Grandstream Networks HT701 User Manual page 25

SIP Registration
Unregister on Reboot
Outgoing Call without
Registration
Register Expiration
Registration Retry Wait
Time
Local SIP port
Local RTP port
Use Random 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
incoming INVITE
Allow Incoming SIP
Messages from SIP
Proxy Only
SIP T1 Timeout
SIP T2 Interval
DTMF Payload Type
Grandstream Networks, Inc.
VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299
request to indicate the E.164 number. If server supports TEL URI format, then this
option needs to be selected.
Controls whether the HT701 needs to send REGISTER messages to the proxy server.
The default setting is Yes.
Default is No. If set to Yes, the SIP user‟s registration information will be cleared on
reboot.
Default is No. If set to "Yes," user can place outgoing calls even when not registered (if
allowed by Internet Telephone Service Provider) but is unable to receive incoming
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 both the local SIP and
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
call will be rejected. If this option is enabled, the device will not be able to make direct
IP calls.
Default is No. Check the incoming SIP messages. If they don‟t come from the SIP
proxy, they will be rejected. If this option is enabled, the device will not be able to make
direct IP calls.
T1 is an estimate of the round-trip time between the client and server transactions.
If the network latency is high, select larger value for more reliable usage. Default is 0.5
Sec.
Maximum retransmission interval for non-INVITE requests and INVITE responses.
Default is 4 Sec.
Sets the payload type for DTMF using RFC2833. Default is 101.
HT-70X User Manual
Firmware Version 1.0.0.18
originator will transfer other parties to
Page 25 of 35
Last Updated: 03/2012