User's Manual Part 1
Firmware Version 1.0.0.1  UCM6200 Series IP PBX User Manual  Page 158 of 320
NAT 
Turn on this option when the PBX is using public IP and communicating 
awith devices behind NAT. If there is one-way audio issue, usually it’s 
related to NAT configuration or SIP/RTP port configuration on the firewall.
Disable This Trunk 
If selected, the trunk will be disabled. 
Note: 
If a current SIP trunk is disabled, UCM will send UNREGISTER message 
(REGISTER message with expires=0) to the SIP provider. 
TEL URI 
If the trunk has an assigned PSTN telephone number, this field should be 
set to "User=Phone". Then a "User=Phone" parameter will be attached to 
the Request-Line and TO header in the SIP request to indicate the E.164 
number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP 
request. The default setting is disabled. 
Caller ID 
Configure the Caller ID. This is the number that the trunk will try to use 
when making outbound calls. For some providers, it might not be possible 
to set the CallerID with this option and this option will be ignored. 
When making outgoing calls, the following rules are used to determine 
which CallerID will be used if they exist: 
  The CallerID configured for the extension will be looked up first. 
  If no CallerID configured for the extension, the CallerID configured for 
the trunk will be used. 
  If the above two are missing, the "Global Outbound CID" defined in 
Web GUI->PBX->Internal Options->General will be used. 
CallerID Name 
Configure the name of the caller to be displayed when the extension has 
no CallerID Name configured. 
Auto Record 
Enable automatic recording for the calls using this trunk (for SIP trunk 
only). The default setting is disabled. The recording files can be accessed 
under web GUI->CDR->Recording Files. 
Advanced Settings 
Codec Preference 
Select audio and video codec for the VoIP trunk. The available codecs 
are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, 
iLBC, ADPCM, H.264, H.263, H.263p and VP8. 
DID Mode 
Configure where to get the destination ID of an incoming SIP call, from 
SIP Request-line or To-header. The default is set to "Request-line". 
DTMF Mode 
Configure the default DTMF mode when sending DTMF on this trunk. 
  Default: The global setting of DTMF mode will be used. The global 
setting for DTMF Mode setting is under web UI->PBX->SIP 
Settings->ToS. 
  RFC2833: Send DTMF using RFC2833. 










