Grandstream Networks HT503 user manual Caller ID Scheme, FSK Caller ID minimum, RX Level dB

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	Defines payload type for AAL2-G726-16.	Default value is 100.	Range is from 96 to
Type	127.
AAL2-G726-24 Payload	Defines payload type for AAL2-G726-24.	Default value is 99.	Range is from 96 to 127.
Type
AAL2-G726-32 Payload	Defines payload type for AAL2-G726-24. Default value is 104.		Range is from 96 to
Type	127.
AAL2-G726-40 Payload	Defines payload type for AAL2-G726-40.	Default value is 103.	Range is from 96 to
Type	127.
VAD	Default is No. VAD allows detecting the absence of audio and conserves bandwidth 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 Tone Detection	Default is Callee. This decides whether Caller or Callee sends out the re-invite for T.38
Mode	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	An adjustable value for the Caller ID signal to help this device to recognize Caller ID
RX Level (dB)	from different networks. (-96 -0dB. Default -40dB)
FSK Caller ID Seizure	Default is: 70bits. Range is from 0 to 800bits.
Bits
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.
Grandstream Networks, Inc.	HT503 User Manual		Page 33 of 38
	Firmware 1.0.4.2		Last Updated: 06/2011