similar to: Spam and that recent 'attack' ...

2009/5/20 Gordon Henderson <gordon+asterisk at drogon.net<gordon%2Basterisk at drogon.net> > > On Tue, 19 May 2009, Olivier wrote: > > > Hi, > > > > In ASTDB, I've got a rather long list of entries like: > > /FamilyA/Key1 Value1 > > /FamilyA/Key2 Value2 > > /FamilyA/Key3 Value3 > > ... > > > > Instead of sending

2014-08-01 16:51 GMT-03:00 Keno Fischer <kfischer at college.harvard.edu>: > Hmm, that's intersting. I guess the difference might be that you're > targetting 32bit and I'm targetting 64bit. Does it still work if you > replace > "s/^.* T _\([^.][^.]*\)$$/\1/p" > by > "s/^.* T \([^.][^.]*\)$$/\1/p" > in > tools/llvm-shlib (and similarly

> > Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes

Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've

I've just posted this to another list where we were talking about the same old issues we've been plagues with recently - I'd already posted some iptables rules, but added more to it for this... This script probably isn't compatable with anything else, but I don't run anything else. It's also designed to act on the incoming interface, not to run in a router, but

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

Hi All I have a server which seems to be getting spam relayed through it. The story is this..... User reported loads of undeliverables being received so I had a trawl through the logs. So the attacker connects to our server using SMTP AUTH........ Oct 5 15:17:53 www sendmail[6972]: AUTH=server, relay=pppoe9.net109-120-27.se1.omkc.ru [109.120.27.9] (may be forged), authid=jon, mech=LOGIN,

Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on 1.4.. Nothing special about the hardware - older TDM400 card, 2 red modules fitted... Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels still work OK, but only for one line - the 2nd line causes it to refuse to dial-out no matter which port it's plugged into. The Lines are bog-standard BT

This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that

Hello - I would like to figure out a way to have backgroundrb start/stop with the same UID that dispatch.fcgi uses, at the same point in time that apache is started/stopped. I see a technique that seems plausable in this archive message: http://rubyforge.org/pipermail/backgroundrb-devel/2007-February/000723.html which addresses the starting aspect, in a lazy fashion. That seems fine,

I've spent all day today trying to fix an echo problem and I've made no ground whatsoever. I have Digium TDM400 with 3 FXO & 1 FXS. I've tried this computer at two completely different sites with different phone providers. I've tried compiling & installing different versions of Zaptel (currently running 1.4.0, started at 1.2.9 and worked my way up). I've tried

Hello everyone, I'm a student at the Universtiy of Delaware, and will be soon starting some research on the effects of lossy audio compression on speech sounds. I will be preforming test with both mp3 and vorbis. First of all, if I use the '--ogg' switch to lame, does lame use GPSYCHO to encode the wave, or some other psychoacoustic model (perhaps one designed for

so last night all my servers were severely probed and they tried to violate me (lol) the attack was so egregious I decided to contact the isp for that ip. Telepacific. The ip has some google searches that point to a few spam and a few attacks...So i assume a compromised server. So I sent them the info and said it must be a hacked server (the ip is on their business network) they responded

Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who

The cell companies are "doing it" like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give "4-10 rings" of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but

The cell companies are "doing it" like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give "4-10 rings" of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but

On Wed, 13 Dec 2006, Wireless wrote: > Hi Gordon > > I too have this problem with one of my two BT lines, very very annoying. I > was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM), > I think the sangoma is very slightly better (less echo) but I might just be > kidding myself. Another think I've found it that using a faster CPU (now on > a 450

http://bugs.freedesktop.org/show_bug.cgi?id=23058 Summary: Intersting info with the Nouveau driver Product: xorg Version: git Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org

Probably not many of you have noticed, since I assume you have your own active spam filters: But we have recently (first time on Friday April 15) had problems with spam filtering on our mail server. The spamassassin daemon (spamd) has "died" for no apparent reason, and hence the mail has been passed through without spam filtering. We have had a look at the log files and haven't got

Due to the large volume of spam on the list (and that stupid anti-spam spam - I can't believe how lame that is) I've blocked postings to the list that are only in text/html format. Multipart postings are still allowed although I wish they weren't. There have been problems with the mime filter breaking other people's mailers so that project is on hold for now. Tim.