Grandstream ht-503 ユーザーズマニュアル
Grandstream Networks, Inc.
HT503 User Manual
Page 28 of 37
T
ABLE
11:
HT503
FXO
PORT
S
ETTINGS
P
AGES
D
EFINITIONS
Account Active
When set to “Yes” the FXO port is activated.
SIP Server
SIP Server’s IP address or Domain name provided by VoIP Service Provider.
Outbound Proxy
IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border
Controller. Used by HT503 for firewall or NAT penetration in different network
environments. If symmetric NAT is detected, STUN will not work and ONLY way to
correct the problem is to use the outbound proxy.
Controller. Used by HT503 for firewall or NAT penetration in different network
environments. If symmetric NAT is detected, STUN will not work and ONLY way to
correct the problem is to use the outbound proxy.
SIP Transport
User can select UDP, TCP or TLS
NAT Traversal (STUN)
This parameter defines whether or not the HT503 NAT traversal mechanism is
activated. If set to “Yes” with a STUN server also specified, the HT503 will perform
according to the STUN client specification. Using this mode, the embedded STUN
client will detect if and what type of firewall/NAT is being used.
activated. If set to “Yes” with a STUN server also specified, the HT503 will perform
according to the STUN client specification. Using this mode, the embedded STUN
client will detect if and what type of firewall/NAT is being used.
If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the
HT503 will use its mapped public IP address and port in all of its SIP and SDP
messages. If the NAT Traversal field is set to “Yes” with no specified STUN server, the
HT503 will periodically (every 20 seconds or so) send a blank UDP packet (with no
payload data) to the SIP server to keep the “hole” on the NAT open.
HT503 will use its mapped public IP address and port in all of its SIP and SDP
messages. If the NAT Traversal field is set to “Yes” with no specified STUN server, the
HT503 will periodically (every 20 seconds or so) send a blank UDP packet (with no
payload data) to the SIP server to keep the “hole” on the NAT open.
SIP User ID
User account information, provided by VoIP service provider (ITSP). Usually in the form
of digit similar to phone number or actually a phone number.
of digit similar to phone number or actually a phone number.
Authenticate ID
The SIP service subscriber’s ID used for authentication. Can be identical to or different
from SIP User ID.
from SIP User ID.
Authenticate Password
SIP service subscriber’s account password.
Name
SIP service subscriber’s name for Caller ID display.
Use DNS SRV
Default is No. If set to “Yes” the client will use DNS SRV to look up the server.
User ID is Phone Number
If the HT503 has an assigned PSTN telephone number, this field should be set to
“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be
attached to the “From” header in SIP request.
“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be
attached to the “From” header in SIP request.
SIP Registration
Controls whether the HT503 needs to send REGISTER messages to the proxy server.
The default setting is Yes.
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.
reboot.
Outgoing Call Without
Registration
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.
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).
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).
Local SIP Port
Defines the local SIP port the HT503 will listen and transmit. The default value for FXS
port is 5062.
port is 5062.
Local RTP Port
This parameter defines the local RTP-RTCP port pair used by the HandyTone ATA. It
is the base RTP port for FXO channel.
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.
for its RTCP.
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.
set to Yes. This is usually necessary when multiple HT503 units are behind the same
NAT.
Refer to Use Target
Contact
Contact
Default is No. If set to YES, then for Attended Transfer, the “Refer-To” header uses the
transferred target’s contact header information.
transferred target’s contact header information.
Validate incoming
message
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.
according to RFC rules. If message will not pass validation process, call will be
rejected.
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.
If the network latency is high, select larger value for reliable usage.
Firmware 1.0.0.9
Last Updated: 9/2007