Grandstream Networks HT503 FXS/FXO Network Card User Manual


 
by preventing the transmission of “silent packets” over the network.
Symmetric RTP
Default is “No.” When set to “Yes” the device will change the destination to send RTP
packets to the source IP address and port of the inbound RTP packet last received by
the device.
Fax Mode
T.38 (Auto Detect) FoIP by default, or fax Pass-Through (must use PCMU/PCMA)
Fax Detection Mode
Default is Callee. This decides whether Caller or Callee sends out the re-invite for T.38
or Fax Pass-Through.
Jitter Buffer Type
Select either Fixed or Adaptive based on network conditions.
Jitter Buffer Length
Select Low, Medium, or High based on network conditions.
SRTP Mode
Secure RTP protocol used for media transmission over VoIP. Disabled by default.
Other modes are: enabled but not forced & enabled and forced.
SLIC Setting
Dependent on standard phone type (and location).
Called ID Scheme
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, & NTT Japan
Caller ID TX Level (dB)
A value for Caller ID information sent by a phone connected to the FXS port.
(-96 – 0dB. Default -14dB)
Polarity Reversal
If set to “Yes”, polarity will be reversed upon call establishment and termination.
Default is No.
Loop Current Disconnect
Set it to “Yes” of the traditional PBX you are using with HT503 uses this method fir
signaling call termination. Default is No.
Loop Current Disconnect
Duration
A configurable period of time in which the FXS port will drop off voltage on the line to
indicate to the local party that the call is disconnected from the remote side.
(100-10000 ms. Default 200 ms)
Hook Flash Timing
The time period when the cradle is pressed (Hook Flash) to simulate a FLASH. Adjust
this time value to prevent unwanted activation of the Flash/Hold and automatic phone
ring-back.
Gain
Handset volume adjustment.
RX is for receiving volume,
TX is for transmission volume.
Default values are 0dB for both parameters. Loudest volume: +6dB Lowest volume: -
6dB.
Call Progress/ Ring
Tones
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: HT503 FXO PORT SETTINGS PAGES DEFINITIONS
Account Active
When set to “Yes” the FXO port is activated.
SIP Server
SIP Server’s IP address or Domain name provided by VoIP Service Provider.
Outbound Proxy
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.
SIP Transport
User can select UDP, TCP or TLS
NAT Traversal (STUN)
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.
SIP User ID
User account information, provided by VoIP service provider (ITSP). Usually in the form
of digit similar to phone number or actually a phone number.
Authenticate ID
The SIP service subscriber’s ID used for authentication. Can be identical to or different
from SIP User ID.
Grandstream Networks, Inc. HT503 User Manual Page 26 of 35
Firmware 1.0.0.6 Last Updated: 6/2007