User manual GRANDSTREAM HT-286

[. . . ] User Manual HandyTone-286 Rev 3. 0 Analog Telephone Adaptor For Firmware Version 1. 0. 8. 32 Grandstream Networks, Inc. www. grandstream. com HandyTone -286 User Manual Grandstream Networks, Inc. Table of Contents 1 2 3 WELCOME. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 INSTALLATION . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5 WHAT IS INCLUDED IN THE PACKAGE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 3. 1 3. 2 4 4. 1 4. 2 5 SAFETY COMPLIANCES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . [. . . ] Please note that this feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling. Dial Plan Prefix This value contains the dial plan prefix string (typically an ASCII numeric string). If it is not blank, then this string will added to the dialed number. Timeout Use # as This parameter allows the user to configure the "#" key to be used as the Send Key "Send"(or "Dial") key. Once set to "Yes", pressing this key will immediately trigger the sending of dialed string collected so far. In this case, this key is essentially equivalent to the "(Re)Dial" key. If set to "No", this # key will then be included as part of the dial string to be sent out. Local SIP port This parameter defines the local SIP port the IP phone will listen and transmit on. Local RTP port This parameter defines the local RTP-RTCP port pair the IP phone will listen and transmit on. 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. Use Random Port This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple IP phones are behind the same NAT. 28 HandyTone -286 User Manual Grandstream Networks, Inc. NAT Traversal This parameter defines whether the phone NAT traversal mechanism will be activated or not. If activated (by choosing "Yes") and a STUN server is also specified, then the phone will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the phone will attempt to detect if and what type of firewall/NAT it is behind by sending appropriate request to the specified STUN server. If this field is set to "Yes" with no specified STUN server, then the phone will only periodically (every 20 seconds by default) send a blank UDP packet (with no payload data) to the SIP server to keep the mapped port open on the NAT. keep-alive interval The HandyTone-286 sends a UDP package to the SIP server periodically in order to keep the port open on the router. This parameter defines the interval time that HT286 send the UDP package. Use NAT IP NAT IP address is used in SIP/SDP message. Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. SUBSCRIBE for Default is "No". When set to "Yes" a SUBSCRIBE for Message Waiting MWI Indication will be sent periodically Offhook This parameter allows the user to configure a User ID or extension number to Auto-Dial be automatically dialed upon offhook. [. . . ] It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP. RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889 SDP Session Description Protocol, is a format for describing streaming media initialization parameters. [. . . ]