48
v1.02
Admin and Installation Guide M900 Description of Settings
Setting Description
Outbound proxy Session border controller DNS or IP address. Alternatively SIP server
outbound proxy address. Set the outbound proxy to the address and port
of the private NAT gateway so that SIP messages are sent via this gateway.
Format: addr:port
Default value: Blank
Conference server For Broadsoft conference server only: If the IP address of the server is set
in this text field, pressing the handset's conference function key during a call
will establish a connection to the conference server. If the field is empty, the
system's 3-party local conference is used.
Call log server Broadsoft call log feature. If the IP address of the server is set in this
text field, pressing the handset's call log/call history key during a call will
establish a connection to the call log server.
Reregistration time (s) The “expires” value signaled in SIP REGISTER requests. This value indicates
how long the current SIP registration is valid, and hence it specifies the
maximum time between SIP registrations for the SIP account.
Permitted value(s): Maximum 65636 seconds. A value below 60 seconds is
not recommended.
SIP Session Timers RFC 4028. A “keep-alive” mechanism for calls. The session timer value
specifies the maximum time between “keep-alive” or more correctly session
refresh signals. If no session refresh is received when the timer expires, the
call will be terminated.
Default: Enabled.
Session timer value (s) Default value: 1800 seconds.
Permitted values: Minimum 90 sec, maximum 65636 sec.
If the setting is disabled, session timers will not be used.
Use SIP as XSI authentication TBA
SIP Transport Default: UDP. Other options: TCP, TLS 1.0
Signal TCP Source Port When SIP Transport is set to TCP or TLS, respectively, a TCP or TLS connection
will be established for each SIP extension. The source port of the connection
will be chosen by the TCP stack, and the local SIP port parameter specified
in the SIP/RTP settings will not be used. The “Signal TCP Source Port”
parameter specifies whether the used source port shall be signaled explicitly
in the SIP messages.
Default: Enabled.
Use one TCP connection per
SIP extension
When using TCP or TLS as SIP transport, choose whether the base station is
to establish one TCL or TLS connection for each SIP extension or one TCL or
TCS connection that all SIP extensions will use. Note: If TLS is used and the
SIP server requires client authentication (i.e., requests a client certificate), this
setting must be disabled.
Settings: Disabled (one TCL or TLS connection for all SIP extensions);
Enabled (one TCL or TLS connection for each SIP extension)
Default: Disabled
RTP from own base station If disabled, the RTP stream will be sent from the base station where the
handset is located. I enabled, the RTP stream will always be sent from the
base station where the handset is registered.
Default: Enabled
Keepalive Default: Enabled. When enabled, a new keepalive message will be sent out to
the Registrar/Proxy port after 30 seconds in order to have the port stay open
and the phone remain reachable.
Show extension on handset
idle screen
Default: Enabled. When enabled, the extension number is displayed on the
idle screen of the handset.
To Next Page
Loading...
