Grandstream HT503 User Manual - Manual (Page 52)

FIRMWARE VERSION 1.0.14.1 HT503 USER MANUAL Page 50 of 64
URI format, then this option needs to be selected.
SIP Registration Controls whether the HT503 needs to send REGISTER messages to the proxy server.
The default setting is Yes.
Unregister on Reboot Default is No . If set to Yes, the SIP user’s registration information will be cleared on
reboot.
Outgoing Call Without
Registration
Default is No . If set to “Yes,” user can place outgoing calls even when not registered (if
allowed by ITSP) but is unable to receive incoming calls.
Register Expiration This parameter allows the user to specify the time frequency (in minutes) the HT503
refreshes its registration with the specified registrar. The default interval is 60 minutes
(or 1 hour). The maximum interval is 65535 minutes (about 45 days).
SIP registration failure
retry wait time
This parameters allows the user to specify the time frame (in seconds) the HT503 will
wait before sending another SIP registration INVITE in case the first INVITE fails.
Local SIP Port Defines the local SIP port the HT503 will listen and transmit. The default value for FXS
port is 5062.
Local RTP Port This parameter defines the local RTP port pair used by the HandyTone ATA. The
default value for FXO port is 5012.
Use Random Port 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.
Refer to Use Target
Contact
Default is No . If set to YES, then for Attended Transfer, the “Refer-To” header uses the
transferred ta rget’s contact header information.
Remove OBP from Route
Header
Default is No. If set to Yes, the Outbound Proxy will be removed from the 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
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 reliable usage.
SIP T2 Interval Maximum retransmission interval for non-INVITE requests and INVITE responses.
SIP Timer D Set the SIP Timer D. Default is 0.
DTMF Payload Type Sends DTMF using RFC2833
Preferred DTMF method
(in listed order)
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.