Grandstream UCM6100 User Manual for 1.0.9.25 User Manual
Page 137
Firmware Version 1.0.9.25
UCM6100 Series IP PBX User Manual
Page 136 of 303
option is disabled.
Music On Hold
Select which Music On Hold class to suggest to extensions when putting
them on hold.
Enable LDAP
If enabled, the batch added extensions will be added to LDAP Phonebook
PBX list; if disabled, the batch added extensions will be skipped when
creating LDAP Phonebook.
SIP Settings
NAT
Use NAT when the PBX is on a public IP communicating with devices
hidden behind NAT (e.g., broadband router). If there is one-way audio
issue, usually it's related to NAT configuration or Firewall's support of SIP
and RTP ports. The default setting is enabled.
Can Reinvite
By default, the PBX will route the media steams from SIP endpoints
through itself. If enabled, the PBX will attempt to negotiate with the
endpoints to route the media stream directly. It is not always possible for
the PBX to negotiate endpoint-to-endpoint media routing. The default
setting is "No".
DTMF Mode
Select DTMF mode for the user 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.
Insecure
• Port: Allow peers matching by IP address without matching port
number.
• Very: Allow peers matching by IP address without matching port
number. Also, authentication of incoming INVITE messages is not
required.
• No: Normal IP-based peers matching and authentication of incoming
INVITE.
The default setting is "Port".
Enable Keep-alive
If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is "Yes".
Keep-alive Frequency
Configure the number of seconds for the host to be up for Keep-alive. The
default setting is 60 seconds.
TEL URI
If the end device/phone 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.
Other Settings