Asterisk Voip

Make a basic call



sip.conf

Rename the sip.conf file (Location: /etc/asterisk/sip.conf) to sip_old.conf for example. Create a new sip.conf empty file and insert the following lines into the file:
[general] 
context = incoming        ; Default context for incoming calls 
allowguest = no        ; Allow or reject guest calls (default is yes, this can also be set to 'osp' 

bindport = 5060          ; UDP Port to bind to (SIP standard port is 5060) 
bindaddr = 0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all) 
srvlookup = yes          ; Enable DNS SRV lookups on outbound calls 
disallow = all        ; First disallow all codecs 
allow = ulaw        ; Allow codecs in order of preference 
allow = alaw 
allow = gsm 
dtmfmode = rfc2833        ; Set default dtmfmode for sending DTMF. 
canreinvite=no 
nat=yes 

////////////////////////////////
Create SIP account
[100] 
type=friend 
context=incoming 
defaultuser=100 
secret=100
host=dynamic 
callerid="Phone1" 

[101] 
type=friend 
context=incoming
defaultuser=101 
secret=101
host=dynamic 
callerid="Phone2"

extensions.conf

The second file to configure is the extensions.conf file (Location: /etc/asterisk/extensions.conf). Rename it to extensions_old.conf and create a new extensions.conf empty file. Insert the following lines into the file:
[general] 
; 
; If static is set to no, or omitted, then the pbx_config will rewrite 
; this file when extensions are modified.  Remember that all comments 
; made in the file will be lost when that happens. 
static=yes 
; 
; if static=yes and writeprotect=no, you can save dialplan by 
; CLI command 'save dialplan' too 
; 
writeprotect=yes 
; 
; If autofallthrough is set, then if an extension runs out of 
; things to do, it will terminate the call with BUSY, CONGESTION 
; or HANGUP depending on Asterisk's best guess (strongly recommended). 
; 
autofallthrough=yes 
; 
; If clearglobalvars is set, global variables will be cleared 
; and reparsed on an extensions reload, or Asterisk reload. 
; 
clearglobalvars=no 
; 
; If priorityjumping is set to 'yes', then applications that support 
; 'jumping' to a different priority based on the result of their operations 
; will do so (this is backwards compatible behavior with pre-1.2 releases 
; of Asterisk). Individual applications can also be requested to do this 
; by passing a 'j' option in their arguments. 
; 
priorityjumping=yes 
; 
;[globals] 
; 
[incoming]
exten => 100,1,Dial(SIP/100) 
exten => 100,2,Hangup() 
exten => 101,1,Dial(SIP/101) 
exten => 101,2,Hangup()
 

Reload the asterisk configuration:

Reload the asterisk configuration by running the following command in the CLI console: (Run "asterisk -r" to connect to the asterisk console)
  • core reload

Download Softphone

X-Lite (First computer - Windows)

Now, the basic installation is done. Download X-Lite (free sip phone) and install it. Download it by following this link: http://asteriskvoipsystem.org/downloads/X-Lite.exe

X-Lite (First computer -Linux)

http://asteriskvoipsystem.org/downloads/X-Lite_Install.tar.gz   Configuration xlite: configuration xlite
xlite phone
Display name : frankc (or whatever you want) User name  : 100 password : 100 Author ....: 100 Domain: IP address of the asterisk server. Do the same with seconds xlite phone. After registration, the sip phone is now ready to make calls.
xlite phone
Verification
  1. RegistrationsIn a CLI console, run the following command: [root@]# sip show peers
    Name/username           Host                       Dyn Forcerport ACL Port     Status
    frankc/100              192.168.1.x                D   N              38116     xyz ms
    101/101                 192.168.1.y                D   N               5061     xyz ms
  2. CallMake a call with sip phone1 (number 100) and dial number 101 Make a call with sip phone2 (number 101) and dial number 100
  3. Don't forget to open SIP UDP 5060 ports in both computer firewalls.
  http://asteriskvoipsystem.org/make-a-basic-call/


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