similar to: sip phone...direct access...

Hello, I'm trying to Compile Samba with ADS support on Solaris 8. I have installed without any problems: /opt/cifs/bdb -> .bdb-4.2.52 /opt/cifs/heimdal -> .heimdal-0.6.3 /opt/cifs/openldap -> .openldap-2.1.25 /opt/cifs/openssl -> .openssl-0.9.7e /opt/cifs/samba -> .samba-3.0.9 and I'm configuring samba with: LDFLAGS="-L/opt/cifs/openldap/lib

Hi friends of SAMBA, I'm using SAMBA 3.0.0 on a Linux and SAMBA 3.0.6 on a Solaris server (my problems are independent from the SAMBA version, I guess). Following entries are existing in /etc/nsswitch.conf: .... passwd: files winbind group: files winbind ... "getent group" produces following output: ... domain1\group1:x:10000:<here is a list of 125.000 users>

Hi everyone, I lost my complete home directory and am facing the problem of retrieving some of the deleted data. I have search the web for this matter, but the only information I found is, that it's not possible for a program to do this and that I have to puzzle the files together by scanning the disk (or disk image) with tools like sleuthkit (www.sleuthkit.org) or lde (lde.sourceforge.net).

Hi, I have the following problem. I wanted to get a matrix of scatterplots and I used pairs. I wanted to add the line y=x in each plot and I created a panel function for this scope. I used points and abline in the following way: ## put y=x in each plot panel.lin<- function(x, y) { points(x,y, pch=21, bg=par("bg"), col = "black",cex=2)

hi, it'd be useul if the centosplus and contrib directory can contain a small readme file or some kind of documentaion about the rpm inside. currently there is not any kind of such docs (ok johnny send a description about the kernel rpms, but if someone missed that email). eg the postfix rpm has the same description and changelog inside the rpm as the original ones. if someone just look into

Hi all, I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the softclient. Any ideas? Thx! sip.conf [2005] type=friend callerid="OptiPoint" <2005>

Shalom! Okay, this is my wine config file, which i have copied out of the wine ini and then changed it to suite my computer. Im using SuSE 7.1 Professionell and have my fat32 partition mounted on /windows/C (<= capital). Im am using Win2000 and this is what wine says: linux:~ # wine --winver win2000 /windows/F/UO/client.exe Warning: no valid DOS drive found, check your configuration

Hi all, I have an optipoint 400 Sip phone with 2.46 sip image which stores its music on hold file localy (on the phone) rather than indicate to the server the call is on hold. i'd like to have Moh play from the server, and i'm not sure if there is a work around to achieve this anyone else has the same phone and came across similar problems ?? thanks fam

For all who are interested: A quick review of CeBIT 2005. :-) CeBIT was a very successfull event. Most of the time, the asterisk-booth was crowded with more people than we could talk to. We had with us a demo-installation including different IP-phones, digital and analog phones as well as a Siemens HiPATH PBX to which our Asterisk-server served as a VoIP-gateway, and many people were impressed

Salve, I'm somewhat stuck on how to get DTMF working with my setup and googling didn't yield anything similar. My setup consists of one CAPI-capable board (AVM Fritz!DSL) connected to a BRI (T-ISDN), one HFC-S board running in NT-mode connected to an internal S0 bus with some ISDN devices (DECT stations, TA) and, of course, some ethernet interfaces. ISDN standard used is Euro-ISDN.

Hi, ztdummy says the following: VoiceBOX:/usr/src/zaptel# modprobe ztdummy /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_register /lib/modules/2.4.18/misc/ztdummy.o: insmod

Hi there, I just finished work on my current PBX dialplan configuration. I didn't find anything compareable yet so I think it might be interesting for someone else too. Short overview: - seperation of the numbers in blocks (just definded) - error/cause handling - advanced access control (even vor ISDN Zap clients) - access levels - denylist with national number blocks (for Germany) -

Hello! I have been googling a lot and asked wiki a few times now, but i cant find a howto for setting up a voicebox. Any link/hint would be great! Thanks, Mario

Hello! I would like to call a number (e.g.35), and when i press a secret code (12345), it should jump to my voicebox menu. On this page http://www.voip-info.org/wiki-Asterisk+cmd+background i found something about "Silently Wait for DTMF Input". In my case it wouldn`t be silence. It woudl just play the away message. Now how can i include such a secret code to my background funktion? I

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

Thought I might just as well share these scripts, they may work with other phones too: ######################################################################## *1)* Dialing from the KDE 3.5.5 address book works with a script that gets triggered from the kaddressbook (Settings - Script Hooks - Phone) with my command snom_dial_number %N The script snom_dial_number itself goes like this:

hi list, I'm looking for a way to execute commands in my dialplan specifically when a caller has hung up. my curretn dialplan looks like this: exten => s,1,Answer exten => s,n(restart),BackGround(intro) exten => s,n,Read(Enter,4,4) exten => s,n,Voicemail(${Enter},u) exten => s,n,agi(process.php|${Enter}) exten => #,1,Playback(thanks) exten => #,n,Hangup it lets a user

I can't check the voicemail for the switchboard. Asterisk hangs up for some unknown reason... ----- s n i p ----- -- Executing [*500 at default:1] Wait("SIP/597-00f0c410", "1") in new stack -- Executing [*500 at default:2] VMAuthenticate("SIP/597-00f0c410", "500 at default|s") in new stack -- <SIP/597-00f0c410> Playing

Steve, The main problem I am having with the system is clipping off the start of someone's speech when they first start talking- the ends of the sentences seem to be handled properly. I am wondering whether this is the fault of the audio playback system or whether this is a speex issue- I also get the musical artifacts problem with the denoiser. This seems to be more of a problem on open

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike