Connecting Two Sip Peers To Two Pstn Interfaces Exclusively; Connecting Two Sip Peers To The Same Pstn Interface - Dialogic 4000 Series SU4.1 Reference Manual

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Dialogic 4000 Media Gateway Series SU4.1 Reference Guide

Connecting Two SIP Peers to Two PSTN Interfaces Exclusively

If you choose to connect two SIP peers to two PSTN interfaces, so that each SIP peer can
use one interface exclusively, then carry out the following configuration steps. The
procedure is similar if you need to configure more PSTN interfaces, e.g., three PSTN
interfaces to three SIP peers.
1. Under PSTN Interfaces, enable and configure the two PSTN interfaces. Confirm
with OK.
2. Under SIP Peers, create both SIP peers and make sure the entry in Domain
exactly matches the domain used by the SIP peer in its SIP address for outgoing
calls. Do not enable the option Default peer for received SIP calls for any of
these peers. Confirm with OK.
3. Under Routing, create Route 1 and do the following:
Enable the first PSTN interface as a source peer.
Enable the first SIP peer configured in Step 2 as a Master destination.
Confirm with OK.
4. Under Routing, create Route 2. Then, repeat Step 3 for the second PSTN interface
and the second SIP peer.
5. Under Routing, create Route 3 and do the following:
Enable the first SIP peer configured in Step 2 as a source peer.
Enable the first PSTN interface as a Master destination.
Confirm with OK.
6. Under Routing, create Route 4. Then, repeat Step 5 for the second PSTN interface
and the second SIP peer.
7. Save the configuration in the main configuration interface.

Connecting Two SIP Peers to the Same PSTN Interface

If you want to connect two SIP peers to the same PSTN interface so that all calls from the
PSTN are sent to the first SIP peer if the numbers begin with "1" and to the second peer if
the numbers begin with "2", configure the parameters as follows:
1. Under PSTN Interfaces, enable and configure the PSTN interface. Confirm with OK.
2. Under SIP Peers, create both SIP peers and make sure the entry in Domain
matches exactly the domain used by the SIP peer in its SIP address for outgoing
calls. Do not enable the option Default peer for received SIP calls for any of
these peers. Confirm with OK.
3. Under Routing, create Route 1 and do the following:
Enable the first PSTN interface as a source peer.
Enable the first SIP peer configured in Step 2 as a Master destination.
Under Conditions, click Add and set the Called address to "1.*".
Confirm with OK.
4. Under Routing, create Route 2 and repeat Step 3 for the second SIP peer with the
only difference that the called address condition for this route is "2.*".
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