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Avaya G860 Media Gateway Setup
d. Verify that the SIP trunks are up between the Avaya Communication Manager and the
G860 Media Gateway and that the transport protocols and ports match (TLS and
5061).
Verifying call routing
Verifying call routing means making calls within a session Internet Protocol (SIP) environment
and between SIP and the public switched telephone network (PSTN).
In general, if a SIP device in the network does not receive an expected response to a SIP
message it has transmitted, the standard procedure is to retransmit that message t an
exponentially decreasing interval (0.5 s, then 1 s, etc.) up to 6 more times. Look for this
behavior at various points in the call routing path if calls are not successful or if calls only remain
established for about 32 s. Most likely, the expected message was not routed properly because
of NAT, dial-peer, DNS, or firewall configuration errors.
1. Make calls from a telephone on the PSTN to a SIP telephone. If inbound calls fail, verify
that the proper Tel to IP routing rules were defined to support routing SIP calls to the
appropriate C-LANS.
2. Make calls between SIP telephones registered within each proxy to verify successful call
completion.
If outbound calls fail, verify that the proper IP to Tel routing rules were defined to support
routing SIP calls to the appropriate DS3/DS1.
3. Log into a SAT session on the media server.
4. Type list trace station to verify that the call is being properly routed through the
SIP trunk to the G860 Media Gateway.
5. Change the transport protocol to TCP on the SIP trunk and use a SIP-capable protocol
analyzer to monitor the signaling messages.
6. Type change signaling group and press Enter.
7. Change the following fields:
Transport Method: tcp
●
Far-end Listen Port: 5060
●
8. Log into the G860 Media Gateway.
9. Lock the TPM
10. Double-click on the locked TPM shown in black.
11. Select the SIP tab.
12. Select the Protocol Settings tab
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Avaya G860 Media Gateway Setup
03-601863, Issue 1, November 2007
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