Here I’m using Grnadstream FXO Gateway GXW4104
Login to GXW4104
Select Profile 1
SIP Server: IP address of Asterisk box
SIP Registration: No
Select FXO Lines
4. Unconditional Call Forward to VOIP: ch1-4:604;
Here 604 is the extension which the incoming call to be routed in asterisk.
In Asterisk , edit the following files.
users.conf
[gxw4104]
trunkname = Grandstream
trunkstyle = voip
type = peer
context = DLPN_incoming
host = IP address of Grnadstream Gateway
hasexten = no
hassip = yes
insecure = port
canreinvite = no
allow = all
extensions.conf
[CallingRule_Call_out]
exten = _X.,1,Macro(trunkdial-failover-0.3,SIP/gxw4104/${EXTEN})
[DLPN_ePillars]
include = CallingRule_Call_out
[DLPN_incoming]
exten = 604,1,Dial(SIP/604)
It will be available in Asterisk-GUI after reloading asterisk.
Tech.. Logs.. 