similar to: SIP CALL ENCRYPTION

> On Thu, May 28, 2009 at 02:00:15PM -0500, research at businesstz.com wrote: >> Hello >> >> May i please know if asterisk is now supporting sip call encryption. It >> has been a requirement from one of my client to ensure that all >> conversation is well secured from any potential sniffers or inside >> hackers >> >> I have reviewed and shall

Thanks. The configure run successfully. but I got the warning below.. checking for the ability of -lsrtp to be linked in a shared object... no configure: WARNING: *** configure: WARNING: *** libsrtp could not be linked as a shared object. configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr configure:

Hi, I'm trying to set up a "secure" VoIP channel between a Windows softphone client and an Asterisk 1.6... server running with OpenBSD. By "secure" I mean to prevent any man in the middle to reconstitute any vocal exchange nor sender/addressee/any header data/ of the VoIP call (in first step, I would be glad to secure vocal data ans see later for the header...) I had a

hi all, can someone help? I have centos 6.8 trying to install asterisk 14.3.0-rc2 on it with options as stated below - ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib --with-jansson=/ --with-pjproject-bundled when I tried to run "make menuselect". i get the error below. Makefile:109: makeopts: No such file or directory **** **** The configure script must be executed

Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam -------------- next part

Is there a way to restrict connection to my asterisk server to users based on their IP addresses, and not just password. I have some hackers who connect to my server to make illegitimate solicitation calls to people. I had to shutdown the server for now until I find a solution. ANY HELP? Thanks. ond -------------- next part -------------- An HTML attachment was scrubbed... URL:

>>> Perhaps if there was a Asterisk RBL we could all contribute to; for >>> which we could then hook into and drop any connection where a >>> source IP is listed ? -- Thanks, Phil >>> >> >> I love the idea of a RBL... count me in for contributing. >> >> Especially considering the ridiculous response I received from >> Amazon.

Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in "maxretry" in jail.conf For example, I get an email saying: "The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK." when "maxretry = 5" in jail.conf Perhaps someone else is experiencing this or has resolved it,

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This isn't asterisk related but I figure several developers on this list have built apps for Twitter (or other 3rd party API's). Just found out a few hours ago I'm being sued by Twitter Feel free to tweet this link ( www.MyTwitterButler.com/I'm_Being_Sued <http://www.mytwitterbutler.com/I'm_Being_Sued> ) or forward on the link to any journalists you know.

Hello again list, I'm running a 1.4.42 install on SUSE with an AEX410P card. The DAHDI release is 2.4.0 because the machine won't properly install 2.5 and also won't install Asterisk 10.0 because I can't get a good SQLite3 library to install. Whenever I enter DTMF very quickly or very slowly, app_read des on me. Has anyone experienced similar joy

Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up Asterisk (with -cvvv) I get an error regarding func_curl.so (lines omitted) ... == Registered custom function STRFTIME == Registered custom function STRPTIME == Registered custom function EVAL == Registered custom function KEYPADHASH == Registered custom function SPRINTF func_strings.so => (String handling

Hello Friends, I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86 version. and here is snap of uname- a command *Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200 x86_64 x86_64 x86_64 GNU/Linux* when I try to run DAHDI distribution dahdi-linux-2.1.0.4 I am getting following error *echo "You do not appear to have the sources for the

Hi All; If I need to set the opermode to King Saudi Arabia, what the name I have to use? For example, to set it for kuwait then I use opermode=KUWAIT. So what will be for Saudi Arabia? Any advise? Regards Bilal

Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev

Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No

I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? Best Regards, -- Guilherme Loch G?es

I have a server with 8 Xorcom Astribank, they're connected to the server via a USB hub, my problem is: when a reboot the server the ID of the Channel Bank changes, cousing a big mess on my server. Anybody know how can I solve this ? Best regards, -- Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre:

We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the

Hello List, I am getting Asterisk set up. I am going to be installing Debian Linux on a laptop later. I would appreciate some detailed instructions on: (A) What to type into the shell to download and install Asterisk. (B) How to open the configuration files (*.conf) (C) If there is a way that I can change the configuration files remotely (SSH?). Thanks in advance. -- -Shane Blog: