For Asterisk setups the following needs to be done.
For this example we will be using Elastix.
Finding Asterisk Verison
Use Putty or use the Asteris-CLI
asterisk -r
core show version
Or
asterisk -r
show version
Modify extensions_custom.conf
Asterisk versions 1.6 or older
[from-outside-redir]
;Dialed Number routing according to SIP To: header
exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(TO),@,1),:,2):1},1)
Asterisk versions 1.8 or newer
[from-trunk]
;Dialed Number routing according to SIP To: header
exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(TO),@,1),:,2):1},1)
Add SIP Trunk
Asterisk versions 1.6 or older
type=peer
insecure=invite,port
dtmfmode=auto
host=sbc.ringlogix.com
context=from-outside-redir
username=ACCOUNT-ID
trustrpid=yes
sendrpid=yes
secret=SIP-PASSWORD
allow=all
canreinvite=no
Asterisk versions 1.8 or newer
type=peer
insecure=invite,port
dtmfmode=auto
host=sbc.ringlogix.com
context=from-trunk
username=ACCOUNT-ID
trustrpid=yes
sendrpid=yes
secret=SIP-PASSWORD
allow=all
canreinvite=no
Asterisk versions 1.6 or older
type=user
insecure=invite,port
dtmfmode=auto
host=sbc.ringlogix.com
context=from-outside-redir
allow=all
Asterisk versions 1.8 or newer
type=user
insecure=invite,port
dtmfmode=auto
host=sbc.ringlogix.com
context=from-trunk
allow=all
ACCCOUNT-ID:SIP-PASSWORD@sbc.ringlogix.com:5060