370x SIP series Version 103 release note Release Date: 2005.08.01 Bug fixed: ============================================================================== 1. Fix some WEB display problems 2. The registration with Asterisk proxy. The registration with Asterisk proxy in previous version firmware will be failed caused by the Call ID issue. 3. PPPoE connection will be failed between 370x and some Linux PPPoE server. 4. If FXO greeting and then on-hook directly, it will cause 37xx reboot automatically. 5. Fix the issue about 37 will not send unregisteration under PPPoE mode. 6. Fix 37 fxs still hear ring back tone when he gets the 487 message. 7. The line2 ~line4 rtp port will error if open the ip-sharing function. Feature and Function added: ============================================================================== 1. Add the "drop" function after the routing table identification. Command: route -add prefix 100 dst 1 e164 1001 min 3 max 7 hunt 0 drop 1 2. Add the "pause" command to delay send the DTMF through the FXO port during the one-stage-dialing. Command: pause -add prefix 9 delay 1 3. "Delay" function for FXO port. When FXO port has an incoming call from IP side and signal connection is established, it will wait the dial tone from PSTN or PBX. But sometimes the dial tone is too late and some errors will occur. Now user can use "sysconf -delay 2" command to change the time waiting for dial tone, where 2 is the delay time, default value is 1 second. 4. Auth command to let Root user customized the Administrator user authority. Command: auth -sip 1 5. Auto or manual configures the DNS IP address when 350x is under DHCP and PPPoE mode. Command: ifaddr -autodns 6. Support the RFC2543 and RFC3261 transfers at the same time. 7. Add the 2nd proxy port function. Command: sip -px2 21.32.223.140 8. Add the 2nd proxy port change function. Command: sip -px2port 6000 9. Add the Outbound proxy port change function. Command: sip -outpxport 6000 Notes: ============================================================================== Modify the SIP local port range from 1024~65535 to 1~65535.