similar to: Someone has hacked into our system

Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary

Hi, I wonder if anyone has any suggestions on how to setup a network to run VPN over wireless. I currently have: Wireless Laptop ----> Router with VPN pass through ---> DSL modem. There is also a wired desktop running Win '98se connected to the router. The Wireless Laptop is running XP Pro sp1. I am open for suggestions on how to run VPN over the wireless. I just want to protect

Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented about RE: [asterisk-users] Configuring Softphone: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz > Sent: Wednesday, December 08, 2010 1:27 PM > To: Asterisk

I'd like to find out how to block everyone outside of the our LAN. The following phone call got through: 1. accountcode: Blank 2. src: Caller*ID number Blank 3. dst: Destination extension 901185294464086 4. dcontext: Destination context DLPN_DialPlan1 5. clid: Caller*ID with text Blank 6. channel: Channel used SIP/xxx-088c48d8 7. dstchannel: Destination channel

Hi, I'm trying to get a softphone configured. In Sip.conf [general] I found an example that said I need: nat=yes localnet=192.168.xxx.xxx Is localnet supposed to be a LAN IP or a WAN IP? Thank you, Gary

I see in sip debug it says Audio is at port 10342 Express Talk suggests Audio at UDP 8000-8020 I've tried changing Express Talk to 10000 and forwarded 10000-10400. Is there a possibility Express Talk won't work in the 10000 range? Is it possible to limit Asterisk to 8000-8020? Thank you, Gary

I have been told that my logic in extentions.conf is wrong in trying to configure a SIP software phone called Express Talk (remote) . I'd like to make outgoing calls and calls to local extensions. Could someone please look at my configuration files at http://pastebin.com/ajp62wqF and see what I did wrong? Thank you, Gary

I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user

I currently have in extensions.conf: exten => 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,n,Monitor(wav,${CALLFILENAME},m) exten => 106,hint,SIP/106 exten => 106,Macro(stdexten,106,${HINT}) When I called x106 this was logged: -- Executing [106 at voicemenu-custom-4:1] Set("DAHDI/7-1", "CALLFILENAME=_xxxxxxx") in new stack --

I had lines 3 and 4 and added line 1 and 2 to extensions.conf exten => 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,2,Monitor(wav,${CALLFILENAME},m) exten => 106,3,hint,SIP/106 exten => 106,4,Macro(stdexten,106,${HINT}) I received this warning: WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for extension

I would like to do something like: exten => s,1,playback(http://my.server.com/file.wav) I tested and it does not work. It seems highly likely that someone would already have done this one way or another. I know I could do a system wget and then play the local file, but wanted something a bit more elegant. Thanks, Mike Clark

Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've been getting a lot of timeouts on non-critical invite transactions. I turned on sip debug. They were the result of SIP invites like this: Retransmitting #10 (NAT) to 185.107.94.10:13057: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4p;received=185.107.94.10;rport=13057 From:

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples:

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

Q: What platform don't run this script: ---------------------------------------- #!/bin/sh export X0='x0' export X1a="$X0/1" export X1b='$X0/1' export X2a="$X1a/2" export X2b='$X1b/2' $SHELL <<EOF_2 $SHELL <<EOF_1 cat <<EOF #define a "$X2a/aa" #define b "$X2b/bb" EOF EOF_1 EOF_2

Hello, all. I must be brain cramping badly on our Asterisk 1.6.1.1 installation. Our MeetMe macros are working fine except they do not prompt for a PIN. So I made a very simple conference room: exten => 7777,1,MeetMe(123456,cMaAsx,123456) Shouldn't this prompt the user who dials 7777 to enter a PIN before entering the conference room whether or not a PIN is defined in meetme.conf? I

Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place