User's Manual Part 1

Firmware Version 1.0.0.1 UCM6200 Series IP PBX User Manual Page 120 of 320
characters, letters, digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
User Password
Configure the password for user portal access. A random numeric
password is automatically generated. It is recommended to use the
randomly generated password for security purpose.
Language
Select the voice prompt language to be used for this extension. The
default setting is "Default" which is the selected voice prompt language
under web GUI->PBX->Internal Options->Language. The dropdown list
shows all the current available voice prompt languages on the UCM6200.
To add more languages in the list, please download voice prompt
package by selecting "Check Prompt List" under web UI->PBX->Internal
Options->Language.
Concurrent Registrations
The maximum endpoints which can be registered into this extension. For
security concerns, the default value is 1.
Table 34: SIP Extension Configuration Parameters->Media
SIP Settings
NAT
Use NAT when the UCM6200 is on a public IP communicating with
devices hidden behind NAT (e.g., broadband router). If there is one-way
audio issue, usually it's related to NAT configuration or Firewall's support
of SIP and RTP ports. The default setting is enabled.
Can Direct Media
By default, the UCM6200 will route the media steams from SIP endpoints
through itself. If enabled, the PBX will attempt to negotiate with the
endpoints to route the media stream directly. It is not always possible for
the UCM6200 to negotiate endpoint-to-endpoint media routing. The
default setting is "No".
DTMF Mode
Select DTMF mode for the user to send DTMF. The default setting is
"RFC2833". If "Info" is selected, SIP INFO message will be used. If
"Inband" is selected, 64-kbit PCMU and PCMA are required. When "Auto"
is selected, RFC2833 will be used if offered, otherwise "Inband" will be
used.
TEL URI
If the phone has an assigned PSTN telephone number, this field should
be set to “User=Phone”. “User=Phone” parameter will be attached to the
Request-Line and “TO” header in the SIP request to indicate the E.164
number. If set to “Enable”, “Tel” will be used instead of “SIP” in the SIP
request.
Enable Keep-alive
If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is "Yes".