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How can I monitor asterisk if all lines are registered etc? I have an asterisk on a remote location and sometime they reporting problems that phone is not ringing, they can not dial out etc. Usually I just restart asterisk and it solves the problem. Is there an application that will email me if case any line looses registration with with asterisk? Or any better solution! -- Joseph

Hello, recently I have seen spike in attacks on my asterisk server, this is what I get on the LCD of my phone: 201 at 76.220.5.205 or calls from 1000 sip1000 at 76.2230.5.205, have any idea on how to stop this calls? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" <sip:4941 at public_ip>' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941"

Ok, the script is attached... I'll post it on www.generationd.com when I have a chance. If you have any updates & improvements please email them to me! (The command line parameter handle is pretty stupid - just grew from testing to production without cleanup). MD _____ From: Craig Huff [mailto:huffcslists at gmail.com] Sent: Saturday, Septem...

2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>: > > Do you really want to detect "ChallengeSent"? That should occur also on > legitimate login processes... > Hi , strange thing is that I still have not this asterisk in production and I see many attempts Connection. Now keep in mind that when a connection of authentication is successful the

Do you have DISA setup? We're seeing lots of attackers running scripts that send digits until they strike a DISA, misconfigured mailbox, etc. (Assuming it wasn't a stupid employee forwarding an inbound call to a 9xxxxxxx number etc). Have a look at SecAst (www.generationd.com) - it detects callers sending too many digits, monitors digit dialing speeds, etc. to help identify and block these types of attacks. The free version is better than nothing (but if you've already suffered one $25k attack then you probably don't mind spending a bit of money). Or have...

Hi all, i am wondering if i can make two leg cdr in mysql cdr table. 1st Leg : Registrar the ATA which registered to the asterisk and it normally logging in cdr table. 2nd Leg : The CDR of carrier for the example if i send call like exten => _x.,1,Dial(SIP/${EXTEN}@AT&TIP) I this cause i can get the accrue duration of call because currently we are facing some call missing not coming

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

Hi everybody, what are the current options to get an Asterisk-system high available? Using two servers as active/passive with DRBD, Pacemaker/Corosync works very good, there are no quality issues of the voice quality, even not on high loaded servers and no problems with a lot of small packages. But for this you need two systems for every Asterisk-system, what is not "economic" in any

I'd suggest taking a look at the free edition of SecAst (www.generationd.com). It handles these messages perfectly (and can also use AMI security events) - so you don't need to constantly be updating fail2ban rules. It's a drop in replacement for fail2ban. -M- P.S. My opinions are my own and do not necessarily represent those of my employer. As an employ...

On Fri, Jan 9, 2015 at 5:24 PM, Michelle Dupuis <mdupuis at ocg.ca> wrote: > I'd suggest taking a look at the free edition of SecAst ( > www.generationd.com). It handles these messages perfectly (and can also > use AMI security events) - so you don't need to constantly be updating > fail2ban rules. It's a drop in replacement for fail2ban. > > -M- > > P.S. My opinions are my own and do not necessarily represent those of...

Anyone using asterisk behind isa2004? I just installed tonight on my sbs server and of course it broke all of my iax and sip connectivity. Anyone out there who knows how to do this already? Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050826/22b1e568/attachment.htm

How should be the macro rewritten? [macro-faxreceive] exten => s,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN}) exten => s,3,rxfax(${FAXFILE}) exten => s,103,Set(EMAILADDR=ronald@elmit.com) exten => s,104,Goto(3) ... [Jan 23 10:43:38] -- Executing Macro("Zap/3-1", "faxreceive") in new

Anyone know the gsm encoding mip requirement from g711? Or number of channels can be transcoded from g711 to gsm at a time. Thnx

Like many users I get my voicemails emailed to me, AND left on the asterisk server, so that I can retrieve them by phone or by email. However, I was frustrated that after I deleted a message in outlook that I still had to delete it from asterisk manually. So, I wrote a script that runs on the asterisk box that allows the user to simply reply to an email (from the asterisk pbx) - which causes

Another useful script for those interested.... On the www.generationd.com web site you will now find the "asteriskcontrol" script file. This script can automatically restart Asterisk (gracefully) following a change in external IP address - for dynamic IP hosts. As well, it can update the SIP/IAX configuration files to reflect the new external IP addresses....

Hi guys, I am planning to connect two Asterisk boxes that are currently running in two different countries, using IAX. I was wondering if anyone could provide me with some links or suggestion regarding best practices in connecting two Asterisk in such way. I guess many of you have already tried this, and already have some know-how (what I should be careful about, what to avoid, etc...)?

Hello I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru an adtran board. I want to add 4 more analog lines. Currently I have a Digium TDM40B. I'm wondering what the best upgrade path is, where I define 'best' as the solution that is most likely to work without problems (like interupt conflicts) and work with my current echo tuning . I see my purchase

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing

android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says "Don't Change the Format Unless You REALLY Know What