User's Manual
Firmware Version 1.0.0.5 
UCM6510 IP PBX User Manual 
Page 155 of 192 
A sample configuration could be as follows: 
192.168.0.0/16 
SIP SETTINGS/TOS 
Table 56: 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. 
RTP Hold Timeout 
When the call is on hold, if there is no RTP activity within the timeout (in 
seconds), the call will be terminated. This value of RTP Hold Timeout 










