Planet VIP-2140PT, Руководство пользователя

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High Definition Color PoE IP Phone with Dual Display
VIP-2140PT
65
Parameters Description
Disable Call Forward on No Answer Set the feature code to dial to the server.
Enable Blocking Anonymous Call Set the feature code to dial to the server.
Disable Blocking Anonymous Call Set the feature code to dial to the server.
Specific Server Type
Set the line to collaborate with specific server type; this is to
be address more in “VIP-2140PT Administration Guide”.
Registration Expiration Set the SIP expiration interval.
Use VPN Set the line to use VPN restrict route.
Use STUN Set the line to use STUN for NAT traversal.
Convert URI Convert non digit and alphabet characters to %hh hex code.
DTMF Type Set the DTMF type to be used for the line.
DTMF SIP Info Mode Set the SIP Info mode to send ‘*’ and ‘#’ or ‘10’ and ‘11’.
Transport Protocol Set the line to use TCP or UDP for SIP transmission.
SIP Version Set the SIP version.
Caller ID Header Set the Caller ID Header.
Enable Strict Proxy
Enables the use of strict routing. When the phone receives
packets from the server, it will use the source IP address, not
the address in via field.
Enable user = phone Sets user = phone in SIP messages.
Enable SCA Enable/Disable SCA (Shared Call Appearance).
Enable BLF List Enable/Disable BLF List.
Enable DNS SRV
Set the line to use DNS SRV which will resolve the FQDN in
proxy server into a service list.
Keep Alive Type
Set the line to use dummy UDP or SIP OPTION packet to
keep NAT pinhole open.
Keep Alive Interval Set the keep alive packet transmitting interval
Sync Clock Time Time Sync with server
Enable Session Timer
Set the line to enable call ending by session timer
refreshment. The call session will be ended if there is not new
session timer event update received after the timeout period.
Session Timeout Set the session timer timeout period.
Enable Rport Set the line to add rport to SIP headers.
Enable PRACK Set the line to support PRACK SIP message.
Keep Authentication
Keep the authentication parameters from previous
authentication.
Auto TCP
Using TCP protocol to guarantee usability of transport for SIP
messages above 1500 bytes.