User's Manual

FIRMWARE VERSION 1.0.7.6 HT502 USER MANUAL Page 37 of 48
3-Way Conference
need to dial *23 + second callee number.
Remove OBP from
Route Header
Default is No. When option YES is chosen, the Out Bound Proxy will be removed from
Route header.
Support SIP Instance ID
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.
Validate incoming SIP
message
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.
Check SIP User ID for
incoming INVITE
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.
SIP T1 Timeout
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.
SIP T2 Interval
Maximum retransmission interval for non-INVITE requests and INVITE responses.
DTMF Payload Type
Sets the payload type for DTMF using RFC2833.
Preferred DTMF method
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.
Disable DTMF
Negotiation
Default is No. If set to yes, use above DTMF order without negotiation
DTMF via RFC2833
Send DTMF via RTP (According to RFC 2833).
DTMF via SIP INFO
Send DTMF via SIP INFO message.
Send Flash Event
Default is No. If set to yes, flash will be sent as DTMF event.
Enable Call Features Default is Yes. (If Yes, call features using star codes will be supported locally)
Offhook Auto-Dial
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.
Offhook Auto-Dial
Delay
Configure the delay time for offhook auto-dial function. Range is 0-60 seconds,
default is 0.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
Use SIP User-Agent
Header
Used to replace SIP User-Agent Header (No Default)
Distinctive Ring Tone
Custom Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is
configured, then the device will ONLY uses this ring tone when the incoming call is from
the Caller ID. System Ring Tone is used for all other calls. When selected but no Caller
ID is configured, the selected ring tone will be used for all incoming calls. Distinctive
ring tones can be configured not only for matching a whole number, but also for