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Newly add the test item for Asterisk IP PBX
1)Asterisk IP PBX server domain name: palmmicro.ddns.us
2)Test account: SIP 222,223;IAX2 422,423; The password of register and voice mail is 1234
3)If the above numbers are all in use, please write to support@palmmicro.com.cn for new test numbers
4)The settings of AR168X IP phone for test account
IAX2 protocol refer
http://www.palmmicro.com.cn/esetiax2_2.htm
SIP protocol refer
http://www.palmmicro.com.cn/esetsip8.htm
5)When IP phone logon success,Please dial the number 420 to more test and consult.
Newly AR1688 IP phone hardware reference update
AR1688 IP phone hardware reference design version 4.06 is ready now. It has the following benefits:
1)Improved the side-tone circuit,the background noise is reduced evidently.
2)5V DC power input is supported, user can use a 5V DC power supply instead of 9V adapter.
3)The reset circuit is modified to obtain more reliable reset.
All old PA1688 customers can now write to support@palmmicro.com.cn to ask for the new design. For new customers,
we will starting selling AR1688 chips in Oct 2006. (right after the long holidays in China), the design is free with any chip purchase.
Newly add program for 0.42 release download.(2008-12-16 update)
Changes and Fixes in 0.42 release:
1. Added Micrel KSZ8842 (2-port switch with 8 bit CPU bus) support
2. Added prefix to number entered through call log and phonebook
3. Bug fixed: hooking when phone tries to connect DHCP server at startup makes the phone COLD BOOT
4. Bug fixed: when entering call log by pressing 'Call' key, and then save a call log entry to phonebook will corrupt the settings
5. Bug fixed: when the phone first boot up, the DSP is not working properly, so there is no dial tone or ringing tone etc
6. Added OEM_VLAN compile option to make special VLAN testing software
7. Added OEM_DEBUG compile option to make special debug software
8. Added VLAN support for DM9003 and KSZ8842
9. Added extensive assembly optimization for RTL8019AS/DM9003/KSZ8842 network performance
10. Removed speech codec module related files, future speech codec module application will use IAX2 or RTP protocol for compressed voice data instead of none standard format
11. Bug fixed: when pressing dtmf key quickly, one side will stop hearing the incoming voice
12. Let timestamp field in the voice to reflect the actual time when the packet is being sent
13. Fixed a IAX2 call hold bug
14. Matthias Siegele fixed a DTMF receive bug using SIP INFO
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