Grandstream Networks UCM6200 Series, Руководство пользователя

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UCM6200 Series User Manual
Version 1.0.20.38
devices behind NAT. If there is one-way audio issue, usually it is 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.
Important Note: When making outgoing calls, the following priority order rule will
be used to determine which CallerID will be set before sending out the call :
•
CID from inbound call ( Keep Original CID Enabled) → Trunk
Username/CallerID ( Keep Trunk CID Enabled) → DOD → Extension CallerID
Number → Trunk Username/CallerID ( Keep Trunk CID Disabled) → Global
Outbound CID.
CallerID Name
Configure the name of the caller to be displayed when the extension has no
CallerID Name configured.
Transport
Configure the SIP transport protocol to be used in this trunk. The default setting is
"UDP".
•
UDP
•
TCP
•
TLS
Direct Callback
Allows external numbers the option to get directed to the extension that last called
them.
For Example: User 2002 has dialed external number 061234575 but they were not
reachable, once they have received missed call notification on their phone, they
would mostly call back the number, if the option “Direct Callback” is enabled then
they will be directly bridged to user 2002 regardless of the configured inbound
destination.
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.265, H.263, H.263p,RTX, OPUS and VP8.
DID Mode Configure where to get the destination ID of an incoming SIP call, from SIP