Grandstream Networks HT502 User Manual page 37

Contact
Transfer on Conference
Hang up
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
SIP T1 Timeout
SIP T2 Interval
DTMF Payload Type
Preferred DTMF method
Disable DTMF
Negotiation
DTMF via RFC2833
DTMF via SIP INFO
Send Flash Event
Enable Call Features
Offhook Auto-Dial
Offhook Auto-Dial
Delay
Proxy-Require
Use NAT IP
FIRMWARE VERSION 1.0.8.4
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 either to continue the conversation or
hang up.
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.
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.
Maximum retransmission interval for non-INVITE requests and INVITE responses.
Sets the payload type for DTMF using RFC2833.
The HT502 supports up to 3 different DTMF methods including in-audio, via RTP
(RFC2833) and via Sip Info. The user can configure DTMF method in a priority list.
Default is No. If set to yes, use above DTMF order without negotiation
Send DTMF via RTP (According to RFC 2833).
Send DTMF via SIP INFO message.
Default is No. If set to yes, flash will be sent as DTMF event.
Default is Yes. (If Yes, call features using star codes will be supported locally)
This parameter allows users to configure a User ID or extension number that is
automatically dialed when off-hook. Only the user part of a SIP address needs is
entered here. The HT502 will automatically append the "@" and the host portion of the
corresponding SIP address.
Configure the delay time for offhook auto-dial function. Range is
default is 0.
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
NAT IP address used in SIP/SDP message. Default is blank.
HT502 USER MANUAL
originator will transfer other parties to
0-60 seconds,
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