DTMF SIP INFO
Mode
Set the SIP INFO mode to send ‘*’ and ‘#’ or ‘10’ and ‘11’
Transportation
Protocol
Set the line to use TCP or UDP for SIP transmission
Local Port
Set the Local Port
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 opened
Keep Alive Interval
Set the keep alive packet transmitting interval
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 in SIP headers
Enable PRACK
Set the line to support PRACK SIP message
Enable DNS SRV
Set the line to use DNS SRV which will resolve the FQDN in
proxy server into a service list
Auto Change Port
Enable/Disable Auto Change Port
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
Enable Feature
Sync
Feature Sycn with server
Enable GRUU
Support Globally Routable User-Agent URI (GRUU)
RTP Encryption
Enable RTP encryption such that RTP transmission will be
encrypted
RTP Encryption
Key
Set the pass phrase for RTP encryption
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