Hello
Our company using hardware SIP phones with FXO ports.
I have configured Asterisk - create user account for this phone, but I can’t understand, how configure dialplan for transfer callee PSTN number to this phone?
As I understand - Asterisk have to modify filed “To” in SIP Invite message to callee PSTN number, but “Requst-URI” must stay as URI hardware SIP phone.
My current configuration:
sip.conf
[code][general]
minexpiry=10
bindport=5060
context=router
allowoverlap=no
udpbindaddr=192.168.10.233
tcpenable=no
language=en
dtmfmode=rfc2833
[555]
type=friend
username=555
callerid=User555 <555>
context=router
secret=Emzior555
qualify=no ; linphone will become unreachable if qualify=yes
host=dynamic
canreinvite=no ; no
nat=no
port=50600
[777]
type=friend
username=777
callerid=User777 <777>
context=router
secret=Emzior777
qualify=no ; linphone will become unreachable if qualify=yes
host=dynamic
canreinvite=no ; yes
insecure=invite,port
nat=nonat[/code]
extensions.conf
[code][router]
include => outbound
exten => 777,1,Dial(SIP/777)
[outbound]
exten => _6.,1,Dial(SIP/555)[/code]
In this configuration I’m trying call PSTN callee from 777 user through 555 (hardware SIP phone)