Jensen Tools 100 Sereis - Manual (Page 34)

BudgeTone-100 User Manual
Grandstream Networks, Inc.
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Local RTP port This parameter defines the local RTP-RTCP port pair the IP phone will listen
and transmit. It is the base RTP port for channel 0. When configured, channel 0
will use this port_value for RTP and the port_value+1 for its RTCP; channel 1
will use port_value+2 for RTP and port_value+3 for its RTCP.
The default value is 5004.
Use Random port Default No. If set to Yes, the device will pick randomly generated SIP and RTP
ports. This is usually necessary and useful when multiple IP Phones are behind
the same full cone NAT router.
NAT Traversal Defines whether the NAT traversal mechanism is activated.
It should be set to YES if the device is behind NAT router.
If Outbound Proxy is NOT configured, STUN server needs to be set to activate
STUN detection mechanism. Usually ITSP will provide these settings for device
to work properly behind NAT/Firewall
If this field is set to “Yes” without STUN server, then the device will
periodically (every Keep-alive interval) send a dummy UDP packet to the SIP
server to pinhole the NAT in the router side.
Keep alive
interval Default is 20 seconds. The interval of sending dummy UDP packet to keep NAT
“pin hole” open in the router side. Min. value is 10 seconds.
Use NAT IP NAT IP address (WAN side) used in SIP/SDP message. Default is blank.
Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Required by some soft switch vendor like Nortel MCS.
Voice Mail User
ID User ID (extension or access number) of a 3 rd party VoiceMail system where the
user may have an account. By defining it, user presses the “MESSAGE” button
on the phone, an INVITE message will send to that ID/number to allow the user
to retrieve VM.
Subscribe for
MWI Default is No. When set to Yes, a SUBSCRIBE for Message Waiting Indication
will be sent periodically to server. BT-100 support both synchronize and non-
synchronized SUBSCRIBE SIP message.
Auto Answer Default is No. When set to Yes, the phone will automatically pick up the call
after a short beep and turn on the speaker.
Offhook
Auto-Dial This parameter allows the user to configure a User ID or extension number to be
automatically dialed upon off hook (like hot line). Please note that only the user
part of a SIP address needs to be entered here. The phone will automatically
append the “@” and the host portion of the corresponding SIP address.
Enable call
features Default is Yes. Advance call features or feature codes functions (Star code, see
Section 5.4 of this manual) are supported locally
Disable Call
Waiting Default is No. User can use * code to use this feature per call basis.