similar to: SMTP Auth Spam Mail Attack

I''m working with a pretty simple config at the moment to track down some issues I''m seeing with custom facts and plugins. I have factsync = true (default locations) and I have a simple fact in /var/lib/puppet/facts/ (actually it''s David''s netmask plugin) my pp file I''m testing with contains only: notice "netmask: ${netmask}" puppet -v

Someone on the opennms mail list just mentioned that RHEL up2date now provides sun java. Is that a possibility for the Centos repositories? -- Les Mikesell lesmikesell at gmail.com

FreeMIS is an open source management information system for schools, built with Ruby on Rails. I haven''t come across any other big open source projects using Rails so far - if nothing else, the availability of the source code might be useful for learners looking for ideas? If anyone would be interested in helping to develop FreeMIS, please get in touch. Details here:

There are several changes where this distribution develops away from classic desktops: - maximized windows use panel instead of title bar - global menu - overlay scrollbars I guess the last two will stay as inconsistencies for a long time (if not someone manages to get the windows menu into the panel), but I thought that maximised windows shouldn't behave different than they used to. However

Programming Party all day on improving Asterisk, & getting a version running for the BTIP voice conference. Anyone: Please email me or the BTIP list if you know of any recent (past 12 months) Asterisk videos. Thanks. :) Join with the friendly, productive, Global FSW community, in the _TWICE_ monthly, Voice over internet meeting, BerkeleyTIP-Global. GNU(Linux) & BSD, Free SW, HW

Hi, I got a broadvoice "business" account under the byod(bring your own device) program. I have applied the patches and created new asterisk debian packages. I have the account working on inbound and outbound. The problem area is outbound caller ID. I have 3 other accounts with IAX2 providers and have no problem setting the caller ID on outbound calls. I called them and they

Using kmail with PLAIN authentication worked fine with beta3 but trying it with beta4 authentication fails. This is because kmail sends "username \0 username \0 password" in the authorization token and the new code to call auth_request_set_login_username() when supplied an authid must be returning failure (certainly commenting this code out returns to the beta3 behaviour of

Windows authentication extends far beyond the CIFS protocol the Samba implements, but it only very recently that work has been done to catch up to Microsoft's extensions in this area. This has caused many administrators pain and toil that their MS counterparts simply don't have. For them, authentication 'just works', with single-sign-on and the lot. I have worked, for over a

I went through the instructions on the Bell Labs wiki (http://www.cs.bell-labs.com/wiki/plan9/Installing_in_Xen/index.html), and am literally at the last step of the process, where I can create a new plan9 domain via xm. However, I get the following: $ xm create plan9 -c Using config file "/etc/xen/plan9". Started domain plan9, console on port 999 ************ REMOTE CONSOLE: CTRL-] TO

Hi Guys, Unoffical patch for mech-plain so the AUTH PLAIN command works with SQL (and possibly other DBs). Thunderbird should work again after this. --- mech-plain.c 2004-12-10 01:56:51.065987304 +0000 +++ mech-plain.c 2004-12-10 01:53:16.974534152 +0000 @@ -27,7 +27,10 @@ authid = (const char *) data; authenid = NULL; pass = ""; - count = 0; +

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone

I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob

>From - Received: from rwcrmhc12.comcast.net ([216.148.227.85]) by sccrmxc11.comcast.net (sccrmxc11) with ESMTP id <20031014185753s1100nos46e>; Tue, 14 Oct 2003 18:57:53 +0000 Received: from rwcrmhc12.comcast.net (localhost[127.0.0.1]) by comcast.net (rwcrmhc12) with ESMTP id <20031014185753014006qmtie>; Tue, 14 Oct 2003 18:57:53 +0000 From: Mail Delivery Subsystem

Hello! Oh, wise ones, ponder with me over two of the surprises that populate the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992

Hey, I''m an xmldb newb, as I expect most are since xquery only got nailed a few months ago. I know nothing more than reading around the subject and a couple of hours tinkering with sleepycats offering. Enough though to make me realise the potential of an xml database though. I dont really have a spare supercomputer to parse xml documents to compete with a relational database on speed but

A great December Solstice to you & yours. :) JOIN the Global All Free SW, HW, Culture meeting via VOIP Dec 20 Sunday, 12N-3PM (Pacific = UTC-8) = 3P-6P Eastern = 8P-11P UTC [Jan 2009 meetings: 2nd, 17th - mark your calendar] http://sites.google.com/site/berkeleytip/schedule == WATCH some VIDEOS: Mark Shuttleworth Interview - 10.04 Lucid Larynx Learning from Code History , Andreas Zeller

Hey everyone, CentOS has a booth this year at LUGRadio Live 2006, and we need some people to come hang out with us, help run the booth and tell the world what a great thing CentOS is. If you are in the UK this weekend, and fancy hanging out with some really interesting ( no, really, even geeks can be interesting ) people, come join us there. http://www.lugradio.org/live/2006/ >From the

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

It would appear that you have the Asterisk server on a public IP address, your two endpoints are behind a NAT, and you have rewrite_contact enabled in pjsip.conf. In which case, what you are seeing is correct. In order to be able to send a call to an extension where it is behind NAT, Asterisk must update the contact to have the current IP and port that the phone registered via (i.e. the WAN IP

Voice over Internet Protocol (VOIP) using Asterisk, Sameer Verma & work on the Programming Party to help get our own Asterisk VOIP conference server working. :) == Join with the friendly productive Global FreeSW HW & Culture community, in the TWICE monthly, Voice over internet Global Conference: BerkeleyTIP-Global: GNU(Linux), BSD, & All Free SW, HW, & Culture TIP = Talks,