Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?
Am 01.10.2014 um 15:48 schrieb Gokan Atmaca: Someone reported me that from a PBX on which someone gained fraudulent access, he could observe hundreds of calls to the same destination number. For curiosity's sake, I'm wondering why would this happen (dialing the same number over and over) ? Some special numbers generate here and there revenues for callees (and not for callers). Beside sharing interests with the callee that get those revenues, why a hacker would like to dial the same numbers over and over ? In other words, in this case, is looking at callee number a promising path to find hackers ? Is there a bot virus ? Do you IP address restrictions ? I have one SIP Proxy without any outbound trunks/routing and this Proxy is just collecting bad source IPs and bad destination numbers for the database blacklist table and I use this blacklist table in my productive System. On Wed, Oct 1, 2014 at 4:36 PM, Administrator TOOTAI ad...@tootai.net wrote: Le 01/10/2014 11:40, Olivier a écrit : Hi, Hi Someone reported me that from a PBX on which someone gained fraudulent access, he could observe hundreds of calls to the same destination number. For curiosity's sake, I'm wondering why would this happen (dialing the same number over and over) ? Some special numbers generate here and there revenues for callees (and not for callers). Beside sharing interests with the callee that get those revenues, why a hacker would like to dial the same numbers over and over ? callee is also the bad men. Go and buy an 899 number in France, hack PBXS and call your number :-) [...] -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sent ami event from AGI?
hello, is there way to send event to all ami clients from AGI script? Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sent ami event from AGI?
You can use the AGI command EXEC to execute a dialplan application, and the application UserEvent can be used to generate custom events that AMI clients can receive. https://wiki.asterisk.org/wiki/display/AST/AGICommand_exec https://wiki.asterisk.org/wiki/display/AST/Application_UserEvent On Thu, Oct 2, 2014 at 4:02 AM, Ilya Awesome jl...@me.com wrote: hello, is there way to send event to all ami clients from AGI script? Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients
Hi, Is there anything I can do with this problem? Re-installing Asterisk does not solve this and the problem still persists. Or is there any other logs or configurations I can provide to help figure out why Asterisk is removing lines from the sdp? Any ideas would be greatly appreciated! I also tried removing everything under /etc/asterisk/ and make samples to restore any errors I could have had in my configurations, then restoring my minimal configuration: asterisk.conf, extconfig.conf, extensions.conf, res_mysql.conf and sip.conf. This did not help. (in case this message comes double, I just canceled posting of previous similar one as it was too big) cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?
On Thu, Oct 02, 2014 at 07:52:34AM +0200, Rainer Piper wrote: Is the destination Number like Country Code +972? +972 59 xx(x) mobile - Jawall [moving to 7-digit subscriber numbers] source - http://www.wtng.info/wtng-972-il.html That page is slightly dated. +972 59 XXX are all the numbers in the Palestinian Authority (there are several providers besides Jawall). My SIP Proxy logs all the unauth. INVITEs and I found the a lot calls go to the Country code +972 xxx As a resident of +972 (+972-4), I'll just note that those hack attempts are typically related to PA numbers (+972-59) as rates there are higher. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sent ami event from AGI?
Works! how I miss that… Thanks. On 02 Oct 2014, at 17:05, Scott Griepentrog sgriepent...@digium.com wrote: You can use the AGI command EXEC to execute a dialplan application, and the application UserEvent can be used to generate custom events that AMI clients can receive. https://wiki.asterisk.org/wiki/display/AST/AGICommand_exec https://wiki.asterisk.org/wiki/display/AST/Application_UserEvent On Thu, Oct 2, 2014 at 4:02 AM, Ilya Awesome jl...@me.com wrote: hello, is there way to send event to all ami clients from AGI script? Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 1.2.0 Released
The AstLinux Team has released 1.2.0. All current users are encouraged to upgrade as this release addresses the bash ShellShock bug. New in 1.2.0: * New Linux Kernel 3.2.x * igb ethernet driver for Intel Atom C2000 * Enable AES-NI support * New sip-user-agent firewall plugin * New versions of Asterisk 11 and 1.8 * Bash ShellShock security fixes A full changelog can be viewed in the release pages: http://www.astlinux.org/release/120-asterisk-11121 http://www.astlinux.org/release/120-asterisk-18300 New AstLinux Documentation Topics: SMTP Local Aliases http://doc.astlinux.org/userdoc:tt_smtp_aliases Updated AstLinux Documentation Topics: Firewall Plugins - sip-user-agent http://doc.astlinux.org/userdoc:tt_firewall_plugins#sip-user-agent --The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstLinux 1.2.0 Released
On Thu, Oct 2, 2014 at 6:04 PM, Darrick Hartman dhart...@djhsolutions.com wrote: The AstLinux Team has released 1.2.0. All current users are encouraged to upgrade as this release addresses the bash ShellShock bug. New in 1.2.0: * New Linux Kernel 3.2.x * igb ethernet driver for Intel Atom C2000 * Enable AES-NI support * New sip-user-agent firewall plugin * New versions of Asterisk 11 and 1.8 * Bash ShellShock security fixes A full changelog can be viewed in the release pages: http://www.astlinux.org/release/120-asterisk-11121 http://www.astlinux.org/release/120-asterisk-18300 New AstLinux Documentation Topics: SMTP Local Aliases http://doc.astlinux.org/userdoc:tt_smtp_aliases Updated AstLinux Documentation Topics: Firewall Plugins - sip-user-agent http://doc.astlinux.org/userdoc:tt_firewall_plugins#sip-user-agent --The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for the info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients
Asterisk is not a SIP Proxy. It is a B2BUA and will *always* replace the SDP with its own. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olli Heiskanen Sent: Thursday, October 02, 2014 9:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients Hi, Is there anything I can do with this problem? Re-installing Asterisk does not solve this and the problem still persists. Or is there any other logs or configurations I can provide to help figure out why Asterisk is removing lines from the sdp? Any ideas would be greatly appreciated! I also tried removing everything under /etc/asterisk/ and make samples to restore any errors I could have had in my configurations, then restoring my minimal configuration: asterisk.conf, extconfig.conf, extensions.conf, res_mysql.conf and sip.conf. This did not help. (in case this message comes double, I just canceled posting of previous similar one as it was too big) cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients
Hi, Thanks Eric for your reply, yes I know Asterisk replaces the sdp, however it should create ice lines when calling to a webrtc client, which it is currently not doing. To recap my problem (check previous messages for details); I have 2 webrtc clients (sip.js on chrome) with realtime information that appears to be correct. When calling from A to B, INVITE coming to Asterisk contains correct sdp, but when the INVITE leaves Asterisk, the sdp lacks ice lines. cheers, Olli 2014-10-02 18:13 GMT+03:00 Eric Wieling ewiel...@nyigc.com: Asterisk is not a SIP Proxy. It is a B2BUA and will **always** replace the SDP with its own. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olli Heiskanen *Sent:* Thursday, October 02, 2014 9:06 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients Hi, Is there anything I can do with this problem? Re-installing Asterisk does not solve this and the problem still persists. Or is there any other logs or configurations I can provide to help figure out why Asterisk is removing lines from the sdp? Any ideas would be greatly appreciated! I also tried removing everything under /etc/asterisk/ and make samples to restore any errors I could have had in my configurations, then restoring my minimal configuration: asterisk.conf, extconfig.conf, extensions.conf, res_mysql.conf and sip.conf. This did not help. (in case this message comes double, I just canceled posting of previous similar one as it was too big) cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice Mail Questions
We are trying to add voice mail to our hotel rooms. Our current phone instruction cards say 'to reach voice mail dial ext 456. Replacing those instructions is not feasible at the moment. We have Feature Code *97 that takes them directly to their voice mail box. Question - What is an easy way to have exten 456 dial *97. We are using AsteriskNow distro, version11. Phil Ledon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?
Is the destination Number like Country Code +972? +972 59 xx(x) mobile - Jawall [moving to 7-digit subscriber numbers] source - http://www.wtng.info/wtng-972-il.html My SIP Proxy logs all the unauth. INVITEs and I found the a lot calls go to the Country code +972 xxx I've seen that a very high percentage of the SIP probing my Asterisk system has seen over the past few years, consist of attempts to phone numbers in +972 (or, more generally, the West Bank and/or Gaza). It's consistent enough that I've set up a Fail2Ban rule which slaps a semi-permanent block on any IP address which tries this, even once. Since the last time I did a firewall-reset, the resulting iptables rules have blocked over 2000 call attempts (one attacker at 142.54.180.50 has tried over 1200 times). These attempts seem to come from all over the world... I'd guess that the majority are being sent through 'botted systems. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to strip +1 out of incoming number
Hello, our VoIP send us caller ID +1(area)(number) for instance +16024224334 is there a way to strip +1 out of caller ID? -- Thanks for your support, Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to strip +1 out of incoming number
On 2/10/14 6:52 pm, motty cruz wrote:
Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?
${CALLERID(num):1} should do what you're after (or :2 if you need to
strip the + as well)
Kind regards,
Chris
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Re: [asterisk-users] how to strip +1 out of incoming number
Try the Filter function
Set(cid=${FILTER(0123456789,${CALLERID(NUM)})})
On Thu, Oct 2, 2014 at 10:52 AM, motty cruz motty.c...@gmail.com wrote:
Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?
--
Thanks for your support,
Motty
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A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
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Re: [asterisk-users] how to strip +1 out of incoming number
I prefer using FILTER() so if somehow CallerID arrived with something nasty it
will be filtered out.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Kiniston
Sent: Thursday, October 02, 2014 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] how to strip +1 out of incoming number
Try the Filter function
Set(cid=${FILTER(0123456789,${CALLERID(NUM)})})
On Thu, Oct 2, 2014 at 10:52 AM, motty cruz
motty.c...@gmail.commailto:motty.c...@gmail.com wrote:
Hello, our VoIP send us caller ID +1(area)(number) for instance +16024224334 is
there a way to strip +1 out of caller ID?
--
Thanks for your support,
Motty
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A human being should be able to change a diaper, plan an invasion, butcher a
hog, conn a ship, design a building, write a sonnet, balance accounts, build a
wall, set a bone, comfort the dying, take orders, give orders, cooperate, act
alone, solve equations, analyze a new problem, pitch manure, program a
computer, cook a tasty meal, fight efficiently, die gallantly. Specialization
is for insects.
---Heinlein
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Re: [asterisk-users] how to strip +1 out of incoming number
On Thu, Oct 2, 2014 at 10:52 AM, motty cruz motty.c...@gmail.com
wrote:
Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?
I've always done it as 2 steps to handle carrier weirdness:
; trim leading +1 from DNIS
same = n, execif($[${DNIS:0:1} =
+]?set(DNIS=${DNIS:1}))
same = n, execif($[${DNIS:0:1} =
1]?set(DNIS=${DNIS:1}))
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Voice Mail Questions
You can create an extension 456, but change the DIAL string to be Local/$97@from-internal The extension can be any type really, but normally in this case you would use Custom rather than SIP to avoid creating an actual extension. On Thu, Oct 2, 2014 at 12:32 PM, Phil Ledon ple...@lodgetech.com wrote: We are trying to add voice mail to our hotel rooms. Our current phone instruction cards say 'to reach voice mail dial ext 456. Replacing those instructions is not feasible at the moment. We have Feature Code *97 that takes them directly to their voice mail box. Question - What is an easy way to have exten 456 dial *97. We are using AsteriskNow distro, version11. *Phil Ledon* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
