Grandstream Networks HT503 User Manual page 38

Tel URI
SIP Registration
Unregister on Reboot
Outgoing Call Without
Registration
Register Expiration
SIP registration failure
retry wait time
Local SIP Port
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
FIRMWARE VERSION 1.0.6.8
-SRV (DNS SRV resource records indicates how to find services for various protocols)
-NAPTR/SRV (Naming Authority Pointer according to RFC 2915)
One mode can be chosen for the client to look up server.
The default value is "A Record".
The default setting is "Disabled". If the phone has an assigned PSTN Number, this field
should be set to "User=Phone" then a "User=Phone" parameter will be attached to the
"From header" in the SIP request to indicate the E.164 number. If server supports TEL
URI format, then this option needs to be selected.
Controls whether the HT503 needs to send REGISTER messages to the proxy server.
The default setting is Yes.
Default is No. If set to Yes, the SIP user's registration information will be cleared on
reboot.
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.
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).
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.
Defines the local SIP port the HT503 will listen and transmit. The default value for FXS
port is 5062.
This parameter defines the local RTP-RTCP port pair used by the HandyTone ATA. It
is the base RTP port for FXO channel.
When configured, the FXO port will use this port _value for RTP and the port_value+1
for its RTCP.
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)
HT503 USER MANUAL
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