User's Manual

Grandstream Networks, Inc. HT701 User Manual Page 26 of 34
Firmware Version 1.0.4.2 Last Updated: 06/2011
Contact
transferred target’s Contact header information.
Transfer on Conference
Hang up
Default is No. In which case if the conference originator hangs up the conference will be
terminated. When option YES is chosen, originator will transfer other parties to
each other so that B and C can choose either to continue the conversation or
hang up.
Enable Ring-Transfer Default is No, this will create a Semi-Attendant Transfer. When set to Yes, device can
transfer the call upon receiving ring back tone.
Disable Bellcore Style
3-Way Conference
Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you
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 HT701 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 HT701 will automatically append the “@” and the host portion of the
corresponding SIP address.
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.
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 matching prefixes. In this case symbol * (star) will be used.
For example:
if configured as *617, Ring Tone 1 will be used in case of call arrived from the area
code 617. Any other incoming call will ring using cadence defined in parameter System
Ring Cadence located under Advanced Settings Configuration page.
Note: If server supports Alert-Info header and standard ring tone set (Bellcore) or
distinctive ring tone 1-10 is specified, then the ring tone in the Alert-Info header from
server will be used. Bellcore rings and tones are independent from custom ring tones.
The custom ring tones can also be specified by alert-info header, for example
Alert-Info: <http://127.0.0.1>;info=ring5