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10/25/2020
WTH is Asterisk !!
1
Asterisk Server
Internet
Softphone/ Hard phone Softphone/ Hard phone
1.1 Basic Call Routing
10/25/2020 2
VOIP
10/25/2020
Definition
1. Voice Over Internet Protocol
2. group of technologies for the delivery of voice communications and multimedia
sessions over Internet Protocol network.
Why Use It ?
1. Cost Effective
2. Functionality
How does it Work ?
1.2 VoIP Routing
3
IP Phone Ethernet Router Internet Router Ethernet IP Phone
Server Server
SIP
10/25/2020
Definition
1. Session Initiation Protocol
2. signaling Protocol for IP communication.
1. initiating, maintaining, modifying and terminating real-time sessions.
3. SIP itself does not provide communication services.
Sends IP Package
IP Communication
Laptop(IP device 1) Phone(IP device 2)
IP network
4
10/25/2020
IP Communication
1. Location (IP Address)
2. Same Codec (For Encoding-Decoding)
SIP
Locate
other
party
How to
construc
t & Send
Packets
How to
Manage
Sessions
Which
Codec
1.3 SIP
5
10/25/2020
SDP
Definition
1. Session description Protocol
2. When SIP packets are send and received the both parties need to know
the media details(transport address, transport address, transport
protocol, codec, ports and other session description metadata).
3. SDP is used to advertise such information.
1.4 A SIP Packet
6
10/25/2020
Asterisk
1. It is a complete PBX(Private Branch exchange) in software.
2. Can Run on Windows, Linux, OS X.
3. How to install Asterisk ?
What Can we do with Asterisk ?
1. Call Conferencing
2. intercom calling extension to extension
3. Interactive Voice Response(IVR)
4. Call Queuing etc.
7
10/25/2020
1.5 A Basic Call Flow (Outgoing)
User Mobile Phone
Asterisk ServerIP phone Cellular Tower
8
10/25/2020
Converts 3G,4G,GSM to SIP
Calling
Asterisk ServerIP phone IPTSP
SIP Trunk
SynesisIT
9
1.6 Basic Call Flow (Extended)
10/25/2020
Asterisk Codebase
❑ File :
1. Path : etc/asterisk
2. Path : var/lib/asterisk
Sip.conf exention.conf
Audio Script
10
10/25/2020
1.7 sip Trunk setup in sip.conf
11
SIP TRUNK
10/25/2020
1.7 extention.conf(Code Demo)
[robi_airtel_360]
exten => _X.,1,Answer()
exten => _X.,2,SayDigits(123)
exten => _X.,3,,SayNumber(123)
exten => _X.,n,Hangup
[robi_airtel_360]
exten => _203,1,Hangup
[robi_airtel_360]
exten => _X.,1,Answer()
exten => _X.,n,Background(welcome&silence/3)
exten => _X.,n,Background(testsounds/allServiceAbout)
exten => 1,1,Playback(welcomeToService)
exten => 1,2,AGI(robi-pg/pg_registration.php,GYN,START)
exten => 2,1,Playback(welcomeToServiceSecond)
exten => 2,2,AGI(robi-pg/pg_registration.php,BAN,START)
exten => _[*#03456789],n,Hangup
exten => h,1,AGI(mkvas/mk_cdr_update.php)
Context
Extension , priority ,
application
Dial plan
12
1. IVR
2. Cli
3. Channel Variable
10/25/2020
[airtel_pg-ivr-free-min]
exten => un,1,Set(trycount=0)
exten => un,2,Background(pg_sounds/pg_sub_free_zero)
exten => un,n,Background(silence/3)
exten => un,n,Background(pg_sounds/pg_sub_free_zero)
exten => un,n,Background(silence/3)
exten => un,n,Background(pg_sounds/pg_no_number)
exten => un,n,Background(pg_sounds/pg_sub_free_zero)
exten => un,n,Background(silence/3)
exten => un,n,Background(pg_sounds/pg_max_no_number)
exten => un,n,Hangup
exten => 0,1,Goto(airtel_pg-unsubscribed,un,1)
exten => 1,1,AGI(robi-pg/pg_recharge.php)
;exten => 1,n,Playback(pg_sounds/pg_request_accepted)
exten => 1,n,GotoIf($["${is_recharged}" = "SUCCESS"]?pg-ivr-incoming,in,4)
exten => 1,n,Background(pg_sounds/pg_sub_recharge_failed)
;exten => 2,1,Goto(airtel_pg-gyan-branches-unsub,bn,1)
exten => _[23456789],1,Playback(pg_sounds/pg_wrong_number)
exten => _[23456789],n,Set(trycount=$[${trycount}+1])
exten => _[23456789],n,GotoIf($["${trycount}" != "3"]?airtel_airtel_,un,2)
exten => _[23456789],n,Playback(pg_sounds/pg_max_try)
exten => _[23456789],n,Hangup
exten => h,1,AGI(robi-pg/pg_cdr_update.php)
1.8 More extention.conf(Code Demo)
13
10/25/2020
References
❑ SIP : https://tools.ietf.org/html/rfc3261#section-8.2.2.3
❑ Mobile VoIP: https://www.voip-info.org/wiki/view/Mobile+VoIP
❑ Asterisk: https://www.voip-info.org/wiki/view/Asterisk
❑ PBX: https://www.voip-info.org/wiki/view/PBX
❑ extentions.conf: http://www.asteriskguru.com/tutorials/extensions_conf.html
14
❑ Icons: https://www.flaticon.com
10/25/2020 15
Thank You

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Asterisk presentation v1.1

  • 2. Asterisk Server Internet Softphone/ Hard phone Softphone/ Hard phone 1.1 Basic Call Routing 10/25/2020 2
  • 3. VOIP 10/25/2020 Definition 1. Voice Over Internet Protocol 2. group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol network. Why Use It ? 1. Cost Effective 2. Functionality How does it Work ? 1.2 VoIP Routing 3 IP Phone Ethernet Router Internet Router Ethernet IP Phone Server Server
  • 4. SIP 10/25/2020 Definition 1. Session Initiation Protocol 2. signaling Protocol for IP communication. 1. initiating, maintaining, modifying and terminating real-time sessions. 3. SIP itself does not provide communication services. Sends IP Package IP Communication Laptop(IP device 1) Phone(IP device 2) IP network 4
  • 5. 10/25/2020 IP Communication 1. Location (IP Address) 2. Same Codec (For Encoding-Decoding) SIP Locate other party How to construc t & Send Packets How to Manage Sessions Which Codec 1.3 SIP 5
  • 6. 10/25/2020 SDP Definition 1. Session description Protocol 2. When SIP packets are send and received the both parties need to know the media details(transport address, transport address, transport protocol, codec, ports and other session description metadata). 3. SDP is used to advertise such information. 1.4 A SIP Packet 6
  • 7. 10/25/2020 Asterisk 1. It is a complete PBX(Private Branch exchange) in software. 2. Can Run on Windows, Linux, OS X. 3. How to install Asterisk ? What Can we do with Asterisk ? 1. Call Conferencing 2. intercom calling extension to extension 3. Interactive Voice Response(IVR) 4. Call Queuing etc. 7
  • 8. 10/25/2020 1.5 A Basic Call Flow (Outgoing) User Mobile Phone Asterisk ServerIP phone Cellular Tower 8
  • 9. 10/25/2020 Converts 3G,4G,GSM to SIP Calling Asterisk ServerIP phone IPTSP SIP Trunk SynesisIT 9 1.6 Basic Call Flow (Extended)
  • 10. 10/25/2020 Asterisk Codebase ❑ File : 1. Path : etc/asterisk 2. Path : var/lib/asterisk Sip.conf exention.conf Audio Script 10
  • 11. 10/25/2020 1.7 sip Trunk setup in sip.conf 11 SIP TRUNK
  • 12. 10/25/2020 1.7 extention.conf(Code Demo) [robi_airtel_360] exten => _X.,1,Answer() exten => _X.,2,SayDigits(123) exten => _X.,3,,SayNumber(123) exten => _X.,n,Hangup [robi_airtel_360] exten => _203,1,Hangup [robi_airtel_360] exten => _X.,1,Answer() exten => _X.,n,Background(welcome&silence/3) exten => _X.,n,Background(testsounds/allServiceAbout) exten => 1,1,Playback(welcomeToService) exten => 1,2,AGI(robi-pg/pg_registration.php,GYN,START) exten => 2,1,Playback(welcomeToServiceSecond) exten => 2,2,AGI(robi-pg/pg_registration.php,BAN,START) exten => _[*#03456789],n,Hangup exten => h,1,AGI(mkvas/mk_cdr_update.php) Context Extension , priority , application Dial plan 12 1. IVR 2. Cli 3. Channel Variable
  • 13. 10/25/2020 [airtel_pg-ivr-free-min] exten => un,1,Set(trycount=0) exten => un,2,Background(pg_sounds/pg_sub_free_zero) exten => un,n,Background(silence/3) exten => un,n,Background(pg_sounds/pg_sub_free_zero) exten => un,n,Background(silence/3) exten => un,n,Background(pg_sounds/pg_no_number) exten => un,n,Background(pg_sounds/pg_sub_free_zero) exten => un,n,Background(silence/3) exten => un,n,Background(pg_sounds/pg_max_no_number) exten => un,n,Hangup exten => 0,1,Goto(airtel_pg-unsubscribed,un,1) exten => 1,1,AGI(robi-pg/pg_recharge.php) ;exten => 1,n,Playback(pg_sounds/pg_request_accepted) exten => 1,n,GotoIf($["${is_recharged}" = "SUCCESS"]?pg-ivr-incoming,in,4) exten => 1,n,Background(pg_sounds/pg_sub_recharge_failed) ;exten => 2,1,Goto(airtel_pg-gyan-branches-unsub,bn,1) exten => _[23456789],1,Playback(pg_sounds/pg_wrong_number) exten => _[23456789],n,Set(trycount=$[${trycount}+1]) exten => _[23456789],n,GotoIf($["${trycount}" != "3"]?airtel_airtel_,un,2) exten => _[23456789],n,Playback(pg_sounds/pg_max_try) exten => _[23456789],n,Hangup exten => h,1,AGI(robi-pg/pg_cdr_update.php) 1.8 More extention.conf(Code Demo) 13
  • 14. 10/25/2020 References ❑ SIP : https://tools.ietf.org/html/rfc3261#section-8.2.2.3 ❑ Mobile VoIP: https://www.voip-info.org/wiki/view/Mobile+VoIP ❑ Asterisk: https://www.voip-info.org/wiki/view/Asterisk ❑ PBX: https://www.voip-info.org/wiki/view/PBX ❑ extentions.conf: http://www.asteriskguru.com/tutorials/extensions_conf.html 14 ❑ Icons: https://www.flaticon.com