similar to: Securing Asterisk

Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg attempts I can't enforce the block until they drop and try again. This allows them to run the box

Hi list! Very strange... I ran the Asterisk CLI for other tasks, and suddenly I got this message: == Using SIP RTP CoS mark 5 -- Executing [000972592603325 at default:1] Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") in new stack == PROXY Call from 0123456 to 000972592603325 -- Executing [000972592603325 at default:2]

Hello Folks; Again, excuse my cluelessness. I have an Asterisk server in the US - and I want to connect it to one in Europe. Here is my scenario: 1. call a phone number, my Asterisk box in the US answers 2. perhaps a 'please wait' voice message 3. it dials an extension on the other Asterisk box in Europe. I am not looking for someone to do this for me, I am just not really

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

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

Hi, Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color in console. Do I miss a package or compilation option ? Thanks

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show <queue_name> I get the following numbers: <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the

Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

Hi all. I've got a perl script that connects to Asterisk's management interface using Asterisk::AMI. So far, its proven to be very useful. I'm hoping to use this to detect and respond to asterisk restarts and sip reloads. However, my script gets disconnected quite frequently, causing false alarms in my monitoring. Here's what the code looks like:

Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s at

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

How can I let asterisk immediately dials a trunk when off hook?

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

Hi Everyone, I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110107/eacb9f1a/attachment.htm>

Hi All, I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend. My dialplan: exten => _XXXX,1,Dial(SIP/${EXTEN},60,rt) Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument In 1.6 there was no problem, I have got Channel is

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only