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...+972-50-7952406 mailto:tzafrir.cohen at xorcom.com > http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir > > > > ------------------------------ > > Message: 18 > Date: Fri, 10 Apr 2009 10:33:36 +0100 (BST) > From: Gordon Henderson <gordon+asterisk at drogon.net<gordon%2Basterisk at drogon.net> > > > Subject: Re: [asterisk-users] Can Asterisk bridge between a SIP client > and a Cisco Call Manager server? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> >...

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

Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Does anyone (other than cynical old me) think that Enum is a spammers best friend? Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) I can see that

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

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 > >...

...rom a Chinese manufacturing company trying to sell me the usual VoIP stuff - phones, etc. (From com-vox.com FWIW). However as well as targetting the email address on my web site, they also targetted a made-up, but plausable, email address at the host that was attacked last weekend - (systems at drogon.net and systems at xxxxx.drogon.net) and that host has no public mentions anywhere that I'm aware of (Although once upon a time I did publish a SIP URI to it, but that was removed well over a year ago). Also that host doesn't run email and has no MX records pointing to it. So it's...

Hey Everyone, I would like to start testing/playing with PRI channels but I don't have access to a PRI line. Is it possible to do the equivilent of a crossover between two PRI Cards (say Digium's TE120P)? What I was thinking is that I could set one asterisk box up with a PRI card set as the TE and provide clocking and another box exactly the same but with the card setup as NT. I think

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

So I've been testing asterisk under LXC for a few days now and am very happy with the results. My test server is a 1.8GHz Celeron with 256KB cache and 512MB RAM and I have 20 containers each running asterisk (and apache/php,sendmail and a few other minor things) More for fun than anything else, I've tried daisy-chaining instances together - so 20 asterisks running on the same host, 0

Hy guys i am having so much hard time to setup asterisk on a virtual machine that i got , i just want to know if i really need to use Dahdi and libpri on a complete Digital PBX i just gonna use sip and iax. I will never use any kind of analog line on this machine. Wait for a feed back. Daniel Abreu.

I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the

...- I think SUBSCRIBE requests are mostly harmless... And then there's other protocols - e.g. IAX - I'll have a look at the data format for IAX and see if I can do the same for that too - although I haven't heard of any slamming type attacks on IAX - yet... The file is at http://unicorn.drogon.net/firewall2 Please don't just run it without having a look through it though - you might find yourself locked out of your own system! The FTP port ranges are set to match those I use in proftpd, so you might have to change those if using FTP and/or other FTP servers... The timing stuff can...

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

Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring

...so i have written IVR apps that hook into our CRM and Accounting systems for fault reporting and credit card payments etc. What you need is the tool for the job like everything in life :-) Jon Farmer Telford, Shropshire, UK ----- Original Message ---- From: Gordon Henderson <gordon+asterisk@drogon.net> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Sent: Wednesday, 31 January, 2007 12:20:11 AM Subject: [asterisk-users] Dialplan programming vs. AGI vs. ??? Just a general question on dialplan programming... I've implemented a fairl...

Anyone know how to change the From field in Asterisk PBX voicemail/email to some other info of my choosing? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070605/2b083cbd/attachment.htm

Anyone recommend anyone who can provide me (actually a customer, but I'm asking on their behalf) DIDs in Sweden? They already have an asterisk box (in Sweden), now want a local number for it! Thanks, Gordon

http://www.theregister.co.uk/2009/02/11/fugitive_voip_hacker_arrested/ Gordon

It looks like something has changed in the HPET kernel code in 2.6.28 (maybe .27 too) that's stopped ztdummy.c compiling (in 1.2 and 1.4 versions of zapata) A kernel structure member has been renamed with some crypic comments in the lkml about it. Anyone know the right thing to do - I'm not up on the linux kernel guts, nor how ztdummy might interact with it, so simply renaming the

Hi, I've just observed something a bit odd - I'm wondering if this is the expected behaviour, a bug/feature, or something I'm doing stupid! 1st person gets into MeetMe. Nothing fancy, just: exten => 987,1,MeetMe(400,iM) They enter the passcode and their name, then listen to MoH. So-far so good. Now the 2nd person dials in. They enter the pin-code, and at that point, the MoH