similar to: SAMBA 3.0.7 domain member can't be browsed

Displaying 20 results from an estimated 300 matches similar to: "SAMBA 3.0.7 domain member can't be browsed"

2004 Sep 09
3
Store data from call to database
Hi, I use asterisk for a phone quiz game. I need to store data in a database (MySql, postgres) : telephone number, name (voice), ... and of course the answers at the quetions. What's the best way to store my data ? - script with system() command ? - AGI script - CDR - others ... Thanks Jerome Vous manquez d?espace pour stocker vos mails ? Yahoo! Mail vous offre
2004 Aug 23
2
[ Multiple drives ]
Hello, I have 3 hdd (120 GB, 120 GB and 80 GB) mounted on /data1 , /data2 and /data3. All these drives must be shared via a public access with Samba. For the moment, I can only share the 'data1' directory. [public] path = /data1 Is there a possibility to share several disks under the same account ? By example : [public] path = /data1, /data2, /data3 Then, under Windows, I'd like
2004 Nov 10
1
Samba BDC with LDAP support
Hi, PDC works fine, but Samba BDC doesn't make its job. In srvmgr.exe PDC, BDC appear, but when I kill smb PDC's process, normaly BDC may give a response to smb request. My problem... BDC do not respond, no PDC :: no authentification. any idea. my smb.conf : [global] # Main Config. netbios name = LYS workgroup = TNN server string = Lys (TNN's PDC) security = user domain
2004 Nov 24
1
Sip test
Hi all, Anybody would be able to call my voicemail just for test sip:infos@neos.yi.org regards harry Le nouveau Yahoo! Messenger est arriv? ! D?couvrez toutes les nouveaut?s pour dialoguer instantan?ment avec vos amis. A t?l?charger gratuitement sur http://fr.messenger.yahoo.com
2004 Sep 07
0
voip gateway connect to a pbx
Hi, I'm trying to set up a voip gateway between a classic pbx and ip network with asterisk. phones -- pbx -- * -- ip network I would like a prefix ( 0 ) for the classic calls and another prefix ( 1 ) for voip calls. The problem is that pbx can talk with asterisk only with S0 synchro (like a terminal) and succeeded not to make call with prefix in this mode. I also try to consider asterisk
2004 Sep 13
1
Read command without #
Hi, For my IVR, I use Read command. It works fine when ending bu # but I can't get anything without ending by # The wiki tell me is it possible with maxdigit option but it don't work for me. my command : exten => 3,1,Read(ILE,as/iles,1) Anybody can tell me howto do thanks Another question about read command: Howto sup file option and keep maxdigits options ? exten =>
2004 Nov 26
0
sip call test
Hi all, I wish to receive calls from anybody to sip:infos@neos.yi.org in order to test asterisk. Listen music and leave me a message. If you speak french send me a mail i'll give you an other sip URI to test voice quality. Sorry I don't speak english fluently. I use ddns so yours calls might failed if dns is not update or my computer is switched off . Thanks harry Vous
2004 Nov 27
0
Built-in Extension Numbers
hi all, I need help ! What are Built-in Extension Numbers ? if i dial *69 with callreturn=yes in zapata.conf i don't get the last caller . How may i use Built-in Extension Numbers ? I should not add extensions in dial plan !? Harry from voip-info.org: There are some "extension numbers" that are built into the Zap channel module. You may override these in your Dialplan, i.e.
2004 Nov 29
0
Problem when I call someone who is busy
Hi, My setup is quite complicated. I have to Asterisk server linked via IAX. My Sip phones are connected to one and go out (PSTN) via the IAX trunk and the other server is connected to a Quintum CMS via H323. Phone---(SIP)---Asterisk1---(IAX)---Asterisk2--(H323)---CMS--> PSTN All work fine but when a call someone who is busy I didn't hear the corresponding tone and asterisk2 go to
2004 Sep 28
20
Polycom IP500
Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm
2004 Jul 26
0
H323/Netmeeting shaping
Hi, Has anyone ever succedded in shaping H323 traffic ? I mean reserve a certain bandwidth for it, in order to have a comfortable Netmeeting, and not be disturbed by downloads & others. I tried with HTB but it doesn''t seem perfect... Thanks for replies, Sam Vous manquez d’espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail
2004 Nov 29
1
plot problem
Dear all, I am having trouble plotting a PCA result. The plot doesn't appear!!! R goes through without any errors but doesn't make a plot appear!! Could it be wrong window parameters? In this case how do I change them? I am under red hat 9 with the latest version of R! Thanks. Vous manquez d??espace pour stocker vos mails ?
2008 Oct 22
3
sip and nat
hi there, I 'm a newbie in "VOIP technologies" ; i 'm implementing asterisk and i 'm wonder what is the best way to resolving "the Asterisk/NAT problem" : some clients are behind a NAT. anyone could help me? thanks johanna _________________________________________________________________ Appelez vos amis de PC ? PC -- C'EST GRATUIT
2004 Oct 29
1
[rmetasim] Need help deciphering this error msg... targeted to those who use rmetasim...
Hello, I am trying to do some simulation using the rmetasim package and I've run to this problem. --beginning of error msg-- Error in "[<-"(`*tmp*`, slice[l, ], slice[l, ], value = c(0.200000002980232, : number of items to replace is not a multiple of replacement length --end of error msg-- Here is the script I used. --script starts here-- ## load 'rmetasim'
2007 Aug 23
3
[PHP-AGI] Problem executing script
Hello,I have succeded in compiling and configuring My TDM Card and asterisk, all works fine. But I have a problem using the PHP Agi.The CLI tells me that when I call my number :-- Starting simple switch on 'Zap/4-1' -- Executing [s at incoming:1] Answer("Zap/4-1", "") in new stack -- Executing [s at incoming:2] AGI("Zap/4-1", "rabot.agi") in
2003 Nov 14
1
What goodness-of-fit measure for robust regression ?
Hi, i. After estimating some coefficients using robust regression with rlm() or lqs(), I wonder if there exist some measures of the goodness-of-fit as those for standard linear model(R2)... or evenly if it's a statistics non-sense to look for since I do not find any mention of that in differents chapters on robust and resistant regression or in severals R documentation (Fox, Ripley and
2003 Apr 15
5
SIP support status
Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.
2008 Jan 30
0
Besoin d'un Financement rapide ?
[1]Si vous n'arrivez pas ? lire correctement ce message cliquez ici [2]Disponis le crdit selon vous [3][email-noel_02.gif] [4][email-noel_03.gif] [5][email-noel_04.gif] [6][email-noel_05.gif] [7][email-noel_06.gif]
2006 Jun 07
1
asterisk-1.2.9 / res_snmp.so
--- hgaillac-sip@yahoo.fr a ?crit : > hello, > > How asterisk could support res_snmp even this module > don't help to monitor all asterisk features? > > monitoring asterisk with snmp would be a good > thing. > Which solution ? > > Harry > --- Kristian Kielhofner <kris@krisk.org> a ?crit : > > > hgaillac-sip@yahoo.fr wrote: > > > I
2006 Jun 08
3
dial pattern
Hello, I have to dial prefix 9 for non local numbers however when i missed calls i Can't redial this number because of "9" is not append . I use polycom phones . What Can i do ? Harry __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicit?s