[Asterisk-Users] Budgetone 101
Erick Perez
eaperezh at gmail.com
Fri Feb 18 12:24:27 MST 2005
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 HTML—we 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 <dean at collins.net.pr> 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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Josh Wilson
> Sent: Friday, February 18, 2005 10:56 AM
> To: Asterisk-Users at lists.digium.com
> Subject: Re: [Asterisk-Users] Budgetone 101
>
>
>
>
> 1.0.5.16 - the latest version.
>
> >>> Michael 'Moose' Dinn <dinn at blend.twistedpair.ca> 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
>
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> _______________________________________________
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>
>
--
-------------------------------------------
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Panama, Republic of Panama
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