Grandstream Networks HT503 User Manual page 32

Local SIP port
Local RTP port
Use Random Port
Refer to Use Target
Contact
Transfer on conference
hangup
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
SIP T1 Timeout
SIP T2 Interval
DTMF Payload Type
Preferred DTMF method
(in listed order)
Disable DTMF
Negotiation
Send Flash Event
Enable Call Features
Offhook Auto-Dial
Proxy-Require
Use NAT IP
FIRMWARE VERSION 1.0.6.8
days).
This parameter defines the local SIP port the HT503 will listen and transmit. The default
value for FXS port is 5060.
This parameter defines the local RTP-RTCP port pair used by the HandyTone ATA. It
is the base RTP port for channel 0.
When configured, the FXS port will use this port _value for RTP and the port_value+1
for its RTCP.
The default value for FXS port is 5004.
Default is No. If set to Yes, the device will pick randomly-generated SIP and RTP ports.
This is usually necessary when multiple HandyTone ATAs 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 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.
Default is No, this will create a Semi-Attendant Transfer. When set to Yes, device can
transfer the call upon receiving ring back tone.
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. If set to Yes, the Outbound Proxy will be removed from the 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.
This parameter sets the payload type for DTMF using RFC2833
The HT503 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
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 to be
automatically dialed when offhook. Please note that only the user part of a SIP address
needs to be entered here. The HT503 will automatically append the "@" and the host
portion of the corresponding SIP address.
Note: User will need this IP address when accessing the IVR via the web configuration
page.
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
HT503 USER MANUAL
Page 32 of 48