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Configuring Your Telephone (cont.)
• The autodial fields are for connecting to the remote proxy without dialtone or ringing.
• When the session-timer feature is enabled, your telephone uses a keepalive
mechanism in order to better detect whether a call has been disconnected.
• If you have a preference, you can influence which audio codec your telephone uses
by specifying a preferred codec.
• The preloaded-route feature adds a SIP “Route:” header in outgoing messages to
request that the outbound proxy be included in all return paths. This is ideal for firewall
traversal for those SIP proxies that support this feature.
• If you would rather download a dial plan from a TFTP server rather than specify it on
the telephone-configuration web page, store the dial plan in a file in the same directory as
the “auto-config” configuration files on your TFTP server. Name the file, "dialplan.conf."
Note that the auto-config path also applies to the dial-plan file and that the dial plan is
downloaded from the TFTP server specified on the network-setup web page.
• Rather than configure your telephone from the web interface or LCD menu, you can
alternatively load a configuration file from a TFTP server (see the Auto-config appendix
for details). The address of the TFTP server is specified on the network-setup page
(this server is also used for downloading a dial plan). The only profile type currently
supported is that described in the Auto-config appendix (Profile-C), so set the auto-
config-profile field to NONE to disable auto-configuration and to Profile-C to enable it.
The auto-config-path field specifies the path, relative to the TFTP server’s root path,
from which your telephone downloads the configuration file (this is also used for
downloading a dial plan). You can also force the telephone to load the configuration file
every time it boots by checking the auto-config-every-boot field; otherwise, it loads the
file whenever you click the commit button on the web page.
• Disabling feature codes causes your telephone to not process and to instead pass
through all “star” feature codes, e.g., *69, *70, and *99, for processing downstream.
• Disabling LINE 1/2 during calls allows for other methods of getting a second dial tone.
• Specify to only use UDP SRV search in order to limit the means of DNS name
resolution.
• If headset support is not enabled, simply plugging a headset into your telephone will
not work.
Make any changes you need on this page and then click the button to commit your
changes. Your changes only take effect when you commit your changes.
There is also a button that you can click to restore the factory defaults, which are 5060
for the default SIP port and nothing (blank) or not-selected for the remaining fields.
