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:
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.
Create SIP account
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:
; 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.
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
; 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).
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
; 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.
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)
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)
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.
- 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
- CallMake a call with sip phone1 (number 100) and dial number 101
Make a call with sip phone2 (number 101) and dial number 100
- Don't forget to open SIP UDP 5060 ports in both computer firewalls.
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