Grandstream Networks, Inc. |
WAN side http access:
PSTN access code:
Admin Password
SIP Server
Outbound Proxy
SIP User ID
Authenticate ID
No | Yes (WAN side access to http server will be rejected if set to No) |
(key pattern to use the PSTN line, default is "*00")
Update
All Rights Reserved Grandstream Networks, Inc. 2004
This contains the password to access the Advanced Web Configuration page. This field is case sensitive.
SIP Server’s URI or IP address
SIP Outbound Proxy Server’s URI or IP address
SIP service subscriber’s User ID
SIP service subscriber’s Authenticate ID. Can be identical to or different from SIP User ID
Authenticate | SIP service subscriber’s account password |
Password |
|
Name | SIP service subscriber’s name which will be used for Caller ID display |
Register | This parameter allows the user to specify the time frequency (in |
Expiration | 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). |
Local SIP port | This parameter defines the local SIP port the HandyTone ATA will listen |
| and transmit. The default value for FXS port is 5060. The default value |
| for FXO port is 5062. |
Local RTP port | This parameter defines the local |
| 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. The |
| default value for FXO port is 5008. |
Enable Call | Default is No. If set to Yes, Call Forwarding & |
Features | supported locally |
Send DTMF | This parameter controls how DTMF events are transmitted. There are 3 |
| ways: in audio which means DTMF is combined in audio signal (not very |
| reliable with |
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