In the Router, you must configure the same SIP and RTP Ports as configured in the SETU VGRX. In other
words, you must configure Port Forwarding for SIP and RTP on the Router.
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Under Handling of Incoming Calls on the SIP Trunk, set the Incoming Call Routing option as Route
all incoming calls (with CLI) - to the Called Party Number.
•
For SIP Trunk 1, select the Destination Port for Routing Calls as Fixed, and create Routing Group
as Mobile Port.
•
For Mobile Port, select the Destination Port for Routing Calls as Fixed, and create Routing Group
as SIP Trunk 1 only.
For instructions on configuring SIP Trunk parameters, see
•
Now, configure the Peer-to-Peer Table.
In this example, you would have to configure the Peer-to-Peer table as follows:
•
At Location B, in Destination Number, enter the Number you want to dial to call the phone at Location
A. In this case, 2001.
•
For the number you entered in the Destination Address, enter the IP Address of the Router
connected at Location A. In this case, 115.118.161.165
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The Peer-to-Peer table you configure for SETU VGRX at Location B would look like this:
•
Configure PBX at location A such that calls received on the FXO Port of the PBX are routed to the FXS
Port in sequential order, that is, calls to 2001 are routed to FXS 1 and so on. Similarly, when any FXS Port
user dials a number starting with '9898', it should be routed using the FXO Port of the PBX to the FXS Port
of the SETU VFXTH.
•
When user 2001 of location A calls 9898012345, the call is routed using the FXO Port of the PBX to FXS
Port of the SETU VFXTH. Further, it will be routed using the SIP Trunk of the SETU VFXTH to the IP
address 121.124.130.110, as the system finds a matching entry for the dialed number in the Peer-to-Peer
table.
Matrix SETU VGRX System Manual
"SIP Trunk"
under Basic Settings.
257
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