similar to: ADS mapping not working after patching

Displaying 20 results from an estimated 900 matches similar to: "ADS mapping not working after patching"

2003 Dec 01
0
No subject
BDC to a SAMBA PDC. Did I read this correctly? Has anything changed since then. If it can't be done, can anyone recommend another solution that has worked for them? I also would like it if SAMBA, Courier, Postfix could share the same username/password file. All of the above I mentioned (According to my reading) support the reading of /etc/passwd, so why would one want to use PAM or LDAP
2003 Nov 17
0
RE: Asterisk-Users digest, Vol 1 #1918 - 9 msgs
An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto:
2007 Jun 15
0
No subject
using Asterisk. =20 Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are you looking to have it replace your existing telephony equipment? =20 As a suggestion if you google Trixbox and Nerd Vittles you will find a fairly detailed explanation of how to set your Trixbox server (a version of Asterisk) up to
2002 Aug 04
0
Re:samba digest, Vol 1 #1482 - 16 msgs
I am out of the office until Tuesday Aug 6th. If you need immediate attention please contact Don Jemtrud at don@magnumlog.com or call 297-6205. >>> "samba@lists.samba.org" 08/04/02 08:26 >>> Send samba mailing list submissions to samba@lists.samba.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.samba.org/mailman/listinfo/samba or, via
2004 May 04
2
Can Asterisk support R2 signaling
Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl >From: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs >Date: Tue, 04 May 2004 13:32:00 -0500 > >Send
2011 Mar 21
4
Multi-Tenant Hosted PBX system with Reseller functionality
2005 Jan 18
5
Open Source QoS .
My router (1605R) currently does not support QoS. Is there any open source software available so that I can set one up before the router? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/38b6e3f2/attachment.htm
2003 Aug 05
0
The American Resolve Prayer and Tribute Poster (PR#3662)
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2003 Jun 11
0
(no subject)
<P>Hi everybody</P> <P>I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! </P> <P>1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't
2013 May 08
4
lme4 y residuales
Buenos días; Estoy intentando aprender algo de sobre los modelos lineales mixtos. Estoy siguiendo el libro "Multilevlel analysis: techniques and applications" de Joop Hox. Estoy intentado reproducir las tablas y gráficos que trae el libro con el paquete lmer. En este libro se indica que en la comprobación de los supuestos del modelo hay que estudiar los gráficos de residuales tanto
2010 Oct 06
2
text() background
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2010 Sep 06
1
Correct coefficients from treatment contrasts?
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2010 Jul 28
1
Help with specifiying random effects in lmer - psychology experiment
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2005 Feb 11
6
i want to load chan_h323.so
2009 Jul 20
0
No subject
=20 arp | grep "192.168.0.1" =20 substituting the IP address of the SIP device. =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. =20 hello, is
2010 Dec 20
0
What's up?
Are you making progress?<DIV> <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt" align=center><SPAN lang=EN-US><FONT face="Comic Sans MS" color=#002060 size=3>Dear Friend,</FONT></SPAN></P> <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt" align=center><SPAN lang=EN-US><?xml:namespace prefix = o ns =
2007 Jul 12
0
No subject
community there is a real possibility this may come off so if you have an interest in this space and want to contribute to the discussion then this is your opportunity to do so. =20 I look forward to all opnions on this topic. =20 The slide deck for the agenda of this call is located here http://voipusersconference.org/2008-05-09-Slides=20 Cheers, Dean=20 ________________________________
2009 Aug 09
0
Killing the Server
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2011 Apr 12
0
No subject
a phone system, and plug it into a SIP Adapter like the PAP2T. Never done it myself, so I can't recommend a suitable intercom. Hopefully s= omeone else can. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<h= ttp://www.keshercommunications.com/hostedpbx.html>
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")