similar to: Re: Antivirus for Mac Wine?

as for me I use ProteMac NetMine ( www.protemac.com )

check your firewall settings or change your firewall to more reasonable one, that doesnt block ordinary processes i prefer http://www.protemac.com/netmine/

On Mac using Wine trying to download database from FTP site for Troopmaster (Windows application) Application opens, I have upgraded program. I have fixed internet provider opening Port for use through modem. I believe I might need to add program to a list for firewall. On the Mac side, I can't see application (TroopMaster.EXE) to add to Firewall in Security. On the wine side I don't know

hi all, i would like to ask some questions regarding best practices for zfs recovery if disk errors occur. currently i have zfs boot (nv62) and the following setup: 2 si3224 controllers (each 4 sata disks) 8 sata disks, same size, same type i have two pools: a) rootpool b) datapool the rootpool is a mirrored pool, where every disk has a slice (the s0, which is 5 % of the whole disk) and this

From: Rodrigo Barbosa <rodrigob at suespammers.org> > Btw, don't quote me on this one :) > I'm only 90% sure of the hotswapping capabilities, and less than 50% > sure about the price :) There _are_ systems with hot-swap CPUs, memory and/or, PCI[-X] slots. They are _not_ commodity and pricey, and require OS-level support. In fact, I believe Linux 2.6 has some support for

I have 3 disks that I want to setup in a raid 1, my questions is should I setup the 3rd disk as a spare-disk or should I mirror all 3 disks all the time? Will it perform *much* better to only mirror two and have the 3rd as a spare. Is it not safer to mirror all 3 drives all the time ?

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

Hi, For a few weeks now, our asterisk server has been experiencing something very odd. From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707

From: Chris Mauritz <chrism at imntv.com> > <snipped> > Oy....I'm terribly sorry. Moderator, please delete my previous > message. It was supposed to be a private message to Peter. Bryan, > please accept my apology. It wasn't my intention to riducule you in > public. honest injun.... > <blush> > Must drink less coffee.... No apology required,

I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are

Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour

I am having difficulty with the calculation of the power law exponent for set of nodes within a graph. Specifically, I am interested in the distribution of in-degree and out-degree among communities of web pages where the web pages are the nodes of the graph and the hyperlinks the edges. According to the literature, the distribution of incoming and outgoing links obeys a power law distribution

Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get "May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:

Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees. Anyone got any skeletons on them? Thanks Alan -- The way out is open!

Hi, Anyone know what the settings in SIPDefault.cnf should be for Cisco 7940 phones this year? Came in today to find they'd all moved one hour ahead (NTP server is correct and ok). Found the "day" was set to "26", but on trying to change the settings to the below, my test phone isn't changing back: dst_start_month: March ; Month in which DST starts dst_start_day:

All, I'm trying to configure queue agents w/ a DUNDi setup so that an agent can login to whatever server they please w/o any custom setup. In general this seems to work, agents login w/ AgentCallbackLogin into the incoming context (not a special queue context) and can receive queue calls. The problem is that since the incoming context is the same context as the normal incoming call context,

I am trying to build a simple queue for the receptionist phone. In other words, there is only 1 agent and that is the receptionist phone. I just defined a few lines in queues.conf [console] strategy = ringall member => SIP/4000 ;4000 is the console extension In extensions.conf, it is: exten => 4000,1,Answer() exten => 4000,n,Queue(console) exten => 4000,n,HangUp() I pressed

Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

I've a big problem with SIP forwarding back into 'ringing groups' creating what can only be described as call storms :-( I have a 'ringing groups' of SIP phones with an effective dialplan (much simplified) like so: ; Purchase ledger [ptsn_inbound] exten => _846061,1,Dial(Local/6061 at groups) .... [groups] exten =>