- Grandstream Networks, Inc. Analog Telephone Adaptor User Manual HandyTone-488
HandyTone-488 User Manual Grandstream Networks, Inc.
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WAN side http
access:
No Yes (WAN side access to http server will be rejected if set to No)
PSTN access code:
(key pattern to use the PSTN line, default is "*00")
Update
All Rights Reserved Grandstream Networks, Inc. 2004
Admin Password
This contains the password to access the Advanced Web Configuration
page. This field is case sensitive.
SIP Server
SIP Server’s URI or IP address
Outbound Proxy
SIP Outbound Proxy Server’s URI or IP address
SIP User ID
SIP service subscriber’s User ID
Authenticate ID
SIP service subscriber’s Authenticate ID. Can be identical to or different
from SIP User ID
Authenticate
Password
SIP service subscriber’s account password
Name
SIP service subscriber’s name which will be used for Caller ID display
Register
Expiration
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).
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 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. The
default value for FXO port is 5008.
Enable Call
Features
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are
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 low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.