Table 11: Fxo Port Settings - Grandstream Networks HT503 User Manual

On Hook Timing
Gain
Disable Line Echo
Canceller (LEC)
Ring Tones
Account Active
SIP Server
Failover SIP Server
Prefer Primary SIP
Server
Outbound Proxy
SIP Transport
NAT Traversal (STUN)
SIP User ID
Authenticate ID
Authenticate Password
Name
DNS mode
FIRMWARE VERSION 1.0.6.8
On-hook timing is the minimum time for an on-hook event to be validated.
Voice path volume adjustment.
•
Rx is a gain level for signals transmitted by FXS
•
Tx is a gain level for signals received by FXS.
Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume: -6dB.
User can adjust volume of call on either end using the Rx Gain Level parameter and
the Tx Gain Level parameter located on the FXS Port Configuration page.
If call volume is too low when using the FXS port (ie. the ATA is at user site), adjust
volume using the Rx Gain Level parameter under the FXS Port Configuration page.
If voice volume is too low at the other end, user may increase the far end volume using
the Tx Gain Level parameter under the FXS Port Configuration page.
Default is No. If set to "Yes" LEC will be disabled per call base. Recommended for
FAX/Data calls.
This function lets you configure ring or tone frequencies according to preference. By
default tones are set to North American frequencies. Frequencies should be
configured with known values to avoid high pitch sounds.

Table 11: FXO PORT Settings

When set to "Yes" the FXO port is activated.
SIP Server's IP address or Domain name provided by VoIP Service Provider.
This Field contains the URL or the IP address of a second SIP server, this one will be
used in case the device loses the connection with the first server.
Default is no. If set to yes it will register to Primary Server if registration with Failover
server expires
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.
User can select UDP, TCP or TLS
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.
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.
User account information, provided by VoIP service provider (ITSP). Usually in the form
of digit similar to phone number or actually a phone number.
The SIP service subscriber's ID used for authentication. Can be identical to or different
from SIP User ID.
SIP service subscriber's account password.
SIP service subscriber's name for Caller ID display.
One from the 3 modes available for "DNS Mode" configuration:
-A Record (for resolving IP Address of target according to domain name)
HT503 USER MANUAL
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