Filter
Top Products
Grandstream Networks, Inc. HT-386 User Manual Page 24 of 34
Firmware 1.0.3.64 Last Updated: 2/2007
INFO.
Send Flash Event
Default is NO. If set to yes, flash will be sent as DTMF event.
Enable Call Features
Default is Yes. Advance call features and feature codes functions are
supported locally
Use Bell-style
3-way Conference
If this parameter is set to “Yes”, user will be able to make Bellcore style 3-way
conference. *23 will be disabled.
Offhook
Auto-Dial
This parameter allows a user to configure a User ID or extension number to be
automatically dialed upon offhook. Please note that only the user part of a SIP
address needs to be entered here. The HT-386 will automatically append the
“@” and the host portion of the corresponding SIP address.
NOTE: Please write down the IP address of the ATA if you use this feature as
it will disable the IVR and the only way to access the HT-386 is via web
configuration page.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Disable Call Waiting
Default is No. User can use * code to use this feature per call basis.
NAT Traversal
This parameter defines whether the HT-386 NAT traversal mechanism will be
activated or not. If activated (by choosing “Yes”) and a STUN server is also
specified, then the HT-386 will behave according to the STUN client
specification. Under this mode, the embedded STUN client inside the HT-386
will attempt to detect if and what type of firewall/NAT it is sitting behind through
communication with the specified STUN server. If the detected NAT is a Full
Cone, Restricted Cone, or a Port-Restricted Cone, the HT-386 will attempt to
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
HT-386 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.
Preferred Vocoder
The HT-386 supports 6 different codec types including :
G.711 A/Ulaw，G.723.1, G.726, G.729A/B, iLBC.
A user can configure codecs in a preference list that will be included with the
same preference order in SDP message.
Voice Frames per TX
This field contains the number of voice frames to be transmitted in a single
packet. When setting this value, the user should be aware of the requested
packet time (used in SDP message) as a result of configuring this parameter.
This parameter is associated with the first codec in the above codec
Preference List or the actual used payload type negotiated between the 2
conversation parties at run time.
e.g., if the first codec is configured as G723 and the “Voice Frames per TX” is
set to be 2, then the “ptime” value in the SDP message of an INVITE request
will be 60ms because each G723 voice frame contains 30ms of audio.
Similarly, if this field is set to be 2 and if the first codec chosen is G729 or
G711 or G726, then the “ptime” value in the SDP message of an INVITE
request will be 20ms.
If the configured voice frames per TX exceeds the maximum allowed value,
the HT-386 will use and save the maximum allowed value for the
corresponding first codec choice. The maximum value for PCM is 10(x10ms)
frames; for G726, it is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames;
for G729/G728, 64 (x10ms) and 64 (x2.5ms) frames respectively. Please be
careful when massage those parameters.
G723 Rate:
Encoding rate for G723 codec. By default, 6.3kbps rate is set.
iLBC frame size:
iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might need
to be set.
iLBC payload type:
Payload type for iLBC. Default value is 97. The valid range is between 96 and