[Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9

[shabanip]

I've found an incorrect timezone in GS firmware:
Tehran timezone is +3:30 not +3:00

> It gets definitely better every day.
>
> List of bug fixes follows:
>
> Release 1.0.5.9          7/26/2004
> 	If SIPRegister doesn't proceed due to conditions unmet, release
> channel resource
> 	Fix the LED flashing issue when connection to the SIP proxy is lost.
>
> 	Fix the issue where the device will not resume registration when it
> loses connection to the outbound proxy for some time.
> 	Fixed the registration interval overflow issue
> 	Fixed the no-host-name in REGISTER message when configured using a
> customer's Perl script
> 	Fixed the bad To header in INVITE after receiving 302 response
> 	Fixed the wrong URI in ACK to non-2xx response
> 	Fixed the issue where 486 would lose registration when outbound
> proxy is configured and when NAT traversal is turned ON with STUN server
> field blank.
>
> Release 1.0.5.8	7/16/2004
> 	Fix the branch ID uniqueness issue of ACK to a 2xx response
> 	Fix the CSeq not incrementing issue associated with sending
>         DTMF via SIP INFO when user is dialing fast and response to
>         SIP INFO is not received fast enough
> 	Fix the bad To header field in our new INVITE request upon
>         receiving 302 response.
> 	Fix the issue that we do not respond to SIP INFO request.
> 	Do not play dial tone if registration is required and device
>         is not registered.
> 	fix the inaccuracy of the timer unit value that causes
>         registration to expire about 2% faster than normal
> 	fix the bug in parsing expire parameter and port when multiple
>         contact items are on the same line (in a same header field)
>         separated by comma.
> 	Fix some accidental issues that break call forwarding and
>         call transfer
>
> Release 1.0.5.7	7/8/2004
>
> 	Fix the issue where we only send ACK only once which causes
>         signaling failure if this ACK is not delivered (due to packet
>         loss, etc) to the callee.
> 	Enable the high-pass filter and post-filter of G723.
> 	Remove the unnecessary dial tone when a user presses *xx when local
>         call features are enabled
> 	If a symmetric NAT is detected, still use mapped IP:port instead of
>         using private IP.
> 	Allow access to 486's Web server using the WAN side IP from LAN port
> 	Send ACK to the server in stead of per Contact header upon receiving
>
>         3xx response to an INVITE.
>
>
>
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