Server Redundancy - Yealink SIP-T2 SERIES Administrator's Manual

Administrator's Guide for SIP-T2 Series/T19(P) E2/T4 Series/CP860 IP Phones
If you place a call via web user interface but the trusted IP address has not been configured, the
web user interface prompts "Call fail".
To place a call via web user interface:
Click on Directory->Phone Call Info.
1.
Select the desired account from the pull-down list of Outgoing Identity.
2.
Enter the callee's number in the Dial Number field.
3.
Click Dial to dial out the number.
4.
The web user interface prompts "Call Success" and the phone will automatically dial out
the number. You can click Hang Up to end the call.
If it is the first time you place a call via web user interface, the LCD screen will prompt the
message "Allow remote control?". Press OK on the phone to allow remote control and then
the phone will automatically dial out the number.
Note
You can also place an IP direct call via web user interface. The IP phone supports either IPv4 or
IPv6 address.

Server Redundancy

Server redundancy is often required in VoIP deployments to ensure continuity of phone service,
for events where the server needs to be taken offline for maintenance, the server fails, or the
connection between the IP phone and the server fails.
Two types of redundancy are possible. In some cases, a combination of the two may be
deployed:
Failover: In this mode, the full phone system functionality is preserved by having a second

equivalent capability call server take over from the one that has gone down/off-line. This
mode of operation should be done using the DNS mechanism from the primary to the
secondary server.
Fallback: In this mode, a second less featured call server with SIP capability takes over call

control to provide basic calling capability, but without some advanced features (for
example, shared line, call recording and MWI) offered by the working server. IP phones
support configuration of two servers per SIP registration for fallback purpose.
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