Grandstream UCM6100 User Manual for 1.0.9.25 - Manual (Page 158)

Firmware Version 1.0.9.25 UCM6100 Series IP PBX User Manual Page 157 of 303
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension’s CID when
the extension has CID configured. The default setting is “No”.
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.
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