Showing posts with label ASTERISK. Show all posts
Showing posts with label ASTERISK. Show all posts

Monday, August 5, 2013

Convert all your Asterisk .wav recordings to .mp3

I have had many jobs (consulting and Salary) that have involved using Asterisk in a “Call-Center” environment. Whenever an agent would make a sale, the call would need to be recorded for verification purposes.  Now I could talk about Asterisk and call-recording for pages upon pages, but I will be focusing on large .wav to .mp3 conversion jobs on the Linux/Asterisk server itself.  First off, Asterisk can record calls using several different methods.  The first method being that you setup a feature called Automon in /etc/asterisk/features.conf and use *1 (or whatever you specify) to record a call in progress.  The second method is recording every call that comes in through a specific DID or enters a specific queue.  Another method is to setup a ‘call genie’ that you conference your phone to and it records the entire call-bridge.  There are pros and cons to every type of recording method and format.  You can use GSM, wav, g729, etc. to record calls.  Not matter what method you use, the simple script I wrote will convert your .wav files to .mp3 and also retain the timestamp of when the file was initially created (recorded). The options I use for mlame are what makes this script retain its awesome-ness over time.  The options allow the final-result mp3 to be compressed and compatable with Asterisk.  This means that you can playback your .mp3 files that you converted over Asterisk.
Installation Instructions:
1. Copy the below script to your server and save it as convert_recordings.sh. 
 ========================================
#!/bin/bash
#Make sure to install Lame and copy mlame to your .wav dir
#If you specify your own filenames, use NO spaces.

recorddir="/var/spool/asterisk/monitor"
for i in `ls $recorddir/*.wav`; do
cd $recorddir   
$recorddir/mlame -f -o "-b 16 -m m -q 9 --resample 8" $i
date=`ls -l $i | awk '{print $7,$6,2008,$8}'`
mp3=`echo $i | sed 's/.wav/.mp3/g'`
touch -d "$date" $mp3

#Uncomment the below line if you want your .wavs moved to a different directory.
#This way you can review the timestamps and mp3 files before deleting the .wav
#mv $i /wav_recordings
done
============================
2. Put the script anywhere you wish, I tend to leave it in my home directory
3. chmod +x convert_recordings.sh
4. You need to install lame.  Get the file from the lame site.
5. Extract the tarball.  For Example…    tar -zxvf lame-398-2.tar.gz
6. cd lame-398-2
7.   ./configure
8. make
9. make install
10. Now copy the mlame file from lame-398-2/misc/mlame to your .wav directory /var/spool/asterisk/monitor (or wherever your .wavs are)
11. chmod +x mlame
12. Edit the convert_recordings.sh  recorddir variable to match your .wav directory path (no trailing ‘/’). The default directory is set to /var/spool/asterisk/monitor
13. Run the script by navigating to the directory and   ./recordings_convert.sh
You should see it take a few seconds for each file and go on.  I usually test it out by letting it convert a few files then Cntrl + C and ls -l the directory to check the filesizes and timestamps.  There is a commented out section at the bottom of recordings_convert.sh that moves the .wav files to a separate directory (easier to check if it worked) and in my case, I keep the old files for a month or so for job security reasons.  If you lose a bunch of verification recordings, you are going to have major problems.
Cron it, at it, and happy converting.
__________________________________
Code for .g729 to MP3 script.  Very dirty, but works, please test on a small group of test files first.


#!/bin/bash
#Author: Gregg Hansen  20080414
#Used to convert from .g729 -in and -out files to .mp3
#Run on Recordings server

for i in `ls /ramdiskunload/*.g729`;
do
#convert all .g729 to wav first, then soxmix
DST=`echo $i | sed 's/.g729/.wav/g'`
asterisk -rx "file convert $i $DST"
rm -f $i
mv $DST /recordings
done
#all files should now be in the /recordings directory
#use soxmix and mlame to convert them to one file => mp3
for j in `ls /recordings/*-in.wav`;
do
INFILE=`echo $j | sed 's/-in/-out/g'`
MIXED=`echo $j | sed 's/-in//g'`
soxmix $j $INFILE $MIXED
rm -f $j
rm -f $INFILE
/root/test/mlame -f -o "-b 16 -m m -q 9 --resample 8" $MIXED
done


-----------------------------------------------------------------

Code for .g729 to wav script.  Very dirty, but works, please test on a small group of test files first.
 
#!/bin/bash
#Author: Gregg Hansen  20080414
#Used to convert from .g729 -in and -out files to .mp3
#Run on Recordings server
for i in `ls /root/mix/*.gsm`;
do
#convert all .g729 to wav first, then soxmix
LDST=/root/mix/recordings
DST=`echo $i | sed 's/.gsm/.wav/g'`
asterisk -rx "file convert $i $DST"
rm -f $i
mv $DST $LDST
done
#all files should now be in the /recordings directory
#use soxmix and mlame to convert them to one file => mp3
for j in `ls /root/mix/recordings/*-in.wav`;
do
INFILE=`echo $j | sed 's/-in/-out/g'`
MIXED=`echo $j | sed 's/-in//g'`
soxmix $j $INFILE $MIXED
rm -f $j
rm -f $INFILE
#/root/lame/misc/mlame -f -o "-b 16 -m m -q 9 --resample 8" $MIXED
done
 

Convert audio files for use in Asterisk

Convert files from the CLI

You just recorded a fabulous audio file to use as you main voice menu. Then you realize that Asterisk does not use WAV format audio for the Playback or Background applications. So what do you do? How can you convert your WAV files into GSM files that still have good sound quality? (This is partially false, Asterisk can play anything it has a format and codec for, including some wav files. See below.)

Note the differences!

gsm: raw gsm encoding, good for VoIP
wav: MS wav format, 16 bit linear
WAV: MS wav format, gsm encoded (wav49)


Converting to sln format

Starting from Asterisk 1.2.0, the .sln (SLINEAR) format seems to be the preferred format.
To convert wav file to sln, use the following command:

sox foo-in.wav -t raw -r 8000 -s -w -c 1 foo-out.sln

Note that sox v14.3.0 and above (installed in Ubuntu 9.10), the -w option has changed to -2

sox foo-in.wav -t raw -r 8000 -s -2 -c 1 foo-out.sln

If you have a directory full of .wav files to convert, try this command. It uses sed to automatically rename the files with the .sln extension (assuming incoming wav files at a sample rate other than 8khz.)

for a in *.wav; do sox "$a" -t raw -r 8000 -s -w -c 1 `echo $a|sed "s/.wav/.sln/"` resample -ql; done

Converting your WAV files to good GSM files is easier than you might think if you have the program Sox installed (on Debian systems the libsox-fmt-gsm package is required in addition to sox). From the shell prompt, enter this command:

sox foo.wav -r 8000 foo.gsm resample -ql

and hit the <ENTER> key. Note that the sox option '-ql' (lower case L) modifies the resample option. It is not a number one (1). In a few moments you will have a new GSM format file in the same directory as the original WAV file. In this example "foo.wav" is your main voice menu audio file in WAV format, and "foo.gsm" is the same file converted to GSM format. If you wanted to, you could use "main-voice-menu.gsm" as the name in place of "foo.gsm": what matters here is the second file name you use in this command ends in ".gsm".

If your WAV file was in stereo, add the -c1 option to convert to mono, or the output will sound very strange.

sox foo.wav -r 8000 -c1 foo.gsm resample -ql

You may get better results if you record your WAV file in 16 bit 8000 Hz mono and then run

sox foo.wav foo.gsm

If you have multiple WAV files in one directory and you want to convert them all, use this command:

for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm" resample -ql; done

You can also put a bash script in /usr/bin and name it wav-gsm-convert. The content can be like this

  1. !/bin/bash
s=`echo $1| sed -e's/\.wav//'|xargs -i{} echo {}.gsm`
sox -t wav $1 -r 8000 -c1 -t gsm $s resample -ql


Next, move your new foo.gsm file to the directory: /var/lib/asterisk/sounds

Now you can easily use the applications Playback and Background in your extensions.conf file to play your fabulous main voice menu. For example:
exten => s,1,Background(foo)
or
exten => s,1,Background(main-voice-menu)
or
exten => s,1,Playback(foo)
or
exten => s,1,Playback(main-voice-menu)

Playing .sln files from the command line

You can play sln files using sox from the command line (play is part of sox):

play -t raw -r 8000 -s -w -c 1 file.sln

Using WAV files

Asterisk has codecs for wav (pcm), gsm, g729, g726, and wav49, all of which can be used for Playback and Background. However, Asterisk does not understand ADPCM WAV files. To convert your WAV files to a format which Asterisk can understand, use the following command:

sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql

Note that sox v14.3.0 and above (installed in Ubuntu 9.10), resample is no longer used, remix is used or leave it as:

sox foo-in.wav -r 8k -c 1 -s -w foo-out.wav

All so if your using sox v14.3.foo and above and you are getting check_header errors with play back try this:

sox foo-in.wav -r 8k -c 1 -e gsm foo-out.wav

This also works converting mp3, just make sure you have libsox-fmt-mp3 installed.

Normalizing volume and reducing volume fluxuations

Using the sox command's "compand" filter, you can reduce or eliminate flutters in volume level, and you can normalize the volume of your sound files. This is called dynamic range compression. The effect is desirable for a PBX system where changes in volume, or voices that are too quiet to understand would be considered unprofessional. It is important to perform the compand effect BEFORE you resample it as to preserve as much quality as possible.

An example command to perform some appropriate dynamic range compression and normalization is shown below:

sox "foo-in.wav" -r 8000 -c1 "foo-out.gsm" lowpass 4000 compand 0.02,0.05 -60,-60,-30,-10,-20,-8,-5,-8,-2,-8 -8 -7 0.05 resample -ql

In this example, foo-in.wav is a 16-bit mono 44khz uncompressed pcm WAV file.

Converting to a CD writable format

So, you've decided to do your call recording in GSM format as you don't care about quality and you don't want to stuff your disks full, but how do you write that file to an audio CD to send to somebody who wants to listen to the call?

sox infile.gsm -r 44100 -a outfile.wav

Wednesday, July 24, 2013

General CLI commands for Asterisk, vicidial, goautodial

! - Execute a shell command
abort halt - Cancel a running halt
cdr status - Display the CDR status
feature show - Lists configured features
feature show channels - List status of feature channels
file convert - Convert audio file
group show channels - Display active channels with group(s)
help - Display help list, or specific help on a command
indication add - Add the given indication to the country
indication remove - Remove the given indication from the country
indication show - Display a list of all countries/indications
keys init - Initialize RSA key passcodes
keys show - Displays RSA key information
local show channels - List status of local channels
logger mute - Toggle logging output to a console
logger reload - Reopens the log files
logger rotate - Rotates and reopens the log files
logger show channels - List configured log channels
meetme - Execute a command on a conference or conferee
mixmonitor - Execute a MixMonitor command.
moh reload - Music On Hold
moh show classes - List MOH classes
moh show files - List MOH file-based classes
no debug channel (null)
originate - Originate a call
realtime load - Used to print out RealTime variables.
realtime update - Used to update RealTime variables.
restart gracefully - Restart Asterisk gracefully
restart now - Restart Asterisk immediately
restart when convenient - Restart Asterisk at empty call volume
sla show - Show status of Shared Line Appearances
soft hangup - Request a hangup on a given channel
stop gracefully - Gracefully shut down Asterisk
stop now - Shut down Asterisk immediately
stop when convenient - Shut down Asterisk at empty call volume
stun debug - Enable STUN debugging
stun debug off - Disable STUN debugging
udptl debug - Enable UDPTL debugging
udptl debug ip - Enable UDPTL debugging on IP
udptl debug off - Disable UDPTL debugging



AEL commands

ael debug contexts - Enable AEL contexts debug (does nothing)
ael debug macros - Enable AEL macros debug (does nothing)
ael debug read - Enable AEL read debug (does nothing)
ael debug tokens - Enable AEL tokens debug (does nothing)
ael nodebug - Disable AEL debug messages
ael reload - Reload AEL configuration



Agents commands

agent logoff - Sets an agent offline
agent show - Show status of agents
agent show online - Show all online agents



AGI commands

agi debug - Enable AGI debugging
agi debug off - Disable AGI debugging
agi dumphtml - Dumps a list of agi commands in html format
agi show- List AGI commands or specific help
dnsmgr reload - Reloads the DNS manager configuration
dnsmgr status - Display the DNS manager status
http show status - Display HTTP server status



Console commands

console active - Sets/displays active console
console answer - Answer an incoming console call
console autoanswer - Sets/displays autoanswer
console boost - Sets/displays mic boost in dB
console dial - Dial an extension on the console
console flash - Flash a call on the console
console hangup - Hangup a call on the console
console mute - Disable mic input
console send text - Send text to the remote device
console transfer - Transfer a call to a different extension
console unmute - Enable mic input



Core related commands

core clear profile - Clear profiling info
core set debug channel - Enable/disable debugging on a channel
core set debug - Set level of debug chattiness
core set debug off - Turns off debug chattiness
core set global - Set global dialplan variable
core set verbose - Set level of verboseness
core show applications - Shows registered dialplan applications
core show application - Describe a specific dialplan application
core show audio codecs - Displays a list of audio codecs
core show channels - Display information on channels
core show channel - Display information on a specific channel
core show channeltypes - List available channel types
core show channeltype - Give more details on that channel type
core show codecs - Displays a list of codecs
core show codec - Shows a specific codec
core show config mappings - Display config mappings (file names to config engines)
core show file formats - Displays file formats
core show file version - List versions of files used to build Asterisk
core show functions - Shows registered dialplan functions
core show function - Describe a specific dialplan function
core show globals - Show global dialplan variables
core show hints - Show dialplan hints
core show image codecs - Displays a list of image codecs
core show image formats - Displays image formats
core show license - Show the license(s) for this copy of Asterisk
core show profile - Display profiling info
core show switches - Show alternative switches
core show threads - Show running threads
core show translation - Display translation matrix
core show uptime - Show uptime information
core show version - Display version info
core show video codecs - Displays a list of video codecs
core show warranty - Show the warranty (if any) for this copy of Asterisk



Database commands

database del - Removes database key/value
database deltree - Removes database keytree/values
database get - Gets database value
database put - Adds/updates database value
database show - Shows database contents
database showkey - Shows database contents



Dialplan commands

dialplan add extension - Add new extension into context
dialplan add ignorepat - Add new ignore pattern
dialplan add include - Include context in other context
dialplan reload - Reload extensions and *only* extensions
dialplan remove extension - Remove a specified extension
dialplan remove ignorepat - Remove ignore pattern from context
dialplan remove include - Remove a specified include from context
dialplan save - Save dialplan
dialplan show - Show dialplan



DUNDI commands

dundi debug - Enable DUNDi debugging
dundi flush - Flush DUNDi cache
dundi lookup - Lookup a number in DUNDi
dundi no debug - Disable DUNDi debugging
dundi no store history - Disable DUNDi historic records
dundi precache - Precache a number in DUNDi
dundi query - Query a DUNDi EID
dundi show entityid - Display Global Entity ID
dundi show mappings - Show DUNDi mappings
dundi show peers - Show defined DUNDi peers
dundi show peer - Show info on a specific DUNDi peer
dundi show precache - Show DUNDi precache
dundi show requests - Show DUNDi requests
dundi show trans - Show active DUNDi transactions
dundi store history - Enable DUNDi historic records



GTalk & Jabber commands

gtalk reload - Enable Jabber debugging
gtalk show channels - Show GoogleTalk Channels
jabber debug - Enable Jabber debugging
jabber debug off - Disable Jabber debug
jabber reload - Enable Jabber debugging
jabber show connected - Show state of clients and components
jabber test - Shows roster, but is generally used for mog's debugging.



IAX2 commands

iax2 provision - Provision an IAX device
iax2 prune realtime - Prune a cached realtime lookup
iax2 reload - Reload IAX configuration
iax2 set debug - Enable IAX debugging
iax2 set debug jb - Enable IAX jitterbuffer debugging
iax2 set debug jb off - Disable IAX jitterbuffer debugging
iax2 set debug off - Disable IAX debugging
iax2 set debug trunk - Enable IAX trunk debugging
iax2 set debug trunk off - Disable IAX trunk debugging
iax2 show cache - Display IAX cached dialplan
iax2 show channels - List active IAX channels
iax2 show firmware - List available IAX firmwares
iax2 show netstats - List active IAX channel netstats
iax2 show peers - List defined IAX peers
iax2 show peer - Show details on specific IAX peer
iax2 show provisioning - Display iax provisioning
iax2 show registry - Display IAX registration status
iax2 show stats - Display IAX statistics
iax2 show threads - Display IAX helper thread info
iax2 show users - List defined IAX users
iax2 test losspct - Set IAX2 incoming frame loss percentage



Manager commands

manager show command - Show a manager interface command
manager show commands - List manager interface commands
manager show connected - List connected manager interface users
manager show eventq - List manager interface queued events
manager show users - List configured manager users
manager show user - Display information on a specific manager user



MGCP commands

mgcp audit endpoint - Audit specified MGCP endpoint
mgcp reload - Reload MGCP configuration
mgcp set debug - Enable MGCP debugging
mgcp set debug off - Disable MGCP debugging
mgcp show endpoints - List defined MGCP endpoints



Module management

module load - Load a module by name
module reload - Reload configuration
module show - List modules and info
module show like - List modules and info
module unload - Unload a module by name



PRI commands

pri debug span - Enables PRI debugging on a span
pri intense debug span - Enables REALLY INTENSE PRI debugging
pri no debug span - Disables PRI debugging on a span
pri set debug file - Sends PRI debug output to the specified file
pri show debug - Displays current PRI debug settings
pri show spans - Displays PRI Information
pri show span - Displays PRI Information
pri unset debug file - Ends PRI debug output to file



Queue commands

queue add member - Add a channel to a specified queue
queue remove member - Removes a channel from a specified queue
queue show - Show status of a specified queue
rtcp debug ip - Enable RTCP debugging on IP
rtcp debug - Enable RTCP debugging
rtcp debug off - Disable RTCP debugging
rtcp stats - Enable RTCP stats
rtcp stats off - Disable RTCP stats
rtp debug ip - Enable RTP debugging on IP
rtp debug - Enable RTP debugging
rtp debug off - Disable RTP debugging
say load - Set/show the say mode
show parkedcalls - Lists parked calls
show queue - Show information for target queue
show queues - Show the queues



SIP commands

sip history - Enable SIP history
sip history off - Disable SIP history
sip notify - Send a notify packet to a SIP peer
sip prune realtime - Prune cached Realtime object(s)
sip prune realtime peer - Prune cached Realtime peer(s)
sip prune realtime user - Prune cached Realtime user(s)
sip reload - Reload SIP configuration
sip set debug - Enable SIP debugging
sip set debug ip - Enable SIP debugging on IP
sip set debug off - Disable SIP debugging
sip set debug peer - Enable SIP debugging on Peername
sip show channels - List active SIP channels
sip show channel - Show detailed SIP channel info
sip show domains - List our local SIP domains.
sip show history - Show SIP dialog history
sip show inuse - List all inuse/limits
sip show objects - List all SIP object allocations
sip show peers - List defined SIP peers
sip show peer - Show details on specific SIP peer
sip show registry - List SIP registration status
sip show settings - Show SIP global settings
sip show subscriptions - List active SIP subscriptions
sip show users - List defined SIP users
sip show user - Show details on specific SIP user



Skinny commands

skinny reset - Reset Skinny device(s)
skinny set debug - Enable Skinny debugging
skinny set debug off - Disable Skinny debugging
skinny show devices - List defined Skinny devices
skinny show lines - List defined Skinny lines per device



Voicemail commands

voicemail show users - List defined voicemail boxes
voicemail show users for - List defined voicemail boxes for target context
voicemail show zones - List zone message formats



Zaptel commands

zap destroy channel - Destroys a channel
zap restart - Fully restart zaptel channels
zap show cadences - List cadences
zap show channels - Show active zapata channels
zap show channel - Show information on a channel
zap show status - Show all Zaptel cards status

Friday, July 12, 2013

Multiple carriers

Multiple Carriers in Asterisk
or Utilizing all the TRUNK's randomly

Below Dialplan will Dial the number randomly on the assigned Carriers

[general]
TRUNKA=SIP/voiptrunk
TRUNKB=SIP/clickdial
TRUNKC=DAHDI/g0


exten => _9044X.,1,Answer
exten => _9044X.,n,Set(Trunk=${RAND(1|3)})
exten => _9044X.,n,GoToIf($[${Trunk} = 1]?trunkA)
exten => _9044X.,n,GoToIf($[${Trunk} = 2]?trunkB)
exten => _9044X.,n,GoToIf($[${Trunk} = 3]?trunkC)
exten => _9044X.,n,Hangup
exten => _9044X.,n(trunkA),Dial(${TRUNKA}/${EXTEN:4},,tToR)
exten => _9044X.,n(trunkB),Dial(${TRUNKB}/${EXTEN:4},,tToR)
exten => _9044X.,n(trunkC),Dial(${TRUNKC}/${EXTEN:4},,tToR)
exten => _9044X.,n,Hangup

Install Fail2ban in Asterisk (Centos)

Installing Fail2ban in centos
1.yum install fail2ban
If your CentOS doesn't find the package, please execute the following command and then try again.
2.rpm -Uvh http://download.fedora.redhat.com/pub/epel/5/i386/epel-release-5-4.noarch.rpm
3.yum install python iptables
or
wget http://downloads.sourceforge.net/project/fail2ban/fail2ban-stable/fail2ban-0.8.4/fail2ban-0.8.4.tar.bz2?use_mirror=transact

tar -xf fail2ban-0.8.4.tar.bz2
cd fail2ban-0.8.4
python setup.py install
cp files/redhat-initd /etc/init.d/fail2ban
chkconfig --add fail2ban
chkconfig fail2ban on

Once installing the Fail2ban  create asteirsk.conf file under the fail2ban directory

4.  vi /etc/fail2ban/filter.d/asterisk.conf

and copy and paste the below

# ===================
# /etc/fail2ban/filter.d/asterisk.conf
# Fail2Ban configuration file
#
#
# $Revision: 250 $
#
[INCLUDES]
# Read common prefixes. If any customizations available -- read them from
# common.local
#before = common.conf
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile.
#The
# host must be matched by a group named "host". The tag "" can
# be used for standard IP/hostname matching and is only an alias
#for
# (?:::f{4,6}:)?(?PS+)
# Values: TEXT
#

failregex = Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - Wrong password
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - No matching peer found
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - Device does not match ACL
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - Username/auth name mismatch
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - Peer is not supposed to register
            NOTICE.* <HOST> failed to authenticate as '.*'$
            NOTICE.* .*: No registration for peer '.*' (from <HOST>)
            NOTICE.* .*: Host <HOST> failed MD5 authentication for '.*' (.*)
            VERBOSE.* logger.c: -- .*IP/<HOST>-.* Playing 'ss-noservice' (language '.*')


# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
# ===================
Add the [asterisk-iptables] section to your /etc/fail2ban/jail.conf file :
# /etc/fail2ban/jail.conf
#====================



5 .  Save and exit the file
6.   vi /etc/fail2ban/jail.conf
      go to the last line of theis file and paste the below lines there

[asterisk-iptables]
enabled = true
filter = asterisk
action = iptables-allports[name=ASTERISK, protocol=all]
sendmail-whois[name=ASTERISK,
dest=youremailaddress@somewhere.com, sender=fail2ban@somewhere.com]
logpath = /var/log/asterisk/full
maxretry = 5
bantime = 600
#====================


7. Also in /etc/fail2ban/jail.conf file you want to add your own IP address range ( ours is192.168.1.0 ) :
ignoreip = 127.0.0.1 192.168.1.0/24

8.  make the fail2ban to start at startup
     chkconfig fail2ban on
9.  start the fail2ban now
    /etc/init.d/fail2ban start
10 . now check whether the fail2ban is installed properly to detect the attacks
       iptables -L –v
      You should see "fail2ban-ASTERISK" in your iptables output.

11. now try to register a extension from outside with wrong password or worng extension and run the iptables command to see the blocked ip addresses

automatic Blocking Hackers ip who access ssh

Automatically Blocking the hackers ip who access the server via ssh with wrong password
using IPtables


Dependencies
1.Iptables
2.postfix/sendmail ( for email alert)


step 1

1.login to your server via ssh
2. go to cd /usr/src/
3. vi scan-secure.sh
4. copy and paste the below script there
#!/bin/sh

# scan /var/log/secure for ssh attempts
# use iptables to block the bad guys

# Looking for attempts on existing and non-existing users. For example:
# Nov 2 22:44:07 pbxer sshd[28318]: Failed password for root from 74.143.42.70 port 52416 ssh2
# Nov 3 00:06:57 pbxer sshd[31767]: Failed password for invalid user mat3 from 192.203.145.200 port 35841 ssh2

tail -1000 /var/log/secure | awk '/sshd/ && /Failed password for/ { if (/invalid user/) try[$13]++; else try[$11]++; }
END { for (h in try) if (try[h] > 4) print h; }' |
while read ip
do
# note: check if IP is already blocked...
/sbin/iptables -L -n | grep $ip > /dev/null
if [ $? -eq 0 ] ; then
# echo "already denied ip: [$ip]" ;
true
else
echo "Subject: denying ip: $ip" | /usr/sbin/sendmail urmailid@gmail.com
logger -p authpriv.notice "*** Blocking SSH attempt from: $ip"
/sbin/iptables -I INPUT -s $ip -j DROP
fi
done

5. type chmod 755 /usr/src/scan-secure.sh
6. make entry in the cron to run in every one or two minutes
crontab -e
* * * * * /usr/src/scan-secure.sh
7. now start the iptables
/etc/init.d/iptables restart


to check for the blocked hackers ip
type iptables -L -n

Free G729 codec for asterisk, vicidial, goautodial

Installing Free g729 codec in asterisk

1. Download the appropriate codec from the below link
    http://asterisk.hosting.lv/

Check your asterisk version  --  asterisk -rx "core show version"
check whether 32bit or 64 bit ---  uname -a

For asterisk 1.4 version and 32 bit Pentium based server (like intel xeon ,p4, dualcore core2duo0
http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-pentium4.so
For asterisk 1.4 and 64bit os (intel xeon all pentium )
http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-x86_64-pentium4.so
For asterisk 1.4 AMD athlon machines
http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so

in linux type
cd /usr/lib/asterisk/modules
wget http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-pentium4.so
mv codec_g729-ast14-gcc4-glibc-pentium4.so codec_g729.so
chmod +x codec_g729.so
asterisk  -vvvvr
>module load codec_g729.so

now check whether the codec is loaded or not
>show translation
output's

g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
     g723    -   -    -    -        -     -    -     -    -     -    -    -    -
      gsm    -   -    2    2        2     2    1     2    4     -    -    2    -
     ulaw    -   2    -    1        2     2    1     2    4     -    -    2    -
     alaw    -   2    1    -        2     2    1     2    4     -    -    2    -
 g726aal2    -   2    2    2        -     2    1     2    4     -    -    2    -
    adpcm    -   2    2    2        2     -    1     2    4     -    -    2    -
     slin    -   1    1    1        1     1    -     1    3     -    -    1    -
    lpc10    -   2    2    2        2     2    1     -    4     -    -    2    -
     g729    -   2    2    2        2     2    1     2    -     -    -    2    -

Configuring Digium cards with Asterisk , goautodial , vicidial , vicidialnow , freepbx

How to configure the Digium PRI cards in Asterisk or vicidial or Goautodial or vicidialnow or Freepbx
  • If you are using the precompiled iso of asteirsk software like ( trixbox , elastix , pbxinaflash , goautodial , vicibox,)  then the Dhadi driver will be by default installed , if not you need to install the dahdi driver manually.(installation of dahdi explained at last of this doc)
steps to configure the Digium cards
  • First you need to check whether card is in E1 or T1 mode.
  • Take card outside and check for the E1/T1 changeover jumper in the card, if your PRI line is E1 then you need close the jumper , if T1 then it should be open. then insert the card into the server.
  • Access your asterisk server via  ssh  , putty will be good tool to connect the server remotely via ssh
  • First you need to check whether the card is  recognised   by the your server 
  1.  type  lspci  this will output pci boards connected in the server , if digium card is  recognised   it  shows output as below     0e:08.0 Ethernet controller: Digium, Inc. Wildcard TE121 single-span T1/E1/J1 card (PCI-Express) (rev 11)    

  • Once the card is  recognised , follow the below steps

  1. type  dahdi_genconf  -vvvvvvv    -this will auto install configure the digium card  driver and conf files

  • [root@localhost ~]# dahdi_genconf -vvvvvv
  • Default parameters from /etc/dahdi/genconf_parameters
  • Generating /etc/dahdi/system.conf
  • Generating /etc/asterisk/dahdi-channels.conf
2. Type  dahdi_cfg  -vvvv     - this outputs the channels if it configured properly


[root@mrlpriprimary ~]# dahdi_cfg -vvvv
DAHDI Tools Version - 2.4.1
DAHDI Version: 2.4.1.2
Echo Canceller(s): MG2
Configuration
======================
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
31 channels to configure.
Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03)
...........................................................................................................
Channel 30: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 30)
Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31)
3. Now configuration is done, some small changes to be done to enable the channels in the asterisk

  •    go to   vi  /etc/asterisk/chan_dahdi.conf
  •    go to last line of the file by pressing shift+g
  •    then add this line    #include dahdi-channels.conf
  •     save and exit the file.
4. Now go to asterisk cli  and type module reload chand_dahdi.so   or restart the server.
5. if the PRI line is up you can see green lights on the card.
6. if it shows RED light then either your pri is down or the pri cable not connected properly.
7. by default the digium cards configured with group 0 or 11 , so in your dialplan you need to mention g0
8. so you need write dialplan as 
    exten => _X.,Dial(DAHDI/g0/${EXTEN},,)
   "if goautodial/vicidial/vicibox  use the below dialplan in carrier or custome dialplan"
    exten => _9XXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
    exten => _9XXXXX.,2,Dial(DAHDI/g0/${EXTEN:1},,TtoR)
    exten => _9XXXXX.,3,Hangup
8a. if your using freepbx based pbx then no need to above line , just you need to add g0 in the DAHDI identifier under trunk (add new dahdi trunk)
9. And the default incomming context will the from-pstn
Installing the DAHDI driver
a. you need to install libpri if you use pri lines
b. go to http://www.asterisk.org/downloads and download the latest libpri
c. as of writing this blog 1.4.12 is latest
d. wget http://downloads.asterisk.org/pub/telephony/libpri/releases/libpri-1.4.12.tar.gz
c. tar -xvzf  libpri-1.4.12.tar.gz
d. cd libpri-1.4.12
e. make
d. make install
e. now you need to install dahdi.
1.go to http://www.asterisk.org/downloads and download the latest dahdi
as of writting this blog its 2.6.1 so
2. wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.6.1+2.6.1.tar.gz
3. tar -xvzf  dahdi-linux-complete-2.6.1+2.6.1.tar.gz
4. cd dahdi-linux-complete-2.6.1+2.6.1.
5. make all
6. make install
7. make config.
once done follow the above steps to configure your card.