Alex
2004-Jul-26 22:57 UTC
[Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9
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.
Brian Capouch
2004-Jul-26 23:08 UTC
[Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9
Alex wrote:> It gets definitely better every day. > > List of bug fixes follows: > > Release 1.0.5.9 7/26/2004Got a link to the firmware? I can't find *any* firmware on their site anymore, via the navigation space. That said, I'm sure someone will correct me. Thx. B.
Dave Cotton
2004-Jul-26 23:23 UTC
[Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9
On Tue, 2004-07-27 at 01:08 -0500, Brian Capouch wrote:> Alex wrote: > > It gets definitely better every day. > > > > List of bug fixes follows: > > > > Release 1.0.5.9 7/26/2004 > > Got a link to the firmware? >Yes in his excitement he did forget that bit of info. :) I just pulled it from http://www.hellofone.com/downloads.html Dave
Jean-Yves Avenard
2004-Jul-26 23:36 UTC
[Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 People have mentioned a lot of crashes making the phone unusable with this totally *unofficial* firmware Check the distribution list for more information Jean-Yves On 27/07/2004, at 4:23 PM, Dave Cotton wrote:> I just pulled it from > > http://www.hellofone.com/downloads.html >- --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBBff2XeDVKqIr3GURAtndAJ4rLXpCjUxT3x9ujFf9n/s4IOtldQCghqoP TwReRmYfTsQc/XlzY6i9QL4=bFot -----END PGP SIGNATURE-----
shabanip
2004-Jul-27 00:34 UTC
[Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9
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. > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Holger Schurig
2004-Jul-27 01:32 UTC
[Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9
On Tuesday 27 July 2004 07:57, Alex wrote:> 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 enoughWhat a support !!! Pro: they fixed a bug reported by me in a quite short time Con: unfortunately, they failed to inform me that it has been fixed and where I can get the new firmware :-(
Tony Mountifield
2004-Jul-27 04:07 UTC
[Asterisk-Users] Re: New Beta version of Grandstream Firmware 1.0.5.9
In article <1090909420.2793.2.camel@robinhood.linuxautrement.local>, Dave Cotton <dcotton@linuxautrement.com> wrote:> On Tue, 2004-07-27 at 01:08 -0500, Brian Capouch wrote: > > Alex wrote: > > > It gets definitely better every day. > > > > > > List of bug fixes follows: > > > > > > Release 1.0.5.9 7/26/2004 > > > > Got a link to the firmware? > > I just pulled it from > > http://www.hellofone.com/downloads.htmlSo why is it that all the recent firmwares (anything after about 1.0.4.55) have never been available from Grandstream, but only from Hellofone? Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org