Filter
Top Products
Grandstream Networks, Inc. HT503 User Manual Page 33 of 38
Firmware 1.0.4.2 Last Updated: 06/2011
iLBC Payload Type:
This defines payload type for iLBC. Default value is 97. The valid range is between 96
and 127.
AAL2-G726-16 Payload
Type
Defines payload type for AAL2-G726-16. Default value is 100. Range is from 96 to
127.
AAL2-G726-24 Payload
Type
Defines payload type for AAL2-G726-24. Default value is 99. Range is from 96 to 127.
AAL2-G726-32 Payload
Type
Defines payload type for AAL2-G726-24. Default value is 104. Range is from 96 to
127.
AAL2-G726-40 Payload
Type
Defines payload type for AAL2-G726-40. Default value is 103. Range is from 96 to
127.
V
AD Default is No. VAD allows detecting the absence of audio and conserves bandwidth by
preventing the transmission of “silent packets” over the network.
Symmet
r
ic 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 Tone 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.
Caller ID Scheme
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, & NTT Japan
FSK Caller ID minimum
RX Level (dB)
An adjustable value for the Caller ID signal to help this device to recognize Caller ID
from different networks. (-96 -0dB. Default -40dB)
FSK Caller ID Seizure
Bits
Default is: 70bits. Range is from 0 to 800bits.
FSK Caller ID mark bits
Default is: 40bits. Range is from 1 to 800bits.
Caller ID Transport Type
According to customer’s choice CID information will be transferred from PSTN network
to VoIP network using following rules:
1. via SIP from - PSTN CID is in the SIP From field
2. via P-Asserted-Identity - SIP From field uses the pre-configured account user
Id. PSTN CID is in the P-Asserted-Identity field
3. Send anonymous - SIP From field uses "anonymous". PSTN CID is put in the
P-Asserted-Identity field
4. Disable - PSTN CID will not be sent. SIP From field uses the pre-configured
account user ID
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
Voice path volume adjustment.
• RX is a gain level for signals transmitted by FXO (FXO-To-VoIP volume ) ,
• TX is a gain level for signals received by FXO( FXO-To-PSTN volume).
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 FXO Port Configuration page. These
parameters affects call volume ONLY for calls placed to/from PSTN and VoIP
networks.
If call volume is too low when using VoIP extension, adjust volume using the Rx Gain
Level parameter under the FXO Port Configuration page.
If voice volume is too low at the other end (PSTN side), user may increase the far end
volume using the Tx Gain Level parameter under the FXO Port Configuration page.