Grandstream ht-503 ユーザーズマニュアル

 

Grandstream Networks, Inc. 

Page 24 of 37  

When set to yes the FXS port is activated. 

This field contains the URL string or the IP address (and port, if different from 5060) of 
the SIP proxy server. e.g., the following are some valid examples:  sip.my-voip-
provider.com, or sip:my-company-sip-server.com, or 192.168.1.200:5066 

IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border 
Controller. Used by ATA for firewall or NAT penetration in different network 
environment. If symmetric NAT is detected, STUN will not work and ONLY Outbound 
Proxy will work.  

User can select UDP or TCP or TLS. 

This setting decides whether the NAT traversal mechanism is activated. It should be 
set to “Yes” if the device is behind a NAT router. If no outbound proxy is configured, a 
STUN server needs to be set to activate STUN detection mechanism. Usually ITSP will 
provide these settings.  If this field is set to “Yes”, then the device will periodically send 
a dummy UDP packet to the SIP server to pinhole the NAT. 

User account information, provided by VoIP service provider (ITSP), usually has the 
form of digit similar to phone number or actually a phone number. This field contains 
the user part of the SIP address for this phone.  e.g., if the SIP address is 
sip:my_user_id@my_provider.com, then the SIP User ID is: my_user_id.   
Do NOT include the preceding “sip:” scheme or the host portion of the SIP address in 
this field. 

ID used for authentication, usually same as SIP user ID, but could be different and 
decided by ITSP. 

Password for ATA to register to (SIP) servers of ITSP. Purposely left blank once saved 
for security. Maximum length is 25.   

SIP service subscriber’s name which will be used for Caller ID display 

Default is No. If set to Yes the client will use DNS SRV to lookup for the SIP server.  

 If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP 
request 

This parameter 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 device will first send registration request to remove all 
previous bindings. Use only if proxy supports this remove bindings request.  

This parameter allows users place outgoing calls even when not registered (if allowed 
by ITSP) but it’s unable to receive incoming calls. 

This parameter allows the user to specify the time frequency (in minutes) the 
HandyTone ATA 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 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. 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. 

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.