Grandstream Networks GRP261 Series Administration Manual page 40

On
Off
Target
Call Forward No Answer
On
Off
Target
Call Forward No Answer Timeout (s)
Accounts �� Account Swap
Swap Account Settings
Phone Settings Page Definitions
Phone Settings �� General Settings
Basic Settings
Local RTP Port
Local RTP Port Range
Use Random Port
Enable Fix for RTP Timestamp Jump
Keep-alive Interval
Configures Call Forward Busy feature code to activate busy call forwarding.
Configures Call Forward Busy feature code to deactivate busy call forwarding.
Configures the extension for the call to be forwarded to.
Configures Call Forward No Answer feature code to activate no answer call
forwarding.
Configures Call Forward Busy feature code to deactivate busy call forwarding.
Configures the extension for the call to be forwarded to.
Configures the timeout (in seconds) before the call is forwarded when there is
no answer. Valid range is 1 to 120. The default setting is 12 seconds.
Allows users to swap the two accounts that they have configured. This will
Increase the flexibility of account management.
Note: Make sure to press "Start" to complete the process.
Table 13: Account Page Definitions
This parameter defines the local RTP port used to listen and transmit. It is the base
RTP port for channel 0. When configured, channel 0 will use this port _value for RTP;
channel 1 will use port_value+2 for RTP. Local RTP port ranges from 1024 to 65400
and must be even.
Default value is 5004.
Gives users the ability to define the parameter of the local RTP port used to listen
and transmit. This parameter defines the local RTP port from 48 to 10000. This
range will be adjusted if local RTP port + local RTP port range is greater than 65486.
Default setting is 200.
When set to "Yes", this parameter will force random generation of both the local SIP
and RTP ports. This is usually necessary when multiple phones are behind the same
full cone NAT. The default setting is "Yes"
Note: This parameter must be set to "No" for Direct IP Calling to work.
Makes RTP timestamps be continuous, if there is audio loss caused by timestamp
jump.
Default is "No"
Specifies how often the phone sends a blank UDP packet to the SIP server to keep
the "ping hole" on the NAT router to open. The default setting is 20 seconds.
The valid range is from 10 to 160.