Owner's Manual
EXP1240 Admin Guide, Version 2.0
Proprietary and Confidential
Chapter: EXP1240 Administration Interface
35
Outbound
Proxy
Empty
This is a Session Border Controller DNS or IP address (OR SIP server
outbound proxy address)
Set the Outbound proxy to the address and port of private NAT gateway so
that SIP messages sent via the NAT gateway.
Permitted value(s): AAA.BBB.CCC.DDD or <URL> or <URL>:<Port-Number>
Examples: “192.168.0.1”, “192.168.0.1:5062”,
“nat.company.com” and “sip:nat@company.com:5065”.
Conference
Server
Empty
Broadsoft conference feature.
Set the IP address of the conference server.
In case an IP is specified pressing handset conference will establish a
connection to the conference server.
If the field is empty the original 3-party local conference on 8630 is used.
Re
-
registration
time
600
The “expires”
value
35
nalyse
35
n
in SIP REGISTER requests. This value
indicates how long the current SIP registration is valid, and hence is
specifies the maximum time between SIP registrations for the given SIP
account.
Permitted value(s): A value below 60 sec is not recommended, Maximum
value 65636
SIP Session
Timers:
Disabled
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 value is 1800 s according to the
RFC. Min: 90 s. Max: 65636.
If disabled session timers will not be used.
Session Timer
Values (s):
1800
Default value is 1800s according to the RFC.
If disabled session timers will not be used.
Permitted value(s): Minimum value 90, Maximum 65636
SIP Transport
UDP
Select UDP, TCP, TLS 1.0
Signal TCP
Source Port
Disabled
When SIP Transport is set to TCP or TLS, 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 hence the local SIP port parameter,
specified within the SIP/RTP Settings (see 5.5.5) will not be used. The
“Signal TCP Source Port” parameter specifies if the used source port shall
be signaled explicitly in the SIP messages.
Secure RTP
Disabled
With enable RTP will be encrypted (AES
-
128) using the key negotiated via
the SDP protocol at call setup.
Secure RTP
Auth
Disabled
With enable secure
RTP is using authentication of the RTP packages.
Note: with enabled SRTP authentication maximum 4 concurrent calls is
possible per base in a single or multicell system.
RTP from own
base station:
Disabled
If disabled RTP stream will be send from the base
, where the handset is
located. By enable the RTP stream will always be send from the base,
where the SIP registration is made.
Keep Alive
Enabled
This directive defines the window period (30 sec.) to keep opening the
port of relevant NAT-aware router(s), etc.
Show
Extension on
Handset Idle
Screen
Enabled
If enabled extension will be shown on handset idle screen.
Hold
Behaviour
RFC 3264
Specify the hold behaviour by handset hold feature.
RFC 3264: Hold is 35nalyse35n according to RFC 3264, i.e. the connection
information part of the SDP contains the IP Address of the endpoint, and










