Grandstream Networks HT503 FXS/FXO Benutzerhandbuch

 

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 lets 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 the HandyTone ATA will listen 
and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use 
this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use 
port_value+2 for RTP and port_value+3 for its RTCP. The default value for FXS port is 
5004.  

Default 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. 

This parameter sets the payload type for DTMF using RFC2833 

Send DTMF as inband (in-audio). 

Send DTMF via RTP (According to RFC 2833) 

Send DTMF via SIP INFO message. 

Default is No. If set to yes, flash will be sent as DTMF event. 

Default is Yes. Toggles support for advanced call features and star code functions. 

This parameter allows users to configure a User ID or extension number to be 
automatically dialed upon offhook. Please note that only the user part of a SIP address 
needs to be entered here. The HT503 will automatically append the “@” and the host 
portion of the corresponding SIP address.  
Note: Please write down the IP address of the ATA if you use this feature as it 
will prevent access to the IVR and the only way to access the device configuration will 
be via the web configuration page. 

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. 

NAT IP address used in SIP/SDP message.  Default is blank 

Customizes Ring Tones 1 to 3 with an associated Caller ID.  When selected, the device 
will ONLY use this ring tone when the incoming call is set up for Caller ID.  The System 
Ring Tone is used for all other calls.  When selected with no Caller ID configured, the 
selected ring tone will be used for all incoming calls. 

Default is No. 

Default is No.  This is to disable the stutter Call Waiting Tone during a CWC. The 
CWCID will still be displayed. 

Sets the time in which an incoming call will stop ringing when not picked up. 

Default is 4 seconds. 

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 

Grandstream Networks, Inc. 

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