Linux Administration Tips and Tricks Asterisk ,FreeSwitch And Opensips

Shell Script to monitor the services and ports running in the linux system in the different Format

2011-10-07T13:30:00.002+05:30

The Script read the service running the Linux system in the different format

The Script read the port number from the sample.txt file and service name

sample .txt#INSTANCE_NAME:PORT:USER:EXTERNAL_PORT
http 80 root 81
https 443 root ssh 22 rrot mail 25 rrotftp 21 root

#!/bin/bash

cat /root/sample.txt|grep -v "^#"| while read line
do
#echo $line

NETSTAT=/bin/netstat
# for old file
INSTANCE_NAME=`echo $line | awk '{print $1}'`
PORT_NO=`echo $line | awk '{print $2}'`
USER=`echo $line | awk '{print $3}'`
# for new file
#INSTANCE_NAME=`echo $line | awk -F':' '{ print $1 }'`
#PORT_NO=`echo $line | awk -F':' '{ print $2 }'`
#USER=`echo $line | awk -F':' '{ print $3 }'`
echo $INSTANCE_NAME
echo  $PORT_NO
echo $USER
RPORTS=$($NETSTAT -utan|awk '{print $4}'

|sed -s "s/::ffff:[[:digit:]]\+\.[[:digit:]]\+\.[[:digit:]]\+\.[[:digit:]]/::/g"

|sed -s "s/0.0.0.0/::/g"|sed -s "s/127.0.0.1/::/g"|sed -s "s/[[:digit:]]\+\.[[:digit:]]\+\.[[:digit:]]\+\.[[:digit:]]/::/g"|

cut -f4 -d":"|grep -v 127.0.0.1)

  status="NO"
  for r in $RPORTS
  do
   if [ "$r" == "$PORT_NO" ]
   then
    status="YES"
    break
   fi
  done
  echo -n " $INSTANCE_NAME is running $status with user as $USER"
  echo ""
done

it monitor the services

0.0.0.0:80

 :::8080

::ffff:127.0.0.1:8005

Cricinfo International Scores

2011-07-07T15:56:00.000+05:30

Cricinfo International Scores: "Get the latest scores of all the international cricket matches from Cricinfo. Add the Cricinfo International Scores widget now!"

Apache, Tomcat cluster, load balancing using mod_jk

2011-01-31T17:32:00.001+05:30

Apache

The Apache HTTP Server Project is a collaborative software development effort aimed at creating a robust, commercial-grade, feature ful, and freely-available source code implementation of an HTTP (Web) server. The project is jointly managed by a group of volunteers located around the world, using the Internet and the Web to communicate, plan, and develop the server and its related documentation. This project is part of the Apache Software Foundation. In addition, hundreds of users have contributed ideas, code, and documentation to the project. This file is intended to briefly describe the history of the Apache HTTP Server and recognize the many contributors.

Tomcat

Apache Tomcat is an open source software implementation of the Java Servlet and JavaServer Pages technologies. The Java Servlet and Java Server Pages specifications are developed under the Java Community Process.
Apache Tomcat is developed in an open and participatory environment and released under the Apache License version 2. Apache Tomcat is intended to be a collaboration of the best-of-breed developers from around the world. We invite you to participate in this open development project.

Download the required software

Apache 2.x Binaries
Tomcat 6.x Binaries
The JK module

Installing of Apache and Tomcat please refer previous posting of installing of Apache and Tomcat

load the mod_jk.so module

LoadModule jk_module modules/mod_jk.so

JkWorkersFile conf/workers.properties
JkLogFile logs/mod_jk.log
JkLogLevel info

below DOcument Root

JkMount /*.jsp loadbalancer
JkMount /servlet/* loadbalancer

And create worker.properties file in /etc/httpd/conf/ directory

#
# workers.properties 
#

# In Unix, we use forward slashes:
ps=/

# list the workers by name

worker.list=tomcat1, tomcat2, loadbalancer

# ------------------------
# First tomcat server
# ------------------------
worker.tomcat1.port=8019
worker.tomcat1.host=localhost
worker.tomcat1.type=ajp13

# Specify the size of the open connection cache.
#worker.tomcat1.cachesize

#
# Specifies the load balance factor when used with
# a load balancing worker.
# Note:
#  ----> lbfactor must be > 0
#  ----> Low lbfactor means less work done by the worker.
worker.tomcat1.lbfactor=100


# ------------------------
# Second tomcat server
# ------------------------
worker.tomcat2.port=8029
worker.tomcat2.host=localhost
worker.tomcat2.type=ajp13

# Specify the size of the open connection cache.
#worker.tomcat2.cachesize

#
# Specifies the load balance factor when used with
# a load balancing worker.
# Note:
#  ----> lbfactor must be > 0
#  ----> Low lbfactor means less work done by the worker.
worker.tomcat2.lbfactor=100


# ------------------------
# Load Balancer worker
# ------------------------

#
# The loadbalancer (type lb) worker performs weighted round-robin
# load balancing with sticky sessions.
# Note:
#  ----> If a worker dies, the load balancer will check its state
#        once in a while. Until then all work is redirected to peer
#        worker.
worker.loadbalancer.type=lb
worker.loadbalancer.balanced_workers=tomcat1, tomcat2

#
# END workers.properties
#

and goto the tomcat server1 and edit the file /opt/tomcat1/conf/server.xml

edit  line

to

to

and goto the tomcat server2 and edit the file /opt/tomcat2/conf/server.xml

to

to

In  apache server

NameVirtualHost *


ServerName localhost1
JkMount /*.jsp tomcat1
JkMount /servlet/* tomcat1


ServerName localhost2
JkMount /*.jsp tomcat2
JkMount /servlet/* tomcat2

And save the files and restart the both tomcat1 and tomcat2 server and apache start

and in browser

http://localhost

apache index.html will dispaly

and http://localhost/index.jsp

request go to server1 and continue if will server load heavy

 it will redirect to server2 or in case of fail over

Redirecting Request from apache to tomcat Server using Mod jk

2011-01-31T17:16:00.002+05:30

Apache

The Apache HTTP Server Project is a collaborative software development effort aimed at creating a robust, commercial-grade, feature ful, and freely-available source code implementation of an HTTP (Web) server. The project is jointly managed by a group of volunteers located around the world, using the Internet and the Web to communicate, plan, and develop the server and its related documentation. This project is part of the Apache Software Foundation. In addition, hundreds of users have contributed ideas, code, and documentation to the project. This file is intended to briefly describe the history of the Apache HTTP Server and recognize the many contributors.

Tomcat

Apache Tomcat is an open source software implementation of the Java Servlet and Java Server Pages technologies. The Java Servlet and Java Server Pages specifications are developed under the Java Community Process.

Apache Tomcat is developed in an open and participatory environment and released under the Apache License version 2. Apache Tomcat is intended to be a collaboration of the best-of-breed developers from around the world. We invite you to participate in this open development project.

using mod_proxy.so and mod_proxy_ajp.so

load module in Apache

edit the file /etc/httpd/conf/httpd.conf

uncomment or add

Download Mod_jk binary from

wget http://www.apache.org/dist/tomcat/tomcat-connectors/jk/binaries/linux

workers.properties File

Create the workers.properties in your apache configuration directory. Example:
```
touch /etc/apache2/workers.properties
```

Using your favorite editor (I like to use vi), edit the file and add the following information (tailor it to suit your system):

   workers.tomcat_home=/opt/tomcat6
   workers.java_home=/opt/java
   ps=/
   worker.list=worker1

   worker.worker1.port=8009
   worker.worker1.host=127.0.0.1
   worker.worker1.type=ajp13
   worker.worker1.lbfactor=1

Virtual Host File

This is probably the most vital step that is missing in a lot of how-to's. The JkMount directive must be added to the virtual host file containing port 80. The virtual host definition may also be present in your apache2.conf or httpd.conf file. If not, create a new file called something like default.conf, add the contents below, and put default.conf somewhere in which it can be properly loaded by apache2.conf (or httpd.conf).


   ServerName your.hostname.com

   # Send servlet for context /jsp-examples to worker named worker1
   JkMount /jsp-examples worker1

   # Send JSPs for context /jsp-examples/* to worker named worker1
   JkMount /jsp-examples/* worker1

save the file and restart the httpd server

and goto tomcat install directory

conf directory and edit server.xml

and restart the tomcat server

Test It

To test your setup, type http://localhost/jsp-examples. It should also work with http://localhost:8080/jsp-examples.

it will redirect to tomcat jsp-examples home page

Creating Secure Site in Tomact and Apache

2011-01-31T17:06:00.001+05:30

Apache Tomcat

Installation of Apache Tomcat

Downloads the tar file for

http://apache.cs.utah.edu/tomcat/tomcat-6/v6.0.30/bin/apache-tomcat-6.0.30.tar.gz

tar -zxvf Apache-tomcat-6.0.30.tar.gz

cp -r tomcat /opt/tomcat

and write a start up scripts

create a file tomcat in /etc/init.d give chmod 777 permissions to tomcat file

copy the

#!/bin/bash
# chkconfig: - 10 20
# Source function library.
. /etc/rc.d/init.d/functions
# Source networking configuration.
. /etc/sysconfig/network
PATH=/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0/bin:$PATH:$HOME/bin:./
export PATH
export JAVA_HOME=/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0/jre
export CLASSPATH=/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0/lib/tools.jar:/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0/lib/dt.jar:./
export CATALINA_HOME=/opt/tomcat/
export CATALINA_TMPDIR=/opt/tomcat/temp
unset USERNAME

start()
{
cd /opt/tomcat6/bin/
sh startup.sh
}
stop()
{
cd /opt/tomcat/bin/
sh shutdown.sh
}
case "$1" in
start)
start
;;
stop)
stop
;;
restart)
stop
start
;;
*)
echo "start | stop | restart"
esac

save the file

and start tomcat /etc/init.d/tomcat start

and stop tomcat /etc/init.d/tomcat stop

http://localhost:8080

to create Secure site https

goto

cd /usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0/bin

keytool -genkey -alias name-keypass admin123 -keystore domainnamecom.bin -storepass admin123

Note:keypass and storepass must be same password

it generate a file with name domainnamecom.bin in same directory

copy the file domainnamecom.bin to /opt/tomcat/webapps

and Goto cd /opt/tomcat/conf

and edit server.xml

and uncommmnet

               maxThreads="150" scheme="https" secure="true"
               clientAuth="false" sslProtocol="TLS" keystoreFile="../webapps/domainnamecom.bin" keystorePass="admin123" />

and save the server.xml

and restart the tomcat server

https://localhost:8443

Apache Server

Compiling and Installing

Downloads source tar http://httpd.apache.org/download.cgi

tar -zxvf httpd-NN.tar

cd httpd

./configure --prefix=PREFIX

make

make install

configuration file is in

/etc/httpd/conf/httpd.conf

to start

/etc/init.d/httpd start

to stop

/etc/init.d/httpd stop

and in browser

http://localhost

To Create secure site

install mod_ssl and open_ssl

then generate a key file

openssl genrsa -out /etc/httpd/ssl/domain.com.key 1024

and using the key file generate certiifcate for 999 days

openssl req -new -key /etc/httpd/ssl/domain.com.key -x509 -out /etc/httpd/ssl/domainname.crt -days 999

and edit the /etc/httpd/conf/httpd.conf



 copy the lines

DocumentRoot "/var/www/html/ssl_doc_root/"

ServerAdmin you@mycorp.com

ServerName servername

SSLEngine On

SSLCipherSuite ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL

SSLCertificateFile /etc/httpd/ssl/domain.crt

SSLCertificateKeyFile /etc/httpd/ssl/domain.com.key

SetEnvIf User-Agent ".*MSIE.*" \

nokeepalive ssl-unclean-shutdown \

downgrade-1.0 force-response-1.0

save the file

restart the httpd server

and in browser

https://localhost

FreeSwitch Installation Steps in centos and Debain

2011-01-20T13:09:00.000+05:30

Freeswitch

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.

FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX. The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.

We support various communication technologies such as Skype, SIP, H.323 and GoogleTalk making it easy to interface with other open source PBX systems such as sipXecs, Call Weaver, Bayonne, YATE or Asterisk.

FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz and can bridge channels of different rates. The G.729 codec is also available under a commercial license.

FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.

FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two.

Centos Or Redhat

Installation Steps

yum install subversion autoconf automake libtool gcc-c++ ncurses-devel make expat-devel zlib zlib-devel

Dependencies For Additional Modules in freeswitch

yum install nano unixODBC-devel openssl-devel gnutls-devel libogg-devel libvorbis-devel curl-devel

Installation git in centos

yum install gettext-devel expat-devel curl-devel zlib-devel openssl-devel

wget http://kernel.org/pub/software/scm/git/git-1.6.1.tar.gz

untar the git source code

tar xvfz git-1.6.1.tar.gz

cd git-1.6.1.tar.gz

make prefix=/usr/local all

make prefix=/usr/local install

Downloads the source of Freeswitch from git and install

cd /usr/src/
git clone git://git.freeswitch.org/freeswitch.git
cd /usr/src/freeswitch
./bootstrap.sh
./configure
make install
make cd-sounds-install
make cd-moh-install

it install  default modules of Freeswitch if you needed additional modules

go to /usr/src/freeswitch

edit the modules.conf

uncomment the modules need

and

make install

In Debian

apt-get install autoconf automake g++ git-core libjpeg62-dev libncurses5-dev libtool make

cd /usr/src/
git clone git://git.freeswitch.org/freeswitch.git
cd /usr/src/freeswitch
./bootstrap.sh
./configure
make install
make cd-sounds-install
make cd-moh-install

it install  default modules of Freeswitch if you needed additional modules
go to /usr/src/freeswitch 
edit the modules.conf
uncomment the modules need
and 
make install

Start and stop commands

start
/usr/local/freeswitch/bin/freeswitch -nc  ;non console

stop

/usr/local/freeswitch/bin/freeswitch -stop

start up scripts

create a file in /etc/init.d/freeswitch

#!/bin/sh
#
# freeswitch.sh - startup script for freeswitch on FreeBSD
#
# This goes in /usr/local/etc/rc.d and gets run at boot-time.

case "$1" in

    start)
    if [ -x /usr/local/freeswitch/bin/freeswitch ] ; then
        echo -n " freeswitch"
        /usr/local/freeswitch/bin/freeswitch -nc &
    fi
    ;;

    stop)
    if [ -x /usr/local/freeswitch/bin/freeswitch ] ; then
        echo -n " freeswitch"
        /usr/local/freeswitch/bin/freeswitch -stop &
    fi
    ;;

    *)
    echo "usage: $0 { start | stop }" >&2
    exit 1
    ;;

esac

and give permissions of

chmod 755

chkconfig --add freeswitch
chkconfig --level 5 freeswitch on
chkconfig --list freeswitch

Read the File With Comma separte and copy the files to directory using Perl script

2011-01-20T11:48:00.000+05:30

To read the File and copy the files to the directorywith comma separate need the the Package

FileHandle.

#!/usr/bin/perl

use FileHandle;

if (@ARGV != 2) {
    print " the argument is invalided \n" ;
exit 1;
}

$file_name = shift;
$output_path = shift;

print "$file_name ,$output_path \n";
$dir1 = "/opt/test";

open (FILE, "$file_name") or die "Cannot open $file_name $!\n";

@fields;
while () {
    chomp;
    @fields = split(/\,/);

      foreach my $item (@fields) {
      #print "$item \n";
   }

    if ($#fields < 1) {
        print " Error in file format \n";
        next;
    }

   $file_id = $fields[4];
    print " $file_id \n";

    `cp -r "$dir1/$file_id" "$output_path/$file_id" `;

}
close (FILE);

give permissons to file of script and

perl test.pl new.txt /opt/test/copy

new.txt file format
1,wer,2,root,welcome.txt
2,wee,3,root,welcome1.log
3,333,4,root,sysd.log

Reading the Csv File and inser into Database using the Perl Script

2011-01-20T11:35:00.000+05:30

To read the CSV File or Xls file the following packages is needed

Spreadsheet::ParseExcel::Simple
Spreadsheet::ParseExcel
DBI

the perl script

#!/usr/bin/perl

use DBI;
use Spreadsheet::ParseExcel::Simple;
use Spreadsheet::ParseExcel;

my $workbook = Spreadsheet::ParseExcel::Workbook->Parse('Agent Details.xls');

#==================================================
#==================================================
foreach $sheet (@{$workbook->{Worksheet}}) {

       foreach $col ($sheet->{MinCol} .. $sheet->{MaxCol}) {

               if ($sheet->{Cells}[0][$col]->{Val} eq "COL1") {
                       $col1= $col;
               }
               if ($sheet->{Cells}[0][$col]->{Val} eq "COL2") {
                       $col2= $col;
               }
               if ($sheet->{Cells}[0][$col]->{Val} eq "COL3") {
                       $col3= $col;
               }
               if ($sheet->{Cells}[0][$col]->{Val} eq "COL4") {
                       $col4= $col;
               }
       }
#==================================================
#==================================================
       foreach $row ($sheet->{MinRow}+1 .. $sheet->{MaxRow}) {

                  $col_1 = $sheet->{Cells}[$row][$col1]->{Val};
                  $col_2 = $sheet->{Cells}[$row][$col2]->{Val};
                  $col_3 = $sheet->{Cells}[$row][$col3]->{Val};
                  $col_4 = $sheet->{Cells}[$row][$col4]->{Val};

#==================================================
#==================================================
                if( $col_1!= ' ' && $col_2!= ' ' && col_3 !=' ' && col_4 != ' ')
                {
                    $dbh = DBI->connect("DBI:mysql:databasename:localhost:3306", "username", "password")
                    or die "Couldn't connect to database: " . DBI->errstr;
                    $stmt = "INSERT INTO databasename.tablename (col1,col2,col3,col4) values ( '$col_1' ,'$col_2','$col_3','$col_4');";
                    $affected_rows = $dbh->do($stmt);
                }
       }
}
#==================================================

Asterisk Integration With Openldap

2010-12-10T12:48:00.000+05:30

LDAP integration for Asterisk

The "light weight directory" offers a means for central user administration for your network, including VoIP data.

To integration openldap with Asterisk

First Install and Configure the Openldap Server
After Installation Asterisk
In Extconfig.config do Following Changes

;     sipusers => ldap,"dc=myDomain,dc=myDomainExt",sip
;     sippeers => ldap,"dc=myDomain,dc=myDomainExt",sip
;     extensions => ldap,"dc=myDomain,dc=myDomainExt",extensions
;     sip.conf => ldap,"dc=myDomain,dc=myDomainExt",config
;      voicemail => ldap,"dc=myDomain,dc=myDomainExt",voicemail

And In res_ldap.conf

[_general]
; Specify one of either host and port OR url. URL is preferred, as you can
; use more options.
;host=192.168.1.1                    ; LDAP host
;port=389                                 ;Ldap port
;url=ldap://ldap3.mydomain.com:3890
;protocol=3                          ; Version of the LDAP protocol to use; default is 3.
;basedn=MyRootDN                     ; Base DN
;user=MyDN                              ; Bind DN
;pass=MyPassword                     ; Bind password

And
[config]
;
; additionalFilter - This specifies an additional set of criteria to be used
; when querying the LDAP server.
additionalFilter=(objectClass=AstConfig)
;
filename = AstConfigFilename
category = AstConfigCategory
variable_name = AstConfigVariableName
variable_value = AstConfigVariableValue
cat_metric = AstConfigCategoryMetric
commented = AstConfigCommented

[extensions]
context = AstExtensionContext
exten = AstExtensionExten
priority = AstExtensionPriority
app = AstExtensionApplication
appdata = AstExtensionApplicationData
additionalFilter=(objectClass=AstExtension)

[sip]
name = cn
amaflags = AstAccountAMAFlags
callgroup = AstAccountCallGroup
callerid = AstAccountCallerID
directmedia = AstAccountDirectMedia
context = AstAccountContext
dtmfmode = AstAccountDTMFMode
fromuser = AstAccountFromUser
fromdomain = AstAccountFromDomain
fullcontact = AstAccountFullContact
fullcontact = gecos
host = AstAccountHost
insecure = AstAccountInsecure
mailbox = AstAccountMailbox
md5secret = realmedPassword    ; Must be an MD5 hash. Field value can start with ,{md5} but it is not required.
nat = AstAccountNAT
deny = AstAccountDeny
permit = AstAccountPermit
pickupgroup = AstAccountPickupGroup
port = AstAccountPort
qualify = AstAccountQualify
restrictcid = AstAccountRestrictCID
rtptimeout = AstAccountRTPTimeout
rtpholdtimeout = AstAccountRTPHoldTimeout
type = AstAccountType
disallow = AstAccountDisallowedCodec
allow = AstAccountAllowedCodec
MusicOnHold = AstAccountMusicOnHold
regseconds = AstAccountExpirationTimestamp
regcontext = AstAccountRegistrationContext
regexten = AstAccountRegistrationExten
CanCallForward = AstAccountCanCallForward
ipaddr = AstAccountIPAddress
defaultuser = AstAccountDefaultUser
regserver = AstAccountRegistrationServer
lastms = AstAccountLastQualifyMilliseconds
additionalFilter=(objectClass=AsteriskSIPUser)

[voicemail]
context = AstAccountContext
mailbox = AstAccountMailbox
password=AstAccountPassword
email=AstAccountEmail
pager=AstAccountPager
tz=AstAccountTz
attach=AstAccountAttach
saycid=AstAccountSaycid
dialout=AstAccountDialout
callback=AstAccountCallback
review=AstAccountReview
operator=AstAccountOperator
envelope=AstAccountEnvelope
sayduration=AstAccountSayduration
saydurationm=AstAccountSaydurationm

sendvoicemail=AstAccountSendvoicemail
delete=AstAccountDelete
forcename=AstAccountForcename
forcegreetings=AstAccountForcegreetings
hidefordir=AstAccountHidefordir
stamp=AstAccountStamp

In extensions.conf
exten => 1234,1,LDAPget(CIDNAME=cidname/${CALLERIDNUM},${CALLERIDNUM},${CALLERIDNUM})
exten => 1234,2,SetCIDName(${CIDNAME})

exten => 1234,3,Dial(SIP/${EXTEN},15

Asterisk Queues Log In to Database

2010-11-15T14:49:00.001+05:30

In order to properly manage ACD queues, it is important to be able to keep track of details of call setups and teardowns in much greater detail  than traditional call detail records provide.  In order to support this, 

extensive and detailed tracing of every queued call is stored in the  queue log, located (by default) in /var/log/asterisk/queue_log. 

For asterisk 1.6.1Above 

Create a table queue_log in MYSQL DB

CREATE TABLE `asterisk.queue_log` (
  `id` int(10) unsigned NOT NULL auto_increment,
  `time` int(10) unsigned default NULL,
  `callid` varchar(32) NOT NULL default '',
  `queuename` varchar(32) NOT NULL default '',
  `agent` varchar(32) NOT NULL default '',
  `event` varchar(32) NOT NULL default '',
  `data` varchar(255) NOT NULL default '',

  `data1` varchar(255) NOT NULL default '',
  `data2` varchar(255) NOT NULL default '',

  `data3` varchar(255) NOT NULL default '',
  `data4` varchar(255) NOT NULL default '',
  PRIMARY KEY (`id`)
);

mysql> select * from queue_log;
+----+------------+--------------+------------------+-------+------------+-------+
| id | time       | callid       | queuename        | agent | event      | data  |
+----+------------+--------------+------------------+-------+------------+-------+
| 1  | 1198356717 | 1198356717.0 | voipsolutions.ru | NONE  | ENTERQUEUE | |serg |
| 2  | 1198356719 | 1198356717.0 | voipsolutions.ru | NONE  | ABANDON    | 1|1|2 |
+----+------------+--------------+------------------+-------+------------+-------+

Data field values is separated by Pipe(|) symbols

CREATE TABLE IF NOT EXISTS `agent_status` (
 `agentId` varchar(40) NOT NULL default '',
 `agentName` varchar(40) default NULL,
 `agentStatus` varchar(30) default NULL,
 `timestamp` timestamp NULL default NULL,
 `callid` double(18,6) unsigned default '0.000000',
 `queue` varchar(20) default NULL,
 PRIMARY KEY  (`agentId`),
 KEY `agentName` (`agentName`),
 KEY `agentStatus` (`agentStatus`,`timestamp`,`callid`),
 KEY `queue` (`queue`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1;


CREATE TABLE IF NOT EXISTS `call_status` (
 `callId` double(18,6) NOT NULL,
 `callerId` varchar(13) NOT NULL,
 `status` varchar(30) NOT NULL,
 `timestamp` timestamp NULL default NULL,
 `queue` varchar(25) NOT NULL,
 `position` varchar(11) NOT NULL,
 `originalPosition` varchar(11) NOT NULL,
 `holdtime` varchar(11) NOT NULL,
 `keyPressed` varchar(11) NOT NULL,
 `callduration` int(11) NOT NULL,
 PRIMARY KEY  (`callId`),
 KEY `callerId` (`callerId`),
 KEY `status` (`status`),
 KEY `timestamp` (`timestamp`),
 KEY `queue` (`queue`),
 KEY `position` (`position`,`originalPosition`,`holdtime`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1;

And then Created a trigger on the DB asterisk

DROP TRIGGER IF EXISTS `asterisk`.`queueEvents`;
DELIMITER //
CREATE TRIGGER `asterisk`.`queueEvents` BEFORE INSERT ON `asterisk`.`queue_log`
FOR EACH ROW BEGIN
 IF NEW.event = 'ADDMEMBER' THEN
   INSERT INTO agent_status (agentId,agentStatus,timestamp,callid) VALUES

 (NEW.agent,'READY',NEW.time,NULL) ON DUPLICATE KEY UPDATE agentStatus = "READY", timestamp = NEW.time, callid = NULL; 
 ELSEIF NEW.event = 'REMOVEMEMBER' THEN
   INSERT INTO `agent_status` (agentId,agentStatus,timestamp,callid) VALUES

(NEW.agent,'LOGGEDOUT',NEW.time,NULL) ON DUPLICATE KEY UPDATE agentStatus = "LOGGEDOUT", timestamp = NEW.time, callid = NULL;
 ELSEIF NEW.event = 'PAUSE' THEN
   INSERT INTO agent_status (agentId,agentStatus,timestamp,callid) VALUES

 (NEW.agent,'PAUSE',NEW.time,NULL) ON DUPLICATE KEY UPDATE agentStatus = "PAUSE", timestamp = NEW.time, callid = NULL; 
 ELSEIF NEW.event = 'UNPAUSE' THEN
   INSERT INTO `agent_status` (agentId,agentStatus,timestamp,callid) VALUES

 (NEW.agent,'READY',NEW.time,NULL) ON DUPLICATE KEY UPDATE agentStatus = "READY", timestamp = NEW.time, callid = NULL;
 ELSEIF NEW.event = 'ENTERQUEUE' THEN
   REPLACE INTO `call_status` VALUES (NEW.callid,replace(replace(substring(substring_index(NEW.data, '|', 2), length(substring_index(New.data, '|', 2 - 1)) + 1), '|', ''), '|', ''),'inQue',NEW.time,NEW.queuename,'','','','',0);
 ELSEIF NEW.event = 'CONNECT' THEN
   UPDATE `call_status` SET callid = NEW.callid,status = NEW.event,timestamp = NEW.time,queue = NEW.queuename,holdtime =

replace(substring(substring_index(NEW.data, '|', 1), length(substring_index(NEW.data, '|', 1 - 1)) + 1), '|', '') where callid = NEW.callid;
INSERT INTO agent_status (agentId,agentStatus,timestamp,callid) VALUES(NEW.agent,NEW.event,NEW.time,NEW.callid)ON DUPLICATE KEY UPDATEagentStatus = NEW.event,timestamp = NEW.time,callid = NEW.callid;
 ELSEIF NEW.event in ('COMPLETECALLER','COMPLETEAGENT') THEN
   UPDATE `call_status` SET callid = NEW.callid,status = NEW.event,timestamp = NEW.time,queue = NEW.queuename,originalPosition =

 replace(substring(substring_index(NEW.data, '|', 3), length(substring_index(NEW.data, '|', 3 - 1)) + 1), '|', ''),holdtime =

replace(substring(substring_index(NEW.data, '|', 1), length(substring_index(NEW.data, '|', 1 - 1)) + 1), '|', ''),callduration = replace(substring(substring_index(NEW.data, '|', 2), length(substring_index(NEW.data, '|', 2 - 1)) + 1), '|', '')

where callid = NEW.callid;
INSERT INTO agent_status (agentId,agentStatus,timestamp,callid) VALUES (NEW.agent,NEW.event,NEW.time,NULL) ON DUPLICATE KEY

UPDATE agentStatus = "READY", timestamp = NEW.time, callid = NULL;
 ELSEIF NEW.event in ('TRANSFER') THEN
   UPDATE `call_status` SET callid = NEW.callid,status = NEW.event,timestamp = NEW.time,queue = NEW.queuename,holdtime = replace(substring(substring_index(NEW.data, '|', 1), length(substring_index(NEW.data, '|', 1 - 1)) + 1), '|', ''),callduration =

 replace(substring(substring_index(NEW.data, '|', 3), length(substring_index(NEW.data, '|', 3 - 1)) + 1), '|', '') where callid = NEW.callid;
INSERT INTO agent_status (agentId,agentStatus,timestamp,callid) VALUES(NEW.agent,'READY',NEW.time,NULL)ON DUPLICATE KEY

UPDATE agentStatus = "READY",timestamp = NEW.time,callid = NULL;
 ELSEIF NEW.event in ('ABANDON','EXITEMPTY') THEN 
   UPDATE `call_status` SET callid = NEW.callid,status = NEW.event,timestamp = NEW.time,queue = NEW.queuename,position = replace(substring(substring_index(NEW.data, '|', 1), length(substring_index(NEW.data, '|', 1 - 1)) + 1), '|', ''),originalPosition =

replace(substring(substring_index(NEW.data, '|', 2), length(substring_index(NEW.data, '|', 2 - 1)) + 1), '|', ''),holdtime = replace(substring(substring_index(NEW.data, '|', 3), length(substring_index(NEW.data, '|', 3 - 1)) + 1), '|', '')

where callid = NEW.callid;
 ELSEIF NEW.event = 'EXITWITHKEY'THEN 
   UPDATE `call_status` SET callid = NEW.callid,status = NEW.event,timestamp = NEW.time,queue = NEW.queuename,position = replace(substring(substring_index(NEW.data, '|', 2), length(substring_index(NEW.data, '|', 2 - 1)) + 1), '|', ''),keyPressed =

 replace(substring(substring_index(NEW.data, '|', 1), length(substring_index(NEW.data, '|', 1 - 1)) + 1), '|', '') where callid = NEW.callid;
 ELSEIF NEW.event = 'EXITWITHTIMEOUT' THEN 
   UPDATE `call_status` SET callid = NEW.callid,status = NEW.event,timestamp = NEW.time,queue = NEW.queuename,position =

replace(substring(substring_index(NEW.data, '|', 1), length(substring_index(NEW.data, '|', 1 - 1)) + 1), '|', '')

where callid = NEW.callid;
END IF;
 END
//
DELIMITER ;

Configuration of Digium TE110p with Asterisk With Difference E1 Provider

2010-08-11T15:53:00.001+05:30

E1 is a physical layer protocol, like ethernet. It defines a 2Mbps link between two endpoints.

T1 is similar to E1. It is used in North America and is 1.544Mbps

A more in depth difference between the two can be found here

A J1 card is the Japanse version of a T1. (the tiny differences between T1 and J1 can be found here

Both T1 and E1 can be used to transmit data or voice, or a mixture of both.

For example: if an E1 is reserved for voice channels only, the 2mbit  will be split into 32 64Kbps telephone channels. 30 of these channels  can carry one telephone conversation each, and 2 carry signalling and  timing information.

A t1 could carry 24 telephone channels, each of which can carry a telephone conversation.

Please note that in the states, its a common practice for carriers to  offer fractional T1s, these have only some of the 24 channels  provisoned.

Lets have a look at the supported audio operating modes:

Supported Audio Modes on E1/T1:

There are a number of protocols which can run on top of E1.

These protocols are grouped into 3 big subgroups, CAS, CCS, RBS.

Lets have a closer look on these subgroups:

a) CAS signalling

CAS stands for Channel Associated Signalling. Examples are FXS

loop start and E&M wink start. These protocols provide information  such as the number that was called, and what state the call is in.  They're limited in what information they can carry, and are slow to set  up.

With this kind of signalling, a set of bits is used to replicate opening  and closing the circuit (as if picking up the telephone receiver and  pulsing digits on a rotary phone), or using tone signalling which is  passed through on the voice circuits themselves

CAS is the original signaling system used by E1. In CAS, channel 16 is  reserved for signaling. In recent years, the term RBS has been replaced  by CAS which is now used to refer to bits that are associated with a  specific channel whether it is in the T1 or E1 format. 

Different subprotocols are:

- E&M

- Wink (this might be only RBS, if you know this for sure, please leave a comment.)

- Feature Group B

- Feature Group D

- FXO & FXS: this seems to only use CAS on E1's

- Ground Start

- Loop Start

- Loop Start with Disconnect Detect

CCS signalling

Common Channel signalling: A  more recent kind of signalling, which  resolves the problems associated with CAS signalling.  In this kind of  signalling, short messages are sent over the signalling channel, with  more information about the call, including caller ID, type of  transmission required, etc. etc.

CCS is used by either T1 or E1 and refers to a system that does not use a  specific bit structure for signaling. Instead, all or part of a channel  is used to pass messages between two systems to indicate how a channel  is being used. 

CCS is used by either T1 or E1 and refers to a system that does not use a  specific bit structure for signaling. Instead, all or part of a channel  is used to pass messages between two systems to indicate how a channel  is being used. This type of system is commonly found in ISDN which uses a  D channel to pass messages.

ISDN signalling and ss7 signalling are a subgroup of CCS signalling.

- ISDN (PRI CPE & PRI NET) 

ISDN uses one channel (called the D channel) for  signalling call information. On E1, this is one of the 2 signalling  channels, leaving 30 channels for voice (called B channels). On T1,  there aren't any spare signalling channels, so one of the voice channels  is used, leaving 23 B channels for voice.

A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it.  ISDN gives fast, reliable call setup and hangup detection, and detailed  information about the call. In the UK, PRI is also called ISDN30.

An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems.

Currently, asterisk has limited support for Q.SIG, it can make and receive calls and retrieve some of the extra information.

- On E1, EuroISDN is the standard for ISDN signalling.

- On T1, there are different standards from different providers:
- NI1

- NI2

- 4ESS (AT&T)

- 5ESS (Lucent)

- DMS100

- SS7 

RBS: Robbed Bit Signalling

RBS is the original signaling system used by T1 and provides either 2 or  4 signaling bits per channel depending on the multiframe format. In  recent years, the term RBS has been replaced by CAS which is now used to  refer to bits that are associated with a specific channel whether it is  in the T1 or E1 format. 

- E&M

- Wink

- Feature Group B

- Feature Group D

- FXO & FXS: this seems to only use RBS on T1's

- Ground Start

- Loop Start

- Loop Start with Disconnect Detect

Framing

AMI, B8ZS, and HDB3 are different types of line coding used in T1 and E1  communications systems. AMI stands for alternate mark inversion and is  used in both T1 and E1 systems. B8ZS stands for Bipolar with 8 Zeros  Substitution and is used in T1 systems while HDB3 stands for  High-Density Bipolar 3 and is used in E1 systems.

- HDB3: High-Density Bipolar 3 -> E1 only 

- AMI: Aternate Mark Inversion -> E1 and T1, the T1 version exists with both ESF (extended super frame) and SF (super frame)

- B8ZS: Bipolar with 8 Zeros Substitution -> T1 only, exists with both ESF (extended super frame) and SF (super frame)

What signalling and framing should i ask my carrier? 

Timing or clock sources

A T1/E1 connection needs a timing device on one of both ends.

A T1/E1 line can be clocked internally or can be clocked by the telco.

Asterisk Compatible E1/T1 cards

available cards

Digium:

- TE110p:  1 port T1/E1 for use in 3.3 or 5 volt pci slots. 

- TE205p: 2 port T1/E1 for use in 5 volt pci slots

- TE210p: 2 port T1/E1 for use in 3.3 volt pci slots

- TE405p: 4 port T1/E1 for use in 5 volt pci slots

- TE410p: 4 port T1/E1 for use in 3.3 volt pci slots

- TE406p:   2nd generation 4 port T1/E1 for use in 5 volt pci slots, with hardware  DTMF recognition and echo cancellation. (now discontinued, but still  supported)

- TE411p:  2nd generation 4 port T1/E1 for use in 3.3 volt pci slots, with  hardware DTMF  recognition and echo cancellation. (now discontinued, but  still supported)

- TE407p:  3rd generation 4 port T1/E1 for use in 5 volt pci slots, with octasic hardware DTMF recognition and echo cancellation.

- TE412p: 3rd generation 4 port T1/E1 for use in 3.3 volt pci slots, with octasic hardware DTMF recognition and echo cancellation. 

- Tormenta 2: Discontinued cards based on the open source project zapatatelephony 

These cards were known as: Wildcard T100P, T400P, E100P, E400P

intel

- Dialogic D/41JCT-LS: quad t1/e1 board, requires additional (paid) drivers from digium to make it work.

sangoma

- A101: One port T1/E1 card

- A102 :Two port T1/E1 card

- A104:  Four port T1/E1 card

varion

- V400P-E: 4 port E1 card, based on the open source zapata telephony project.

These are the same cards as the discontinued digium cards.

- V400P-T: 4 port E1 card, based on the open source zapata telephony project.

These are the same cards as the discontinued digium cards.

Eikon

- none public available yet.

What card should you pick ?

I recommend against using cards based on the tormenta project, they are  way older and take up a lot more cpu. (and their development seems  stalled).

The newest digium cards (TE406p and TE411p) as well as the latest  firmware versions of the te405p and te410p are optimized for speed.

Only the digium TE406p and the digiun TE411p have hardware echo  cancellation,  causing a big difference in cpu usage. But this comes at a  slightly higher cost.

If you need the 40% speed gain with on board echo cancellers, go for  these cards, otherwise go for the cheaper te410p or te405p. (or the 1 or  2 port versions).

It is said on the mailinglists that the hardware echo canceller also has  better quality than the echo cancellation done in software. (I can  confirm nor deny this claim - it's based on a single source).

Asterisk E1/T1 channel drivers

There are two ways to get the E1/T1 cards to work:

First one is chan_zap (requires the zaptel kernel modules), this is recommended for all digium + sangoma cards.

The second one is chan_mISDN (requires the mISDN kernel patches)

This is probably only usefull for eicon cards. chan_misdn is written for BRI cards, but also supports some E1/T1 cards.

Building, installing and configuring asterisk with E1/T1 cards

We will only discuss using chan_zap for now (its the only recommended thing to do).

Tata IndiCom Provider

 Cross cable with tata indicom PRI line

Few Things  need to Conform from Provider

check with your provider if CRC is on
signaling type
Cross Cable or not

 Configuration In 

     zaptel.conf  

   span=1,1,0,ccs,hdb3 

   bchan=1-15,17-31 

   dchan=16 

   loadzone=uk 

   defaultzone=uk

   zapata.conf  

   [trunkgroups] 

   [channels] 

   language=en 

   context=from-zaptel 

   switchtype=euroisdn 

   pridialplan=unknown 

   priindication=outofband 

   usecallerid=yes 

   hidecallerid=no 

   callwaiting=yes 

   usecallingpres=yes 

   threewaycalling=yes 

   transfer=yes 

   cancallforward=yes 

   callreturn=yes 

   callgroup=1 

   pickupgroup=1 

   immediate=no 

   echotraining=800 

   echocancel=yes 

   echocancelwhenbridged=no 

   relaxdtmf=yes 

   musiconhold=default 

   immediate=no 

   txgain=0.0 

   rxgain=0.0 

   group=1 

   signalling=pri_cpe 

   channel=>1-15,17-31 

   overlapdial=yes

Hardware Configuration for  TE110 digium card or Sangoma A101 card 

2 CPU dual core Xeon 2.6/3.0 Ghz 

4 GB RAM 

15 k RPM SAS 4 disks raid 10 or 2 disks raid 1 or single disk raid 0( for recordings) or SATA disk for no recordings 

1000 mbps ethernet NIC 

The motherboard should have a PCI 3.3v/5v slot depending on the card or  you may need to get a PCI express card or a PCI riser card, order the  card after u check the PCI slots on the board 

Airtel

E1 PRI working perfectly fine..

/etc/dahdi/system.conf 

loadzone = uk

defaultzone = uk

span = 1,1,0,ccs,hdb3,crc4 

bchan = 1-15,17-31 

dchan = 16 

/etc/asterisk/chan_dahdi.conf 

[trunkgroups] 

[channels] 

context=airtel 

switchtype=euroisdn

pridialplan=unknown

;prilocaldialplan=unknown 

;overlapdial=yes 

signalling=pri_cpe 

;rxwink=300             ; Atlas seems to use long (250ms) winks 

usecallerid=yes 

hidecallerid=no 

callwaiting=yes 

usecallingpres=yes 

callwaitingcallerid=yes 

threewaycalling=yes 

;transfer=yes 

canpark=yes 

cancallforward=yes 

callreturn=yes 

echocancel=no 

echocancelwhenbridged=no 

rxgain=0.0 

txgain=0.0 

group=1 

inbanddisconnect=yes

callgroup=1 

pickupgroup=1 

immediate=no 

channel => 1-15,17-31

FTP Server Installation and Configuration In Linux Server

2010-08-06T17:48:00.001+05:30

Linux comes with various ftp servers to setup FTP service such as:

proftpd - Versatile, virtual-hosting FTP daemon
vsftpd - The Very Secure FTP Daemon
ftpd - FTP server
wu-ftpd - powerful and widely used FTP server
wzdftpd - A portable, modular, small and efficient ftp server
pure-ftpd - Pure-FTPd FTP server

Install and Configuration of vsftpd 

In centos 

yum install vsftpd 

Configure /etc/vsftpd/vsftpd.conf

# Allow anonymous FTP?
anonymous_enable=YES
...
# The directory which vsftpd will try to change
# into after an anonymous login. (Default = /var/ftp)
anon_root=/data/directory
...
# Uncomment this to allow local users to log in.
local_enable=YES
...
# Uncomment this to enable any form of FTP write command.
# (Needed even if you want local users to be able to upload files)
write_enable=YES
...
# Uncomment to allow the anonymous FTP user to upload files. This only
# has an effect if global write enable is activated. Also, you will
# obviously need to create a directory writable by the FTP user.
#anon_upload_enable=YES
...
# Uncomment this if you want the anonymous FTP user to be able to create
# new directories.
#anon_mkdir_write_enable=YES
...
# Activate logging of uploads/downloads.
xferlog_enable=YES
...
# You may override where the log file goes if you like.
# The default is shown below.
xferlog_file=/var/log/vsftpd.log

...

anon_root=/data/directory   the root directory that visitors will visit

FTP Security

In /etc/vsftpd/ftpusers

For added security, you may restrict FTP access to certain users by adding them to the list of users in the /etc/vsftpd.ftpusers file. The VSFTPD package creates this file with a number of entries for privileged users that normally shouldn't have FTP access. As FTP doesn't encrypt passwords, thereby increasing the risk of data or passwords being compromised, it is a good idea to let these entries remain and add new entries for additional security

Anonymous Upload

If you want remote users to write data to your FTP server, then you  should create a write-only directory within /var/ftp/pub. This will  allow your users to upload but not access other files uploaded by other  users

. The commands you need are: 

# mkdir /var/ftp/pub/upload
# chmod 722 /var/ftp/pub/upload

FTP Greeting Banner

Change the default greeting banner in the vsftpd.conf file to make it  harder for malicious users to determine the type of system you have.  The directive in this file is. 

ftpd_banner= New Banner Here

Start VSFTP

service vsftpd start

Monitoring the Linux Server Using Nagios

2010-08-05T15:41:00.001+05:30

 How to" install, configure and secure your monitoring server the most simply possible.

Download the nagios and plugins
Take care of the prerequisites
Create user and group for nagios
Install nagios
Configure the web interface
Compile and install nagios plugins
Start Nagios
Login to web interface

Download the nagios and plugins

from the website : http://www.nagios.org/download/core/thanks/

Take care of the prerequisites

yum install gcc gd

Create user and group for nagios

useradd nagios

groupadd nagios

usermod -G nagios nagios

usermod -G nagios apache

Install nagios

tar xvf nagios-3.x.x.tar.gz

cd nagios

/configure --with-command-group=nagios

make all

make install

make install-config

make install-commandmode

Configure the web interface.

make install-webconf

htpasswd -c /usr/local/nagios/etc/htpasswd.users nagiosadmin

Compile and install nagios plugins

tar xvf nagios-plugins-1.4.xx.tar.gz

cd nagios-plugins

./configure --with-nagios-user=nagios --with-nagios-group=nagios

make

make install

Start Nagios

chkconfig --add nagios

chkconfig nagios on

command line

/usr/local/nagios/bin/nagios -v /usr/local/nagios/etc/nagios.cfg

service nagios start

Login to web interface

http://localhost/nagios/

Configuration File located at /usr/local/nagios/etc/

nagios.cfg – This is the primary Nagios configuration file where lot of global parameters that controls the nagios can be defined.
cgi.cfg - This files has configuration information for nagios web interface.
resource.cfg – If you have to pass some sensitive information (username, password etc.) to a plugin to monitor a specific service, you can define them here. This file is readable only by nagios user and group.

Following are the other configuration files under /usr/local/nagios/etc/objects directory:

contacts.cfg: All the contacts who needs to be notified should be defined here. You can specify name, email address, what type of notifications they need to receive and what is the time period this particular contact should be receiving notifications etc.
commands.cfg – All the commands to check services are defined here. You can use $HOSTNAME$ and $HOSTADDRESS$ macro on the command execution that will substitute the corresponding hostname or host ip-address automatically.
timeperiods.cfg – Define the timeperiods. for e.g. if you want a service to be monitored only during the business hours, define a time period called businesshours and specify the hours that you would like to monitor.
templates.cfg – Multiple host or service definition that has similar characteristics can use a template, where all the common characteristics can be defined. Use template is a time saver.
localhost.cfg – Defines the monitoring for the local host. This is a sample configuration file that comes with nagios installation that you can use as a baseline to define other hosts that you would like to monitor.
printer.cfg – Sample config file for printer
switch.cfg – Sample config file for switch
windows.cfg – Sample config file for a windows machine

Asterisk Channels Chanspy and Barging for Manager To Monitor the Agents

2010-08-04T16:12:00.001+05:30

Allows eavesdropping on a conversation on any specified channel     (this is different from ZapBarge()/ZapScan()     which are bound to Zap channels only). Note that this application only     listens on single channels, rather than the conversation per     se, even though it does capture incoming and outgoing audio on     the channel.

Chanspy([][,]) 

Valid Options: 

- b: Only spy on channels involved in a bridged call. 

- g(grp): Match only channels where their ${SPYGROUP} variable is set to contain 'grp' in an optional : delimited list. 

- q: Don't play a beep when beginning to spy on a channel, or speak the selected channel name. 

- r[(basename)]: Record the session to the monitor spool directory. An optional base for the filename may be specified. 

  The default is 'chanspy'.     

- v([value]): Adjust the initial volume in the range from -4 to 4. A negative value refers to a quieter setting. 

Since 1.4: 

- w Enable 'whisper' mode, so the spying channel can talk to the spied-on channel. 

- W Enable 'private whisper' mode, so the spying channel can talk to the spied-on 

  channel but cannot listen to that channel.

Since 1.6 

  -b :Only spy on channels involved in a bridged call

   -d  :Override the typical numeric DTMF functionality and instead  use DTMF to switch between spy modes 

            4 -spy mode

            5 -wispher mode

            6 -Bridge mode

 -g :Only spy on channels in which one or more of the groups listed ingrp            matches one or more groups from the SPYGROUP  variable set on the channel to be spied upon.

-n :Say the name of the person being spied on if that person has recorded his/her name. If a context is             specified, then that voicemail context will be searched when retrieving the name, otherwise the default contextbe used when searching for the name (i.e. if SIP/1000 is the channel being

           spied on and no mailbox is specified, then1000 will be used when searching for the name).

-q : Don't play a beep when beginning to spy on a channel, or speak the selected channel name.

-r :Record the session to the monitor spool directory. An optional base for the filename  may be specified. The default is chanspy.

-v : Adjust the initial volume in the range from -4  to 4. A negative value refers to a quieter setting.

-o:Only listen to audio coming from this channel

-X:Allow the user to exit ChanSpy to a valid single digit numeric extension in the current context or the context specified by the SPY_EXIT_CONTEXT channel variable. The name of the last channel that was spied on will be stored in the SPY_CHANNEL variable. 

-s :Skip the playback of the channel type (i.e. SIP, IAX, etc) when speaking the selected channel name

-e :Enable enforced mode, so the spying channel can only monitor extensions whose name is in the ext

-B:Instead of whispering on a single channel barge in on both channels involved in the call. 

exten => 007,1,ChanSpy(SIP/${EXTEN}|dq) 

Linux Monitoring Tools

2010-08-04T15:41:00.002+05:30

Need to monitor Linux server performance? Try these built-in command and  a few add-on tools. Most Linux distributions are equipped with tons of  monitoring. These tools provide metrics which can be used to get  information about system activities. You can use these tools to find the  possible causes of a performance problem. The commands discussed below  are some of the most basic commands when it comes to system analysis and  debugging server

Monitoring the system:

pstree	Processes and parent-child relationships
top	Show top processes
iostat	Report CPU statistics and input/output statistics for devices and partitions.
ps -auxw	process status
uname -a	print system information
cat /proc/version	Display Linux kernel version in use.
cat /etc/redhat-release	Display Red Hat Linux Release. (also /etc/issue)
uptime	Tell how long the system has been running. Also number of users and system's load average.
w	Show who is logged on and what they are doing.
/sbin/lsmod	List all currently loaded kernel modules. Same as cat /proc/modules
/sbin/runlevel	Displays the system's current runlevel.
hostname	Displays/changes the system's node name. (Must also manually change hostname setting in /etc/sysconfig/network. Command will change entry in /etc/hosts)
service	Red Hat/Fedora command to display status of system services. Example: service --status-all Help: service --help

Process Management:

pstree -p
OR
ps -auxw
OR
top
kill <process-id-number>
killall <command-name>

Memory Usage:

vmstat	Monitor virtual memory
free	Display amount of free and used memory in the system. (Also: cat /proc/meminfo)
pamp	Display/examine memory map and libraries (so). Usage: pmap pid
top	Show top processes
sar -B	Show statistics on page swapping.
time -v date	Show system page size, page faults, etc of a process during execution. Note you must fully qualify the command as "/usr/bin/time" to avoid using the bash shell command "time".
cat /proc/sys/vm/freepages	Display virtual memory "free pages". One may increase/decrease this limit: echo 300 400 500 > /proc/sys/vm/freepages
cat /proc/meminfo	Show memory size and usage.

  Filesystems and Storage Devices:

df -k	report filesystem disk space usage. (-k reports in Kbytes)
du -sh	Calculates file space usage for a given directory. (and everything under it) (-s option summarizes)
mount	Displays all mounted devices, their mount point, filesystem, and access. Used with command line arguments to mount file system.
cat /proc/filesystems	Display filesystems currently in use.
cat /proc/mounts	Display mounted filesystems currently in use.
showmount	Displays mount info for NFS filesystems.
cat /proc/swaps	Displays swap partition(s) size, type and quantity used.
cat /proc/ide/hda/any-file	Displays disk information held by kernel.

cfdisk - Curses based disk partition table manipulator. (very nice)
fdisk- Manipulate/configure the partition table.
sfdisk - Manipulate/configure the partition table.
fsck - Manipulate/configure the partition table.

User Info:

who	Displays currently logged in users. Use who -uH for idle time and terminal info.

w	Displays currently logged in users and processes they are running.
whoami	Displays user id.
groups	Display groups you are part of. Use groups user-id to display groups for a given user.
set	Display all environment variables in your current environment.
id	Display user and all group ids. Use id user-id to display info for another user id.
last	Listing of most recent logins by users. Show where from, date and time of login (ftp, ssh, ...) Also see lastlog command. Show last 100 logins: last -100
history	Shell command to display previously entered commands.

Virtualization Software on CentOS using Xen

2010-08-03T16:13:00.002+05:30

Xen is a free open source software virtual machine monitor for Intel / PowerPC architectures. It is software that runs on a host operating system and allows several guest operating systems to be run on top of the host on the same computer hardware at the same time (just like VMWare software). Luckily Installing and managing XEN is quite easy under CentOS 5 Linux. 

Xen lets you create guest operating systems (*nix operating systems like Linux and FreeBSD), so called "virtual machines" or domUs, under a host operating system (dom0). Using Xen you can separate your applications into different virtual machines that are totally independent from each other (e.g. a virtual machine for a mail server, a virtual machine for a high-traffic web site, another virtual machine that serves your customers' web sites, a virtual machine for DNS, etc.), but still use the same hardware. This saves money, and what is even more important, it's more secure. If the virtual machine of your DNS server gets hacked, it has no effect on your other virtual machines. Plus, you can move virtual machines from one Xen server to the next one.

I will use CentOS 5.0 (i386) for both the host OS (dom0) and the guest OS (domU).

yum install xen virt-manager kernel-xen

chkconfig xend on

reboot 

Before Reboot 

Change the file /boot/grub/menu.lst then you should probably use 

to default =0

After Login 

type command 

uname -r

2.6.18-8.1.4.el5xen

Xen commands

xm list

CentOS comes with a nice tool called virt-install with which we can create virtual machines for Xen. To start it, we simply run

virt-install 

What is the name of your virtual machine? <-- vm01

How much RAM should be allocated (in megabytes)? <-- 256

What would you like to use as the disk (path)? <-- /vm/vm01.img

How large would you like the disk (/vm/vm01.img) to be (in gigabytes)? <-- 4

Would you like to enable graphics support? (yes or no) <-- no

What is the install location? <-- http://wftp.tu-chemnitz.de/pub/linux/centos/5.0/os/i386

virt-install has created the vm01 configuration file /etc/xen/vm01 for us (in dom0).

It should look like this:  # Automatically generated xen

config file name = "vm01" memory = "256" disk = [ 'tap:aio:/vm/vm01.img,xvda,w', ] vif = [ 'mac=00:16:3e:13:e4:81, bridge=xenbr0', ]  uuid = "5aafecf1-dd66-401d-69cc-151c1cb8ac9e" bootloader="/usr/bin/pygrub" vcpus=1 on_reboot   = 'restart' on_crash    = 'restart'

to login into

vm01  xm console vm01

To shut down vm01,

do this: xm shutdown vm01

To start vm01 again,

run xm create /etc/xen/vm01

If you want vm01 to start automatically at the next boot of the system,

then do

this:  ln -s /etc/xen/vm01 /etc/xen/auto

Here are the most important Xen commands:

xm create -c /path/to/config - Start a virtual machine. xm shutdown  - Stop a virtual machine.

xm destroy  - Stop a virtual machine immediately without shutting it down. It's as if you switch off the power button.

xm list - List all running systems.

xm console  - Log in on a virtual machine.

Linux BackUp Tools

2010-08-03T15:08:00.001+05:30

If you’ve ever lost data due to a system crash, you know how crucial backing up important files can be. Here are 5 Linux Backup Solutions you should check out. I recommend you implement at least one of these Linux Backup Solutions before it’s too late.

rsync

There are tons of Linux users and administrators out there who have customized rsync scripts to handle incremental backups automatically on a daily, weekly, or monthly schedule. From the manual, rsync is described as a program that behaves in much the same way that rcp does, but has many more options and uses the rsync remote-update protocol to greatly speed up file transfers when the destination file is being updated. The rsync remote-update protocol allows rsync to transfer just the differences between two sets of files across the network connection, using an efficient checksum-search algorithm described in the technical report that accompanies this package.

In centos 

yum -y install rsync xinetd

Add xinetd service to system

# chkconfig --add xinetd

Make sure xinetd running on init 3 and 5

# chkconfig --list xinetd

Enable rsync

# vi /etc/xinetd.d/rsync

Change disable = yes into disable = no

Create username and password for rsync client to use

# vi /etc/rsyncd.secrets

root:hispassword

Create configuration and shares for rsync daemon

# vi /etc/rsyncd.conf

———————-

max connections = 2

log file = /var/log/rsync.log

timeout = 300

[shares]

comment = shared data stored here

path = /home/adminname/shares

read only = false

list = yes

uid = root

gid = root

auth users = root

secrets file = /etc/rsyncd.secrets

hosts allow = mac1.hostname.com mac2.hostname.com mac3.hostname.com

———————-

Secure /etc/rsyncd.*

# chown root.root /etc/rsyncd.*

# chmod 600 /etc/rsyncd.*

Restart xinetd

# service xinetd restart

Make sure rsync now running

# chkconfig --list

testing 

rsync -a -e ssh /www/backup.tar.gz root@mac2.hostname.com:/home/backup

mondorescue

Mondorescue backs up your GNU/Linux server or workstation to tape, CD-R, CD-RW, DVD-R[W], DVD+R[W], NFS or hard disk partition. In the event of catastrophic data loss, you will be able to restore all of your data [or as much as you want], from bare metal if necessary. Personally, I like using Mondorescue to create DVD disk images of my system periodically. Upon initial installation and configuration of my Linux or Windows machine, I create a DVD disk image with Mondo so that if anything ever gets screwed up, I can pop in the DVD disk and restore back to my original configuration. The mondorescue team is great and the lead developers of the project are very active on the public mailing list offering help to normal users whenever needed.

For CentOS Downloads

http://mondorescue.muskokamug.org/rhel/5/

rpm -ivh afio

rpm -ivh buffer

yum install mkisofs

rpm -ivh mindi-busybox

rpm -ivh mindi

rpm –ivh mondo

After Installation 

Rum commands mondoarchive 

Amanda

AMANDA, the Advanced Maryland Automatic Network Disk Archiver, is a backup system that allows the administrator to set up a single master backup server to back up multiple hosts over network to tape drives/changers or disks or optical media. Amanda uses native dump and/or GNU tar facilities and can back up a large number of workstations running multiple versions of Unix. Amanda uses Samba or Cygwin to back up Microsoft Windows desktops and servers.

For CentOS

Step 1:

yum -y install amanda*

Step 2:

after install Modify 

/etc/xinetd.d/amanda

/etc/xinetd.d/amandaidx

/etc/xinetd.d/amidxtape

Change Disable = yes to Disable = no.

Step 3:

cp -r /etc/amanda/DailySet1 /etc/amanda/intra

Step 4:

Edit Your Amanda.conf file

/etc/amanda/intra/amanda.conf

Step 5:

Edit /etc/amanda/intra/disklist

Step 6:

Edit hosts file: /etc/hosts

Step 7:

Create backup directory

mkdir -p -m 770 /backup/intra/slots

chown -R amanda:disk /backup

Step 8:

Create tape File

touch /etc/amanda/intra/tapelist

Step 9:

Create slots

su - amanda

cd /backup/intra/slots

for ((i=1; $i<=15; i++)); do mkdir slot$i; done

ln -s slot1 data

Step 10:

Test virtual tapes: 

/usr/sbin/ammt -f file:/backup/intra/slots status

Step 11:

Edit .amandahosts to allow communications from clients:

/var/lib/amanda/.amandahosts

Step 12:

Start service

service xinetd start

chkconfig --list

BroadCast Voice message through Asterisk

2010-06-18T09:56:00.000+05:30

Broad casting Message through Asterisk

Creating a call file to call 1000 context default in the sip.conf file

; set the caller ID to sales call placing the call

; A correct Caller ID may be required when using some VOIP to PSTN gateway services.

;

; cat < /var/spool/asterisk/tmpoutgoing

; Channel: SIP/1000

; Callerid: sales call

; MaxRetries: 5

; RetryTime: 300

; WaitTime: 45

; Context: default

; Extension: 1234

; Priority: 1

; EOF

; mv /var/spool/asterisk/tmp01 /var/spool/asterisk/outgoing





This is a sample section from an extensions.conf file showing how to auto-dial an outbound కాల్



[default]



exten=> 1234,1,Answer()

exten => 1234,2,Wait(1)

exten => 1234,3,Background(welcome to company)         ; "play outbound msg"

exten => 1234,4,Background(preview )   ; "Press 1 to replay or 2 to acknowledge receiving this మెసేజ్ ప్రెస్ ౯ తో కాంటాక్ట్ యువర్ agent"

exten => 1,1,Goto(default ,1234,3)   ; replay message

exten => 2,1,Goto(bye,1,1) ; acknowledge message

exten => t,1,Playback(vm-goodbye) timeout  play bye

exten => t,2,hangup hangup





[bye]



exten=> 1,Playback(thank u for your valuable time)

exten=> 2,palyback(goodbye)

exten => 3,Hangup()





If you have good knowledge in agi scripting



through orignate commands



Agi scritping using agi::asterisk and net ::telenet



#!/usr/bin/perl





# Namespace Declarartion

#####################################################################################################

use Asterisk::AGI;

use Net::Telnet;



$US='_';

######################################################################################################



$AGI = new Asterisk::AGI;



 $|=1;

while() {

       chomp;

       last unless length($_);

       if ($V)

      {



               if (/^agi_(\w+)\:\s+(.*)$/)

               {

                      $AGI{$1} = $2;

               }

       }



       if (/^agi_uniqueid\:\s+(.*)$/)          {$unique_id = $1;}

       if (/^agi_channel\:\s+(.*)$/)           {$channel = $1;}

       if (/^agi_extension\:\s+(.*)$/)         {$extension = $1;}

       if (/^agi_type\:\s+(.*)$/)              {$type = $1;}

       if (/^agi_callerid\:\s+(.*)$/)          {$callerid = $1;}

       if (/^agi_context\:\s+(.*)$/)           {$context = $1;}



}







&Originate();





sub Originate

{



	$Username       = "cron";

	$Secret         = "1234";

	$MyPhone        = "SIP/1000";

	$MyPhoneEscaped = "SIP\\/1000";



	# Connect to localhost via telnet on the Asterisk manager port.

	$TelnetClient = new Net::Telnet (Timeout => 10,

									 Errmode => "die",

					  Host    => "localhost",

					  Port    => 5038);



	# Login with our username and secret.

	$TelnetClient->open  ();

	$TelnetClient->print ("Action: Login");

	$TelnetClient->print ("Username: $Username");

	$TelnetClient->print ("Secret: $Secret");

	$TelnetClient->print ("");



	# Check for login success.

	($Prematch, $Match) = $TelnetClient->waitfor ("/Message: .*/");





	if (($Prematch =~ m/Success/) && ($Match =~ m/Authentication/)) {

	  print "Good authentication.\n";

	}

	else {

	  print "Authentication failed.\n";

	  exit;

	} 





    print "Dialing $CurrentNumber...\n";

   $TelnetClient->print ("Action: Originate");

   $TelnetClient->print ("Channel: $MyPhone");

   $TelnetClient->print ("Exten: 1234");

   $TelnetClient->print ("Priority: 2");

   $TelnetClient->print ("Callerid: \"nagarjuna\" <$CurrentNumber>");

   $TelnetClient->print ("");



   ($Prematch, $Match) = $TelnetClient->waitfor ("/Message: .*/");



   if (($Match =~ m/Originate/) && ($Prematch =~ m/Success/)) {

     $OriginateSuccess = 1;

     print "Origination success.\n";

   }

   else {

     print "Origination failure.\n";

     sleep (2);

   }



	 print "Waiting...\n";



 $OnThePhone = 1;



 while ($OnThePhone) {

   ($Prematch, $Match) = $TelnetClient->waitfor (match   => "/Uniqueid: .*/",

                                                 timeout => 60);

   print "Event detected.\n";



   if (($Prematch =~ m/Hangup/) && ($Prematch =~ m/$MyPhoneEscaped/)) {

     $OnThePhone = 0;

   }

 }

}

##################################Check 200 Reply ####################################



sub checkresult {

       my ($res) = @_;

       my $retval;

       $tests++;

       chomp $res;

       if ($res =~ /^200/) {

               $res =~ /result=(-?\d+)/;

               if (!length($1)) {

                       print STDERR "FAIL ($res)\n";

                       $fail++;

               } else {

                       print STDERR "PASS ($1)\n";

                       $pass++;

               }

       }

        else {

               print STDERR "FAIL (unexpected result '$res')\n";

               $fail++;

              }

}

noika code

2010-01-01T14:55:00.000+05:30

just note, I found from http://www.programmerworld.net/articles/tips/nokia_codes.ఫ్ప్
* *3370# This Nokia code activates Enhanced Full Rate Codec (EFR) - Your Nokia cell phone uses the best sound quality but talk time is reduced my approx. 5% *
#3370# Deactivate Enhanced Full Rate Codec (EFR) .
* *#4720# Activate Half Rate Codec - Your phone uses a lower quality sound but you should gain approx 30% more Talk Time.
* *#4720# With this Nokia code you can deactivate the Half Rate Codec.
* *#0000# Displays your phones software version, 1st Line : Software Version, 2nd Line : Software Release Date, 3rd Line : Compression Type .
* *#9999# Phones software version if *#0000# does not work.
* *#06# For checking the International Mobile Equipment Identity (IMEI Number) .
* #pw+1234567890+1# Provider Lock Status. (use the "*" button to obtain the "p,w" and "+" symbols).
* #pw+1234567890+2# Network Lock Status. (use the "*" button to obtain the "p,w" and "+" symbols) .
* #pw+1234567890+3# Country Lock Status. (use the "*" button to obtain the "p,w" and "+" symbols).
* #pw+1234567890+4# SIM Card Lock Status. (use the "*" button to obtain the "p,w" and "+" symbols) .
* *#147# This lets you know who called you last (Only vodofone).
* *#1471# Last call (Only vodofone) .
* *#21# This phone code allows you to check the number that "All Calls" are diverted to.
* *#2640# Displays phone security code in use .
* *#30# Lets you see the private number.
* *#43# Allows you to check the "Call Waiting" status of your cell phone.
* *#61# Allows you to check the number that "On No Reply" calls are diverted to .
* *#62# Allows you to check the number that "Divert If Unreachable (no service)" calls are diverted to .
* *#67# Allows you to check the number that "On Busy Calls" are diverted తో
* *#67705646# Phone code that removes operator logo on 3310 & 3330
* *#73# Reset phone timers and game scores.
* *#746025625# Displays the SIM Clock status, if your phone supports this power saving feature "SIM Clock Stop Allowed", it means you will get the best standby time possible .
* *#7760# Manufactures code.
* *#7780# Restore factory settings.
* *#8110# Software version for the nokia 8110.
* *#92702689# Displays - 1.Serial Number, 2.Date Made, 3.Purchase Date, 4.Date of last repair (0000 for no repairs), 5.Transfer User Data. To exit this mode you need to switch your phone off then on again.
* *#94870345123456789# Deactivate the PWM-Mem.
* **21* number# Turn on "All Calls" diverting to the phone number entered.
* **61* number# Turn on "No Reply" diverting to the phone number entered .
* **67* number# Turn on "On Busy" diverting to the phone number entered .
* 12345 This is the default security code . * press and hold # Lets you switch between lines.

Asterisk integration with Opensips

2010-01-01T14:51:00.000+05:30

Only voicemail and conference services do require DB support. While the conference DB table will be exclusively be used by Asterisk (OpenSIPS does not require any information form there), the voicemail service do require a tight sharing of DB information about users between OpenSIPS and Asterisk.

Considering OpenSIPS as the core SIP element of the platform, the core DB will be also belonging to OpenSIPS - we will use the OpenSIPS DB to drive both OpenSIPS and Asterisk. The tables required by Asterisk (for voicemail service) will be mapped over the OpenSIPS tables.

This approach assumes two steps:

1. creating the OpenSIP tables and extend them to contain also the fields (info) required by Asterisk.
2. creating the mysql views over the OpenSIPS tables, in order to simulate the Asterisk tables

Creating the OpenSIPS tables

In file /etc/opensips/opensipsctlrc enable the mysql backend (see DBENGINE) and also configure the host where the mysql server is located, the name to be used for creating the opensips table, the read-only (ro) and read-write (rw) usernames and passwords to be created for accessing the opensips DB - see all the varaibles with DB prefix in the file.

Proceed with creation of the OpenSIPS standard database:

$ opensipsdbctl create

If you use the default values to DB host and mysql access users, you can access the DB by:

$ mysql -h'localhost' -u'opensips' -p'opensipsrw' opensips
>

Once the default tables are created, we need to alter the subscriber table in order to add some additional fields required by the Asterisk DB view. These changes are:

* add vm_password to be used as voicemail pin
* add first_name and last_name as subscriber name
* add email_address as subscriber's email address
* add datetime_created as timestamp of the subscriber creation.

Login to the opensips database (use the above login command) and run:

> alter table subscriber add column `vmail_password` varchar(8) NOT NULL default '1234';
> alter table subscriber add column `first_name` varchar(25) NOT NULL default '';
> alter table subscriber add column `last_name` varchar(45) NOT NULL default '';
> alter table subscriber add column `email_address` varchar(50) NOT NULL default '';
> alter table subscriber add column `datetime_created` datetime NOT NULL default '0000-00-00 00:00:00';

Creating the Asterisk tables

Create a separate database to be used by Asterisk - login as root into mysql server and run:

> create database asterisk;
> GRANT ALL PRIVILEGES ON asterisk.* TO 'asterisk' IDENTIFIED BY 'asterisk_pwd';

Create the Asterisk tables which are exclusively used only by Asterisk (as mysql tables):

# create table for the meetme service
CREATE TABLE `meetme` (
`confno` varchar(80) NOT NULL default '0',
`username` varchar(64) NOT NULL default '',
`domain` varchar(128) NOT NULL default '',
`pin` varchar(20) default NULL,
`adminpin` varchar(20) default NULL,
`members` int(11) NOT NULL default '0',
PRIMARY KEY (`confno`)
) ENGINE=MyISAM

# create table to store the voicemail massages
CREATE TABLE `voicemessages` (
`id` int(11) NOT NULL auto_increment,
`msgnum` int(11) NOT NULL default '0',
`dir` varchar(80) default '',
`context` varchar(80) default '',
`macrocontext` varchar(80) default '',
`callerid` varchar(40) default '',
`origtime` varchar(40) default '',
`duration` varchar(20) default '',
`mailboxuser` varchar(80) default '',
`mailboxcontext` varchar(80) default '',
`recording` longblob,
PRIMARY KEY (`id`),
KEY `dir` (`dir`)
) ENGINE=MyISAM

Create the Asterisk tables (as mysql views) that import the information from OpenSIPS tables:

# create the asterisk users tables as a view over the OpenSIPS subscriber table
CREATE VIEW `asterisk`.`sipusers` AS select
`opensips`.`subscriber`.`username` AS `name`,
`opensips`.`subscriber`.`username` AS `username`,
_latin1'friend' AS `type`,
NULL AS `secret`,
`opensips`.`subscriber`.`domain` AS `host`,
concat(`opensips`.`subscriber`.`rpid`,_latin1' ',_latin1'<',`opensips`.`subscriber`.`username`,_latin1'>') AS `callerid`,
_latin1'default' AS `context`,
`opensips`.`subscriber`.`username` AS `mailbox`,
_latin1'yes' AS `nat`,
_latin1'no' AS `qualify`,
`opensips`.`subscriber`.`username` AS `fromuser`,
NULL AS `authuser`,
`opensips`.`subscriber`.`domain` AS `fromdomain`,
NULL AS `insecure`,
_latin1'no' AS `canreinvite`,
NULL AS `disallow`,
NULL AS `allow`,
NULL AS `restrictcid`,
`opensips`.`subscriber`.`domain` AS `defaultip`,
`opensips`.`subscriber`.`domain` AS `ipaddr`,
_latin1'5060' AS `port`,
NULL AS `regseconds`
from `opensips`.`subscriber`;

# create the asterisk voceimail users table as a view over the OpenSIPS subscriber table
CREATE VIEW `asterisk`.`vmusers` AS select
concat(`opensips`.`subscriber`.`username`,`opensips`.`subscriber`.`domain`) AS AS `uniqueid`,
`opensips`.`subscriber`.`username` AS `customer_id`,
_latin1'default' AS `context`,
`opensips`.`subscriber`.`username` AS `mailbox`,
`opensips`.`subscriber`.`vmail_password` AS `password`,
concat(`opensips`.`subscriber`.`first_name`,_latin1' ',`opensips`.`subscriber`.`last_name`) AS `fullname`,
`opensips`.`subscriber`.`email_address` AS `email`,
NULL AS `pager`,
`opensips`.`subscriber`.`datetime_created` AS `stamp`
from `opensips`.`subscriber`;

#create the asterisk voicemail aliases table as a view over the OpenSIPS dbaliases table
CREATE VIEW `asterisk`.`vmaliases` AS select
`opensips`.`dbaliases`.`alias_username` AS `alias`,
_latin1'default' AS `context`,
`opensips`.`dbaliases`.`username` AS `mailbox`
from `opensips`.`dbaliases`;

Configure the access for Asterisk to the created database (via unixodbc):

* in etc/odbcinst.ini

[MySQL]
Description = MySQL driver
Driver = /usr/lib/odbc/libmyodbc.so
Setup = /usr/lib/odbc/libodbcmyS.so
CPTimeout =
CPReuse =
UsageCount = 1

* in /etc/odbc.ini

[MySQL-asterisk]
Description = MySQL Asterisk database
Trace = Off
TraceFile = stderr
Driver = MySQL
SERVER = localhost
USER = asterisk
PASSWORD = asterisk_pwd
PORT = 3306
DATABASE = asterisk

* in /etc/asterisk/res_odbc.conf

[asterisk]
enabled => yes
dsn => MySQL-asterisk
username => asterisk
password => asterisk_pwd
pre-connect => yes

* in /etc/asterisk/extconfig.conf

sipusers => odbc,asterisk,sipusers
sippeers => odbc,asterisk,sipusers
voicemail => odbc,asterisk,vmusers
meetme => odbc,asterisk,meetme

1.7 Asterisk dialplan

The following extensions are set for Asterisk:

1. VMR_ prefix indicates that the caller (identify by SIP FROM header) leaves a voicemail message to the user indicated by the RURI (after the prefix).
2. VM_pickup RURI indicates that the caller (identify by SIP FROM header) accesses his own voicemailbox IVR (to listen his messages); the access is done directly, without any PIN required from Asterisk
3. AN_notavailable plays the "Not Available" audio message
4. AN_time plays the current time message (for the given timezone)
5. AN_date plays the current time message (for the given timezone)
6. AN_echo accesses the echo audio service
7. CR_ prefix indicates access to a conference room; the number of the room is part of the RURI, after the prefix (like CR_3322); Asterisk will perform the checks for conference room existence and for the access PIN code.

Set in /etc/asterisk/extensions.conf' :

[general]
static=yes
writeprotect=no

[default]

; Voicemail
exten => _VMR_.,n,Ringing
exten => _VMR_.,n,Wait(1)
exten => _VMR_.,n,Answer
exten => _VMR_.,n,Wait(1)
exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
exten => _VMR_.,n,Hangup

; Allow users to call their Voicemail directly
exten => VM_pickup,n,Ringing
exten => VM_pickup,n,wait(1)
exten => VM_pickup,n,VoicemailMain(${CALLERIDNUM}|s)
exten => VM_pickup,n,Hangup

; announcement: not available
exten => AN_notavailable,1,Ringing
exten => AN_notavailable,2,Playback(notavailable)
exten => AN_notavailable,3,Hangup

; announcement: time
exten => AN_time,1,Ringing
exten => AN_time,2,Wait(1)
exten => AN_time,3,SayUnixTime(,Europe/Bucharest,HMp)
exten => AN_time,4,Hangup

; announcement:date
exten => AN_date,1,Ringing
exten => AN_date,2,SayUnixTime(,Europe/Bucharest,ABdY)
exten => AN_date,3,Hangup

; announcement: echo
exten => AN_echo,1,Ringing
exten => AN_echo,2,Answer
exten => AN_echo,3,Echo

; Conference service
exten => _CR_.,1,Ringing
exten => _CR_.,n,Wait(1)
exten => _CR_.,n,MeetMe(${EXTEN:3}|Mi)

1.8 OpenSIPS configuration

In this example we take the OpenSIPS default config file that provides user registration and authentication against the DB and extends the script for adding the following media oriented services:

1. voicemail
1. leaving a message - if the called user is not registered on OpenSIPS, the call will be forwarded by OpenSIPS (by adding VMR_ prefix) to Asterisk
2. listening messages -if a subscriber calls to the *1111 number, OpenSIPS will rewrite it to VM_pickup and send it to Asterisk
2. announcements
1. service number *2111 for listening the current time message - OpenSIPS will rewrite it to AN_time and send it to Asterisk
2. service number *2112 for listening the current date message - OpenSIPS will rewrite it to AN_date and send it to Asterisk
3. service number *2113 for accessing the echo service - OpenSIPS will rewrite it to AN_echo and send it to Asterisk
3. conference
1. service number *3XXX for dialling into the conference room XXX - OpenSIPS will rewrite it to CR_XXX and send it to Asterisk

In /etc/opensips/opensips.cfg place (see the ASTERISK_HOOKS markers to find the script parts relevant to Asterisk integration):

####### Global Parameters #########

debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes

port=5060

/* uncomment and configure the following line if you want opensips to
bind on a specific interface/port/proto (default bind on all available) */
#listen=udp:192.168.1.2:5060

####### Modules Section ########

#set module path
mpath="/lib/opensips/modules/"

loadmodule "db_mysql.so"
loadmodule "signaling.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "uri_db.so"
loadmodule "uri.so"
loadmodule "xlog.so"
loadmodule "acc.so"
loadmodule "auth.so"
loadmodule "auth_db.so"
loadmodule "domain.so"

# ----------------- setting module-specific parameters ---------------

# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")

# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)

# ----- usrloc params -----
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")

# ----- uri_db params -----
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "db_url", "")

# ----- acc params -----
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* account triggers (flags) */
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
/* uncomment the following lines to enable DB accounting also */
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)

# ----- auth_db params -----
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("auth_db", "load_credentials", "")

# ----- domain params -----
modparam("domain", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("domain", "db_mode", 1) # Use caching

# ----- multi-module params -----
/* uncomment the following line if you want to enable multi-domain support
in the modules (dafault off) */
modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1)

####### Routing Logic ########

# main request routing logic

route{

if (!mf_process_maxfwd_header("10")) {
send_reply("483","Too Many Hops");
exit;
}

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ->
# ignore and discard
exit;
}
}
send_reply("404","Not here");
}
exit;
}

#initial requests

# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans())
t_relay();
exit;
}

t_check_trans();

# authenticate if from local subscriber
if (!(method=="REGISTER") && is_from_local()) {
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!check_from()) {
send_reply("403","Forbidden auth ID");
exit;
}

consume_credentials();
# caller authenticated
}

# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
send_reply("403","Preload Route denied");
exit;
}

# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();

# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}

# if not a targetting a local SIP domain, just send it out
# based on DNS (calls to foreign SIP domains)
if (!is_uri_host_local()) {
append_hf("P-hint: outbound\r\n");
route(1);
}

# requests for my domain

if (is_method("REGISTER")) {
# authenticate the REGISTER requests
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
exit;
}
if (!check_to()) {
send_reply("403","Forbidden auth ID");
exit;
}

if (!save("location"))
sl_reply_error();

exit;
}

if ($rU==NULL) {
# request with no Username in RURI
send_reply("484","Address Incomplete");
exit;
}

# ASTERISK HOOK - BEGIN
# media service number? (digits starting with *)
if ($rU=~"^\*[1-9]+") {
# we do provide access to media services only to our
# subscribers, who were previously authenticated
if (!is_from_local()) {
send_reply("403","Forbidden access to media service");
exit;
}
#identify the services and translate to Asterisk extensions
if ($rU=="*1111") {
# access to own voicemail IVR
seturi("sip:VM_pickup@ASTERISK_IP:ASTERISK_PORT");
} else
if ($rU=="*2111") {
# access to the "say time" announcement
seturi("sip:AN_time@ASTERISK_IP:ASTERISK_PORT");
} else
if ($rU=="*2112") {
# access to the "say date" announcement
seturi("sip:AN_date@ASTERISK_IP:ASTERISK_PORT");
} else
if ($rU=="*2113") {
# access to the "echo" service
seturi("sip:AN_echo@ASTERISK_IP:ASTERISK_PORT");
} else
if ($rU=~"\*3[0-9]{3}") {
# access to the conference service
# remove the "*3" prefix and place the "CR_" prefix
strip(2);
prefix("CR_");
rewritehostport("ASTERISK_IP:ASTERISK_PORT");
} else {
# unknown service
seturi("sip:AN_notavailable@ASTERISK_IP:ASTERISK_PORT");
}
# after setting the proper RURI (to point to corresponding ASTERISK extension),
# simply forward the call
t_relay();
exit;
}
# ASTERISK HOOK - END

# do lookup
if (!lookup("location")) {
# ASTERISK HOOK - BEGIN
# callee is not registered, so different to Voicemail
# First add the VM recording prefix to the RURI
prefix("VMR_");
# forward to the call to Asterisk (replace below with real IP and port)
rewritehostport("ASTERISK_IP:ASTERISK_PORT");
route(1);
# ASTERISK HOOK - END
exit;
}

# when routing via usrloc, log the missed calls also
setflag(2);

# arm a failure route in order to catch failed calls
# targeting local subscribers; if we fail to deliver
# the call to the user, we send the call to voicemail
t_on_failure("1");

route(1);
}

route[1] {
if (!t_relay()) {
sl_reply_error();
};
exit;
}

failure_route[1] {
if (t_was_cancelled()) {
exit;
}

# if the failure code is "408 - timeout" or "486 - busy",
# forward the calls to voicemail recording
if (t_check_status("486|408")) {
# ASTERISK HOOK - BEGIN
# First revert the RURI to get the original user in RURI
# Then add the VM recording prefix to the RURI
revert_uri();
prefix("VMR_");
# forward to the call to Asterisk (replace below with real IP and port)
rewritehostport("ASTERISK_IP:ASTERISK_PORT");
t_relay();
# ASTERISK HOOK - END
exit;
}
}

Asterisk 1.6 Installation on Centos

2009-12-11T16:29:00.000+05:30

INTRODUCTION

Asterisk is a complete PBX in software. It runs on Linux BSD And Windows (emulated) and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services. Asterisk needs no additional hardware for Voice-over-IP, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism (for certain applications such as conferencing). A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers) DEPENDENCIES Install Centos 5.3 from net Install cd Asterisk Prerequests

Linux Package Requirements

ODBC: unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel libtool.

Asterisk:

C ++ : gcc gcc-c++

Kernel Prerequests : kernel-devel bison bison-devel gnutls-devel openssl openssl-devel libtermcap-devel ncurses ncurses-devel doxygen curl-devel libidn and libidn-devel .

Zttool: newt-devel.

DUNDi: zlib zlib-devel

Mysql comnnector -yum install mysql-connector-odbc

Asterisk 1.6.2 and Higher

Yum install libxml2 libxml2-python libxml2-devel

### INSTALLING ZAPTEL ###

cd /usr/src/zaptel

make clean
make
make install
make config

If you have any zaptel hardware it is now recommended to edit /etc/default/zaptel or /etc/sysconfig/zaptel and set there an optimal value for the variable MODULES ..

chkconfig zaptel on
service zaptel start

### INSTALL LIBPRI ###

cd /usr/src/libpri

make clean
make
make install

### INSTALLING ASTERISK###

cd /usr/src/asterisk/asterisk-VERSION
make clean
./configure

Choose which options to install (voicemail storage, codecs etc.)

make menuselect
make
make install

Install sample files in /etc/asterisk, install docs/manpages and set to start on boot.

make samples
make config
chkconfig asterisk on
service asterisk start

How to uninstall Asterisk...

First you need stop Asterisk and unload its modules that it may be using, e.g Zaptel’s.

You have to kill safe_asterisk first then asterisk second. (The safe_asterisk script is the preferred way to run Asterisk for most of us. This script starts Asterisk and monitors it to make sure it is still running. If the Asterisk process dies, the script will attempt to restart it.)

killall -9 safe_asterisk

killall -9 asterisk

Then you’ll have to unload the Zaptel drivers; check which ones are loaded by issuing a:

# lsmod | grep zaptel

zaptel 214820 2 wcfxo,wctdm

crc_ccitt 2113 1 zaptel

This means that the submodules wcfxo and wctdm are loaded for zaptel. You will have to remove them in reverse order:

modprobe -r wcfxo

modprobe -r wctdm

..repeat for all zaptel submodules….

modprobe -r zaptel

If you repeat the lsmod | grep zaptel command now, it should find nothing.

Now to Delete Asterisk files :-
rm -rf /etc/asterisk

rm -f /etc/zaptel.conf

rm -rf /var/log/asterisk

rm -rf /var/lib/asterisk

rm -rf /var/spool/asterisk

rm -rf /usr/lib/asterisk

Now Asterisk has been removed from your system