Jensen Tools 100 Sereis user manual Mwi

BudgeTone-100 User Manual Grandstream Networks, Inc.

Local RTP port

Use Random port

NAT Traversal

Keep alive interval

Use NAT IP

Proxy-Require

Voice Mail User

ID

Subscribe for

MWI

Auto Answer

Offhook

Auto-Dial

Enable call features

Disable Call

Waiting

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.

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.

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.

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.

NAT IP address (WAN side) used in SIP/SDP message. Default is blank.

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. Required by some soft switch vendor like Nortel MCS.

User ID (extension or access number) of a 3rd 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.

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.

Default is No. When set to Yes, the phone will automatically pick up the call after a short beep and turn on the speaker.

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.

Default is Yes. Advance call features or feature codes functions (Star code, see Section 5.4 of this manual) are supported locally

Default is No. User can use * code to use this feature per call basis.

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