Monday, January 31, 2011

Creating Secure Site in Tomact and Apache

Apache 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. 

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

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. 

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
 

Thursday, January 20, 2011

FreeSwitch Installation Steps in centos and Debain

     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

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

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);
                }
       }
}
#==================================================



Friday, December 10, 2010

Asterisk Integration With Openldap

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

Monday, November 15, 2010

Asterisk Queues Log In to Database

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 ;





Wednesday, August 11, 2010

Configuration of Digium TE110p with Asterisk With Difference E1 Provider

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