Firmware Version 1.0.9.1 - Grandstream Networks HT503 User Manual

Local RTP Port
Use Random Port
Refer to Use Target
Contact
Remove OBP from Route
Header
Support SIP instance ID
Validate incoming
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
Proxy Require
Use NAT IP
Use SIP User-Agent
Header
Ring Timeout
Early Dial

FIRMWARE VERSION 1.0.9.1

This parameter defines the local RTP port pair used by the HandyTone ATA. The
default value for FXO port is 5012.
This parameter forces the random generation of both the local SIP and RTP ports when
set to Yes. This is usually necessary when multiple HT503 units 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. 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 reliable usage.
Maximum retransmission interval for non-INVITE requests and INVITE responses.
Sends DTMF using RFC2833
The HT503 supports up to 3 different DTMF methods including in-audio, via RTP
(RFC2833) and via Sip Info. User can configure DTMF method in a priority list.
Default is No. If set to yes, use above DTMF order without negotiation
SIP Extension to notify SIP server that the unit is behind a NAT/Firewall.
NAT IP address used in SIP/SDP message. Default is blank.
Used to replace SIP User-Agent Header (No Default)
Sets the time in which an incoming from PSTN call will stop ringing when not picked up.
Default is No. Use only if proxy supports 484 response. This parameter controls
whether the phone will send an early INVITE each time a key is pressed when a user
dials a number. If set to "Yes", an INVITE is sent using the dial-number collected thus
far. Otherwise, no INVITE is sent until the "(Re-)Dial" button is pressed or after about 5
seconds have elapsed. The "Yes" option should be used ONLY if there is a SIP proxy
configured and the proxy server supports 484 Incomplete Address response.
Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error).
Note: This feature is NOT designed to work with and should NOT be enabled for direct
HT503 USER MANUAL
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