Sip settings/tos, Table 94: sip settings/tos – Grandstream UCM6510 User Manual User Manual
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Firmware Version 1.0.2.5
UCM6510 IP PBX User Manual
Page 265 of 313
Local Network Address
Specify a list of network addresses that are considered inside of the NAT
network. Multiple entries are allowed. If not configured, the external IP
address will not be set correctly.
A sample configuration could be as follows:
192.168.0.0/16
SIP SETTINGS/TOS
Table 94: SIP Settings/ToS
ToS For SIP
Configure the Type of Service for SIP packets. The default setting is None.
ToS For RTP Audio
Configure the Type of Service for RTP audio packets. The default setting is
None.
ToS For RTP Video
Configure the Type of Service for RTP video packets. The default setting is
None.
Default Incoming/Outgoing
Registration Time
Configure the default duration (in seconds) of incoming/outgoing
registration. The default setting is 120.
Max Registration/Subscription
Time
Configure the maximum duration (in seconds) of incoming registration and
subscription allowed by the UCM6510. The default setting is 3600.
Min Registration/Subscription
Time
Configure the minimum duration (in seconds) of incoming registration and
subscription allowed by the UCM6510. The default setting is 60.
Music On Hold Interpret
Configure the Music On Hold class for the channel when being put on hold.
This is used when the Music On Hold class is not set on the channel and
the peer channel placing the call on hold doesn't have "Music On Hold
Suggest".
Music On Hold Suggest
Configure the Music On Hold class to suggest to the peer channel when
placing the peer on hold.
Enable Relaxed DTMF
Select to enable relaxed DTMF handling. The default setting is "No".
DTMF Mode
Select DTMF mode to send DTMF. The default setting is RFC2833. If "Info"
is selected, SIP INFO message will be used. If "Inband" is selected, 64-kbit
codec PCMU and PCMA are required. When "Auto" is selected,
"RFC2833" will be used if offered, otherwise "Inband" will be used. The
default setting is "RFC2833".
RTP Timeout
During an active call, if there is no RTP activity within the timeout (in
seconds), the call will be terminated. The default setting is no timeout.
Note:
This setting doesn't apply to calls on hold.