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

Alex asterisk-users at connectglobe.com
Mon Jul 26 22:57:31 MST 2004


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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