GXE502X User Manual 7/9/2008
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Version 03
o Session Keep Alive: This field allows users to enable or disable usage of the session
timer. This is enabled by default to: Automatic/Session Timer.
o Session Expiration: The session timer enables SIP sessions to be periodically
“refreshed” via a SIP request. If enabled, before the session interval expires, the
GXE502X will send a SIP re-INVITE message to the SIP server, if this message is
not replied the session will be terminated. This will avoid stalled sessions occupying
the channel/bandwidth. Session Expiration is the timer (in seconds) at which the
session is considered timed out if no successful session refresh transaction occurs
beforehand. The default value is 180 seconds. Incorrect configurations could cause
calls to get dropped.
o Min-SE: The minimum session expiration (in seconds). The default value is 90
seconds.
o Use DNS SRV: Default is No. Select “Yes” if the domain name of the SIP server or
Outbound Proxy is compliant with RFC2782. The SIP trunk will not function properly
if this field is not configured correctly.
o Unregister on Reboot: Default is No. If set to “Yes”, The GXE502X will first send a
registration request to remove all previous or multiple bindings by adding “*” in the
SIP Contact Header. Please use this feature ONLY if the proxy supports removal
binding requests. Otherwise the SIP trunk will not function properly.
o Register Active: This parameter controls whether the GXE502X needs to send
REGISTER messages to the Proxy Server before making or receiving calls. The
default setting is “Yes” (for most dynamic SIP trunks).
Note:
For security reasons: If this is set to “No” (for most static SIP trunks) any incoming
request MUST match the SIP server IP or the FQDN resolution must match the
sending IP address.
o CBCOM Encryption: If set to Yes, RTP will be encrypted as per the algorithms of
CBCOM. The Default setting is No.
o DID Switch: This field is used to route incoming calls when a DID (Direct Inward
Dialing #) is used instead of the account ID. It is very helpful when there are several
DIDs related to the same (static) trunk provided by your SIP Trunk Service Provider.
Note:
All SIP requests are verified by the GXE502X so the User Part of the SIP URI has to
match the account ID or the DIDs. For outgoing calls when DID switch is enabled,
the DID number in the list will be used in the From, Contact and PAI (P-Asserted-
Identity) value.
If a prepend prefix is configured, it will also be used as prefix for the DIDs.
For example, if the DID is +16175669300 and for outgoing calls the service provider
requires “+”, then configure the prepend prefix to “+” and add the DID 16175669300.
Then +16175669300 will be used to match both incoming calls and also outgoing
calls.