Grandstream Networks HT701 User Manual page 37

Register Expiration
Registration Retry Wait
Time
Local SIP port
Local RTP port
Use Random SIP Port
Use Random RTP Port
Refer to Use Target
Contact
Transfer on Conference
Hang up
Enable Ring-Transfer
Disable Bellcore Style
3-Way Conference
Remove OBP from
Route Header
Support SIP Instance ID
Validate incoming SIP
message
Check SIP User ID for
FIRMWARE VERSION 1.0.3.1
calls.
This parameter allows the user to specify the time frequency (in minutes) the HT70X
refreshes its registration with the specified registrar. The default interval is 60 minutes
(or 1 hour). The maximum interval is 65535 minutes (about 45 days).
Retry registration if the process failed. Default is 20 seconds.
Defines the local SIP port the HT70X will listen and transmit. The default value for FXS
port is 5060.
Defines the local RTP-RTCP port pair the HT70X will listen and transmit. It is the base
RTP port for channel 0. When configured,
channel 0 uses this port _value for RTP and the port_value+1 for its RTCP
The default value for FXS port is 5004.
Default is No. This parameter forces the random generation of The local SIP ports
when set to Yes. This is usually necessary when multiple HT70X are behind the same
NAT.
Default is No. This parameter forces the random generation of the local RTP ports
when set to Yes. This is usually necessary when multiple HT70X are behind the same
NAT.
Default is No. If set to YES, then for Attended Transfer, the "Refer-To" header uses the
transferred target's Contact header information.
Default is No. In which case if the conference originator hangs up the conference will
be terminated. When option YES is chosen,
each other so that B and C can choose to either continue the conversation or
hang up.
Default is No, this will create a Semi-Attendant Transfer. When set to Yes, device can
transfer the call upon receiving ring back tone or SIP message 180.
Default is No. you can make a Conference by pressing 'Flash' key. If set to Yes, you
need to dial *23 + second callee number.
Default is No. When option YES is chosen, the Out Bound Proxy will be removed from
Route header.
Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP
Instance ID as defined in IETF SIP Outbound draft.
Default is No. If set to yes all incoming SIP messages will be strictly validated
according to RFC rules. If message will not pass validation process, call will be
rejected.
Default is No. Check the incoming SIP User ID in Request URI. If they don't match, the
HT70X USER MANUAL
originator will transfer other parties to
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