go to: http://www.grandstream.com/BETATEST/
Release 1.0.5.22 1/21/2005  Changed polarity reversal logic per customer request and fixed the polarity reversal issue  Add support for syslog server (HT286 only)  Change the choice between tftp upgrade and http upgrade to mutual exclusive  Fixed we cannot dial SIP call without using # key in HT486 Rev 2.0 when PSTN access code is non-default.  Fixed we do not use RFC2833 to send DTMF when the incoming SDP contains iLBC and the immediate next "a" line is not the fmtp line for iLBC  Fixed we use G.722/8000 instead of G.722/16000 (BT100 only). (Fixed compatibility problem with some other vendor)  Fixed: we will retry http download if we received TCP RST.  Send DHCP NAK if a DHCP request is not within our memory record (to force a restart of DHCP Discovery process).  Fix the issue related to Record Route under some situations  Fix the layer-3 TOS issue  Fix the port forwarding issue Release 1.0.5.19 12/3/2004  Do not display SIP authentication password on the HTML page  Release DHCP or PPPoE connection before rebooting  Re-enable fax tone detection upon which switching to PCM if previously using low bit rate codec  Add configuration support for European Caller ID (286 & 486 Rev 2.0 only)  Add system up time in the "status" page of the Web interface  Allow 96 to be used for DTMF payload type  Fix the DHCP length issue which causes our DHCP offer to be rejected by Linksys router when it is on the LAN side of 486/487  Add support for call waiting caller ID (HT286 and HT486 rev 2.0 only)  Add support for polarity reversal configuration parameter (HT286 and HT486 rev 2.0 only)  Fix the request URI and Route Set bug  Use "terminated" as Subscription-State for termination NOTIFY as transferee  Add register state information on the "Status" web page.  Release 1.0.5.18 11/14/2004  Change the factory default setting of "enable call feature" to TRUE; change the factory default setting of "NAT Traversal" to "Yes"; change the factory default setting of "Inter-key timeout" value to 4 seconds; change the factory default setting of "upgrade checking frequency" to 7 days.  Fix the issue (resulting from recent code change) that causes bad IP header checksum in HTTP Upgrade packet  Fix the issue that if no response is received due to bad HTTP GET packet or packet drops, HTTP upgrade retry will not happen and SIP registration will not happen or take very long time to happen  Support attended call transfer and server-side 3-way conferencing for Nortel  Fixed send NTP to STUN server when STUN server is in FQDN form.  Fixed dialing bad URI when offhook auto dial is enabled.  Fixed BT-100 dialing bad URI when using the message button.  Fixed BT-100 show only first caller in caller-history.  Allow lower case encoding in Replaces.  Change the wording of "do not disturb" to "disable call waiting" in Web interface  Support 3-page Web configuration interface  Add configuration parameter to support special feature for Nortel's MCS, Broadsoft (except HT486 rev 1.0)  Change the target MAC address from ff.ff.ff.ff.ff.ff to 00.00.00.00.00.00 in ARP request packet  Always Unregister (not all contacts but only the binding that it registered as) and re-register when "UPDATE" is pressed in Web configuration interface. (HT486 2.0 only)  Fixed transferee stops playing ring-back tone if transferor hang up before transfer target answers on blind-transfer.  Fixed answering ARP for IP address of the wrong port (eg. we answer to ARP for 192.168.2.1 even though the ARP comes from WAN port).  ALWAYS set the http upgrade URL to: fm.grandstream.com/gs and enable firmware upgrade (YES) upon reset to factory default.  Add support for 501 not implemented response  Fix the problem where in the Web user interface, pressing the UPDATE button will not get response if no parameter is changed  Fix the issue of exposed password on HTMLâwe will not display password in the Web interface and will not take empty password.  HTML 1.0.0.42 11/11/2004  Use new graphic user interface with 3 different tabs (status, basic settings and advanced settings)  add configuration parameter to support special feature for Nortel MCS, Howdy, etc. (HandyTone 486 Rev 2.0 only)  do not display password on HTML pages  Key bug fixes and enhancements since Release 1.0.5.11  Release 1.0.5.16 10/18/2004  Improved routing performance for HTTP traffic  Support enable/disable of caller ID, and call waiting via keypad  Fix the issue related to processing encrypted configuration file  Fix the issue causing 400 bad response to be sent for NOTIFY after blind transfer  Support Fragmented UDP frames for SIP processing  Fix the missing Contact field for SUBSCRIBE and INFO request  Add support for upgrading firmware or modifying configuration via http. Support file path for http url.  Add logic to detect and decline duplicate IP during DHCP application stage.  Add call time ticking display for callee (BudgeTone 100 only)  Support file content authentication checking using AES during firmware upgrade  Support for release of IP upon detecting the link is down for more than 15 seconds and re-application for IP address as soon as the link is up again  Support attended transfer and Replace header  Support Proxy-Require header and its configurable content  Support pre-scheduled firmware upgrade checking frequency and add control flag to allow or prohibit auto firmware upgrade.  Support configurable PSTN access key string  Support 2 different Web login screens (1 for end user and the other for admin). The login interface is shared between 2 different user modes but the edit screen is different. Add port forwarding, DMZ and DHCP server related configuration options to end user configuration screen  Fix the loss of registration issue  Fix the issue that a HOLD initiated by 1 party can be released by the other when the other party presses HOLD and then releases the HOLD.  Fixed the extra "@" character in "From" header when user ID is blank.  Fix the issue related to negotiating and using the right MTU when remote end uses a smaller MTU (HT486 only)  Fix the PPPoE link state monitoring issue if CHAP is used.  Fix the issue where our RTP sequence ID is randomly changed when a 183 response is initially received and then a 200 OK response is received.  Fixed layer 2 QoS (VLAN and 802.1p) issue  Maintain the credential information for all subsequent REGISTER after the initial registration is successful, as opposed to restart challenge-authenticate cycle for each new REGISTER transaction  Fix the "reset to factory default" which is recently broken  Increase the timeout value for PPPoE call establishment. This will better accommodate some Chinese DSL modems' slow response. Also reset IP upon detecting the pppoe link is down for more than 15 seconds.  Fix the issue where improperly deleting an un-initialized timer can cause timer malfunction  Fix the issue that PPP PAP timer interferes with CHAP negotiation  Fix the issue related to processing multiple IP addresses of DNS A record response  Fix the issue that PCMU is always included in SDP even if it is never configured on HandyTone products  Fix a bug to better handle very long Contact header, e.g., 500+ characters long  Fix the ptime negotiation issue where we didn't use the default ptime when the remote end responds with a codec that is different from our first offered codec and which has no ptime in its SDP  Fix the issue that after firmware upgrade the device should (but previously does not) reboot automatically. On Fri, 18 Feb 2005 12:01:22 -0500, dean collins <[EMAIL PROTECTED]> wrote: > > > 1.0.5.22 is available for downloading here http://gs-firmware.gratissip.dk/ > > I don't know why these are available if Grandstream don't update their > webpages to indicate newer versions are available. > > > > > ________________________________ > > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Josh Wilson > Sent: Friday, February 18, 2005 10:56 AM > To: [email protected] > Subject: Re: [Asterisk-Users] Budgetone 101 > > > > > 1.0.5.16 - the latest version. > > >>> Michael 'Moose' Dinn <[EMAIL PROTECTED]> 2/18/2005 8:14:41 AM > >>> > > > What firmware are you running on your 101? > > On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: > > Everytime that I make a call to a Budgetone 101 phone. I always see the > > following: > > > > -- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack > > -- Called 1000 > > -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 > > -- SIP/1000-465e is busy > > > > I can use X-Lite all the time to make a call without a problem, but any > > of the budgetone 101 phones I can not get to work anymore. Anybody know > > how to fix this? > > > > Josh > > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
