similar to: Somebody known how to disable concurrent access on mailbox?

Hi, I have a linux box loaded with RH7.1, 3 ethernet with load balance on 2 ISP. In theory, if one router stop to work (I have turn off for this try), the Dead Gateway Detection, would re-instradate alla traffic on other link. This not work...someone have more experience on this? Giorgio Biondi

Hi, I have simple question about Skype. What are the methods of selecting packets which belongs to Skype?? I know about 7layer but I don''t belive that is only way. Is 7layer realy good and stable solution for routers which must handle more than 1000 users ? Thanks in advance Pozdrawiam Szymon Turkiewicz

I have a file server with two shares accessible to 2 different groups. After the last update ( from debian 2:3.2.5-4 to 2:3.3.2-1 ) i cannot any more access ONLY ONE of the two shares and I can't understand the reason! Can anyone hel me? I'm getting mad! Thanks Giorgio from smb.conf: [documenti_movi] path = /home/documenti_movi valid users = @staffmovi read

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes.... Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio.

Hi all, I’m exploring puppet for first time, trying to see if I can simplify my sysadmin tasks. So I decided to install it on FreeBSD 7 beta 3 and play with it, using another host as a server (Debian 4). As you can imagine I’m stuck. Here’s an excerpt of my super trivial manifest file: > node ''puppetclient.refactor.it'' { > package { > apache20: > ensure

I would like to perform a bootstrap validation of a backward stepwise logistic regression analysis, but I am a beginner with R and I am not sure of how to do it. Is there anyone that can send me a sample file in tab format (that I can modify in Excel by pasting my data) and the pertinent R algorithm? Many thanks Giuseppe -- ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Dr. Giuseppe Biondi

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found

Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __________Asterisk fxo ---- line -----| -----------------Analog phone The analog phone rings immediately when calling, while asterisk shows the message

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

Hi, Just thought I would let the list know of a new release of RubyStack for Windows, since there is a lot of overlap in functionality and may be of interest to some of you also using OS X and Linux. Please let me know if it is inappropriate. Like InstantRails, RubyStack is an open source, self-contained distribution of Ruby on Rails that is easy to install and use. It runs on Windows, Linux and

Hi, I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) using gcc 4.0.2. Compilation does not give me errors so after a 'make install' I try to load zaptel module with insmod but the following error arise: *insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio --

Hi all, When I do a test call into the box (which is running latest version of Trixbox) it all goes fine. If i decide to hangup the cellphone (during the ivr playing options) the system does not recognize the hangup and the system continues through and ends up at the timeout option. What settings do I need to change to fix this. Is it the rxgain? If so is there something i can use to figure

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hello list, using asterisk 1.4.30 I have the strangest problem that some SIP accounts can register to my Asterisk and others not. I see no connection between all those that can register or all those that can't. It's not a firewall problem as all register to port 5060 and the range 5060 --> 5064 is open. It's just very strange that some can register and other not. Any