_sam_logon: user BETA\usuario1 has user sid S-1-5-21-528226156-890416033-2029241632
but group sid S-1-5-21-528226156-890416033-2029241632-513.
The conflicting domain portions are not supported for NETLOGON calls
What
2006 Jan 26
6
Fail over to Pri on VoIP connection failure
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
Any point in the right direction would be great
Thanks,
CLI output (cleansed to protect the
2006 Oct 16
7
tdm2400p question
Hi all,
I'm confused, in digium website, it says:
TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a
total of 24 lines.
6 plus 6 is 12, how come it's 24?
if I have 24 PSTN lines, i'll be needing 24 FXOs.
Pls. elaborate.
thanks.
Lito
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2009 Jun 12
1
SAMBA+PDC+Mysql authentication Backend
I ne w in samba world but i was configured a Samba with shares folder linkable to users and it was successfull.
Now i try to extend to PDC but the client can't logon into the server:
the log.smbd could this
[2009/06/12 15:51:21, 0] smbd/server.c:main(1209)
smbd version 3.2.3 started.
Copyright Andrew Tridgell and the Samba Team 1992-2008
[2009/06/12 15:51:21, 1]
2009 Jul 06
0
Iax trunk quality
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<div class="moz-text-flowed"
style="font-family: -moz-fixed; font-size: 13px;" lang="x-western">Hi,
<br>
<br>
I try to find a solution for this problem : <br>
2008 Feb 22
1
Message waiting light on polycom 301 using asterisk 1.4.14
All,
I am setting up asterisk on a nslu2 (Linksys) using unslug.
Everything is working great except that I have a polycom 301 and I
cannot get the message indicator to work. I have created the users and
mailbox in users.conf and I can manually dial the mailbox (*986000).
Last thing is I am not using config files for the polycom just web
browser.
Can anyone point me in the right direction
I
2005 Jul 13
3
Meet Me Configuration
I am trying to configure MeetMe so that external callers can enter the
conference rooms after an IVR menu. I have created Conf rooms for all
internal Ext's with a prefix of 8. When I call into the system from my
vonage trunck the IVR picks up but will not let me dial a conf room. It
tells me it is a invalid extension.
Can anyone help with a sample conf on this?
Thanks,
RC
2004 Apr 07
1
Out of trunk data space on call number 16386, dropping
Hi all,
We keep getting these and all the calls between these two asterisk boxes get
dropped. what is going on here, I have been trying to solve this problem on
my own but maybe I don't have the trunk setup right. also I have posed the
output of my full log of the machine with the zap interface, the other is
using ztdummy.
IAX.conf on machine 1:
[general]
port=5036
;iaxcompat=yes
2005 May 13
0
Problem with IAX trunking
Hi all,
I'm trying to get IAX2 trunking between two * boxes and am having
extreme difficulty :) What happens is when the sending * server (the one
initiating the call) receives the ACCEPT back from the receiving server
it immediately replies with INVAL. I've checked the code and it seems to
be not matching the accept packet with the relevant item in the iaxs
array due to the following
2005 Mar 21
4
Upcoming 3.0.13 release -- please test now
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Heads up everyone:
Due to the win98 explorer bug (https://bugzilla.samba.org/bug/2501),
we will be release 3.0.13 on Thursday morning, March 24 (GMT-6).
So if you have any outstanding bugs in the 3.0.12 that we
should know about, let us know now. Please file any defect
reports at https://bugzilla.samba.org/.
Thanks.
cheers, jerry
2003 Oct 21
0
Iitter Buffer Settings
I'm trying to come up with good jitterbuffer related settings for my
Asterisk boxes.
I ran 4 pings for about 2 days from my main Asterisk server to remote
Asterisk servers. During that time there were some large file uploads
which caused the max rtt to be quite large.
Here are the results:
pkts loss min avg max mdev
132013 %0 70.36 78.13 1967.37 36.04
132013 %0 98.95 120.46 2419.24 111.26
2004 Apr 06
3
Problems with IAX2?
Are there open problems/issues with iax2 and jitter (quality)?
Just upgraded to today's dev cvs about an hour ago, and it seems the iax
conversations are lower quality then a month or two ago. iax2 show firmware
says version 13. (Test call originated from C7960 with g711.)
Using the demo as an example,
iax2 show channels
Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter
2004 May 05
1
SIP Pick up groups
All,
I know the question has been asked before, but any of the solutions
posted in the past have not solved my problem.
I have got a Asterisk setup using a P4 1.8 / 512mb server running Redhat
Enterprise 3 and 3 grandstream budgetone phones (plus a couple of xten
clients on windows) and I'm at advanced stage of testing to see if
asterisk will fill our needs as a PBX using voice over IP
2004 May 18
0
Asterisk to IAXTel help
I'm trying to make a call from an IAXPhone client - through the * PBX
to an 888 number using the IAXTel link. I'm using the basic conf files
for extensions and iax. I get successfully connected (at the
"Attempting native bridge" line of the output) and am then able to talk
both ways for 20 to 30 seconds and then the IAX phone appears off line.
If I wait on the PSTN line for