Install Asterisk
sudo apt-get install asterisk
Enter your country code when prompted for.
If you have to reconfigure it again removing asterisk without other dependencies will not get rid of config files and reinstall wil not create it so be careful and try uninstalling asterisk-config if required.
Port forwarding will be required from gateway to receive calls from Registrars.
sudo vi /etc/asterisk/users.conf
[1000]
type=friend
host=dynamic
context=home
secret=password
callerid=CIA FBI ATF <1-555-555-5555>
dtmfmode=rfc2833
nat=yes
mailbox=200@home
disallow=all
allow=ulaw
[1000]
type=friend
host=dynamic
context=home
secret=password
callerid=CIA FBI ATF <1-555-555-5555>
dtmfmode=rfc2833
nat=yes
mailbox=200@home
disallow=all
allow=ulaw
sudo vi /etc/asterisk/sip.conf
add under [general] just below follwoing default line
add under [general] just below follwoing default line
;register => 3456@mydomain:5082::@mysipprovider.com
register => sipid:sippassword@sipgate/sipid
;localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
localnet=192.168.0.0/255.255.255.0
;localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
localnet=192.168.0.0/255.255.255.0
; externhost=foo.dyndns.net ; refreshed periodically
; externrefresh=180 ; change the refresh interval
externhost=fqdn.com
externrefresh=180
At the bottom of file add below
[sipgate]
type=peer
secret=sippassword
insecure=invite
username=sipid
defaultuser=sipid
fromuser=sipid
context=sipgate_in
fromdomain=sipgate.co.uk
host=sipgate.co.uk
outboundproxy=sipgate.co.uk
qualify=yes
disallow=all
allow=alaw
dtmfmode=rfc2833
sudo vi /etc/asterisk/extensions.conf
[sipgate_in]
;exten => sipid,1,Dial(SIP/1000) ; <-- instead of extension you should define the corresponding peer
exten => sipid,1,Dial(SIP/77@localhost:5080) ; forward calls to Skype SIP URI
exten => sipid,n,Hangup
[sipgate_out]
exten => _X.,1,Set(CALLERID(num)=sipid)
exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg)
exten => _X.,3,Hangup
[home]
exten => 1000,1,Playback(demo-echotest) ; Let them know what's going on
exten => 1000,2,Echo ; Do the echo test
exten => 1000,3,Playback(demo-echodone) ; Let them know it's over
type=peer
secret=sippassword
insecure=invite
username=sipid
defaultuser=sipid
fromuser=sipid
context=sipgate_in
fromdomain=sipgate.co.uk
host=sipgate.co.uk
outboundproxy=sipgate.co.uk
qualify=yes
disallow=all
allow=alaw
dtmfmode=rfc2833
sudo vi /etc/asterisk/extensions.conf
[sipgate_in]
;exten => sipid,1,Dial(SIP/1000) ; <-- instead of extension you should define the corresponding peer
exten => sipid,1,Dial(SIP/77@localhost:5080) ; forward calls to Skype SIP URI
exten => sipid,n,Hangup
[sipgate_out]
exten => _X.,1,Set(CALLERID(num)=sipid)
exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg)
exten => _X.,3,Hangup
[home]
exten => 1000,1,Playback(demo-echotest) ; Let them know what's going on
exten => 1000,2,Echo ; Do the echo test
exten => 1000,3,Playback(demo-echodone) ; Let them know it's over
Restart Asterisk
sudo /etc/init.d/asterisk restart
Check Asterisk Status
sudo asterisk -r
sipgateway*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sipgate:5060 N 8373647367 105 Registered Sat, 24 Sep 2011 01:04:12
1 SIP registrations.
myPortal*CLI>
Host dnsmgr Username Refresh State Reg.Time
sipgate:5060 N 8373647367 105 Registered Sat, 24 Sep 2011 01:04:12
1 SIP registrations.
myPortal*CLI>
So using above configuration we have set up a Sipgate account successfully registered on Asterisk to forward all incoming calls to Skype URI which will in turn call skype id or numbers.