Asterisk Settings
If you're new here, you may want to subscribe to my RSS feed. Thanks for visiting!
Asterisk is the leading open source VoIP PBX. It is widely deployed by small businesses and homes. You can configure Asterisk to take advantage of Vbuzzer’s high quality, low price VoIP service with rich feature sets.
A. Install Fedora core 6 on a PC
B. Install Asterisk on the Fedora Core
1. Download the latest version of Asterisk from http://www.asterisk.org/downloads
The current version is Asterisk 1.4.18. The current package is asterisk-1.4.18.tar.gz.
2. Put asterisk-1.4.18.tar.gz in a folder of Fedora Core 6, for example /home/sip
3. Unpack this file asterisk-1.4.18.tar.gz
By running “tar –zxvf asterisk-1.4.18.tar.gz”, this package will be unpacked into a new folder named asterisk-1.4.18, for example /home/sip/asterisk-1.4.18
4. Go into the folder asterisk-1.4.18
5. ./configure
6. ./make
7. ./make install
8. ./make samples
Step2. Design the topology architecture
SIP clients register on the Asterisk PC. The Asterisk PC registers on the vBuzzer system. SIP clients can call each other by the Asterisk. Any SIP client can dial out by the Asterisk routes all outgoing calls to the vBuzzer system. Any incoming call from the vBuzzer system can reach a SIP client through the Asterisk. The topology is shown below.
In the picture, we illustrate two vbuzzer users, including vbuzzeruser1 and vbuzzeruser2, and two asterisk users including asteriskuser1 and asteriskuser2. Calling scenarios are illustrated below.
A. An internal call can be placed among asteriskuser1 and asteriskuser2. All internal calls are switched by the Asterisk.
B. An incoming call of a vBuzzer user from vBuzzer system will be routed to a corresponding Asterisk user. We assume that an incoming call of vbuzzeruser1 is forwarded to asteriskuser1, and an incoming call of vbuzzeruser2 is forwarded to asteriskuser2.
C. All outgoing calls of an Asterisk user will be forwarded to the vBuzzer system as placed by a vBuzzer user. We assume that an outgoing call of asteriskuser1 is thought of as placed by vbuzzeruser1, an outgoing call of asteriskuser2 is thought of as placed by vbuzzeruser2.
Step3. Write the configuration files of Asterisk sip.conf and extensions.conf
All configuration files of Asterisk are stored in /etc/asterisk. Please do not change other configuration files except for sip.conf and extensions.conf.
A. Modify sip.conf
1. Go to /etc/asterisk
2. Open sip.conf
3. Remove all contents
4. Type the following contents into sip.conf
;sip.conf
;—————————————–Begin of sip.conf————————————–
[general]
register => vbuzzeruser1:vbuzzerpassword1@vbuzzer_vbuzzeruser1
register => vbuzzeruser2:vbuzzerpassword2@vbuzzer_vbuzzeruser2
allow=all
[vbuzzer_vbuzzeruser1]
; Vbuzzer configuration
type=friend
context=vbuzzer_incoming
host=vbuzzer.com
username= vbuzzeruser1
canreinvite=no
dtmf=rfc2833
dtmfmode=auto
fromdomain=vbuzzer.com
fromuser= vbuzzeruser1
hidecallerid=yes
insecure=very
nat=route
port=80
qualify=2000
secret= vbuzzerpassword1
username= vbuzzeruser1
allow=all
[vbuzzer_vbuzzeruser2]
; Vbuzzer configuration
type=friend
context=vbuzzer_incoming
host=vbuzzer.com
username= vbuzzeruser2
canreinvite=no
dtmf=rfc2833
dtmfmode=auto
fromdomain=vbuzzer.com
fromuser= vbuzzeruser2
hidecallerid=yes
insecure=very
nat=route
port=80
qualify=2000
secret= vbuzzerpassword2
username= vbuzzeruser2
allow=all
[asteriskuser1]
type=friend
host=dynamic
context=phones_asteriskuser1
secret= asteriskpassword1
nat=route
allow=all
[asteriskuser2]
type=friend
host=dynamic
context=phones_ asteriskuser2
secret= asteriskpassword2
nat=route
allow=all
;—————————————–End of sip.conf——————————————
5. Save sip.conf
B. Modify extensions.conf
1. Go to /etc/asterisk
2. Open extensions.conf
3. Remove all contents
4. Type the following contents into extensions.conf
; extensions.conf
;—————————————–Begin of extensions.conf ————————————–
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[internal]
exten => vbuzzeruser1,1,NoOp()
exten => vbuzzeruser1,n,Dial(SIP/asteriskuser1, 30)
exten => vbuzzeruser1,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => vbuzzeruser1,n,Hangup()
exten => vbuzzeruser2,1,NoOp()
exten => vbuzzeruser2,n,Dial(SIP/asteriskuser2, 30)
exten => vbuzzeruser2,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => vbuzzeruser2,n,Hangup()
exten => asteriskuser1,1,NoOp()
exten => asteriskuser1,n,Dial(SIP/${EXTEN}, 30)
exten => asteriskuser1,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => asteriskuser1,n,Hangup()
exten => asteriskuser2,1,NoOp()
exten => asteriskuser2,n,Dial(SIP/${EXTEN}, 30)
exten => asteriskuser2,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => asteriskuser2,n,Hangup()
[remote_ vbuzzeruser1]
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/vbuzzer_vbuzzeruser1/${EXTEN})
exten => _X.,n,Hangup()
[remote_ vbuzzeruser2]
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/vbuzzer_vbuzzeruser2/${EXTEN})
exten => _X.,n,Hangup()
[vbuzzer_incoming]
include => internal
[phones_ asteriskuser1]
include => internal
include => remote_ vbuzzeruser1
[phones_ asteriskuser2]
include => internal
include => remote_ vbuzzeruser2
;—————————————–End of extensions.conf ——————————————
5. Save extensions.conf
Step4. Configure SIP clients
You can find configuration information about some SIP adapter/softwares on Vbuzzer Forum or Vbuzzer Blog. Then, you can configure them to register on the Asterisk.
Popularity: 22% [?]
