THOMSON
ST2030
SIP
SG
1.59.3
Release
Notes
Mar 07 2008
Contents
1
Release package description
2
Main changes implemented in SG.080227.1.59.3 (wrt SG 1.58.6)
3
Relevant known limitations in ST2030 SG.080227.1.59.3
1 Release package description
ST2030 SIP – SG.080227.1.59.3
contains the following elements:
/Binary
·
Application binary file : v2030SG.080227.1.59.3.zz
·
Audio configuration file: TelConf2030SG_v1.59.3.txt
/Documents
·
Release notes (this document): ST2030S SG.080227.1.59.3 Release
Notes.html
/APS & fw upgrade:
·
Upgrade instructions: ST2030 SIP SW upgrade via Web GUI v8.txt
·
ST2030-ST2022 AutoProvisioning-V0030.pdf
·
ST20XXS_Config File Syntax_V0026.pdf
/New Features
·
ST20XX SIP New Features SG vx.59.doc
The documents provided
with this release are a complement to the Admin Guide, which you can find
at http://www.thomsontelecompartner.com
2
Main changes implemented in SG.080227.1.59.3
(wrt SG 1.58.6)
Functionality added:
2.1-
Capability
of downloading and updating Tone and Language tables.
See “ST20XX SIP New Features SG vx.59.doc “.
2.2-
New Tracing tool through telnet interface: it allows
getting SIP level traces remotely. See “ST20XX SIP New Features SG
vx.59.doc “.
2.3-
SIP INFO
messages allowed for DTMF in early dialogs.
2.4-
New Subscribe mechanism for
Supervision + Call-PickUp: Up to now, each time a pick-up operation was
performed, phone always sent a new SUBSCRIBE to know dialog status of phone
line which was picked up, both when using “Pickup” softkey and using a blinking
supervising FK. Now, if phone line that will be picked up is being
supervised in a FK, ST2030 uses
status data provided by NOTIFYs received from monitoring lines. Otherwise, if
no phone line configured on FK, pick up procedure will be performed as it is
already known, phone has to send a new SUBSCRIBE to get the state of the
extension before sending an INVITE.
2.5-
New Reset to
default behavior. The effect is: erase the configuration files and
re-launch the APS process. As a result, the phone will reload all the files
filled in the *.inf file, even if the filenames haven’t been modified
(including Fw and telconf).
Fixes provided:
2.7-
Sylantro
SIP-B: Centralized conference is now working properly.
2.8-
Problem
with attended transfer in Asterisk server is now OK. Example: external line
gets picked up by phone 1. Phone 1 transfers to phone 2 with an attended
transfer. Phone 2 then talks to the external party and tries to transfer back
to phone 1 with an attended transfer. There is no TRANSFER button on phone 2 if
that phone enters the extension of phone 1.
2.9-
Click-to-Dial
feature is now working with some PBX (Asterisk). The phone didn’t allow an
incoming call where the From header was itself: From=(To=Request-URI).
ST2030 is currently supporting it.
2.10-
SIP
Stack behavior:
o
Request
with Cseq field with an overlarge
value (rfc 4475 Page 22) are now rejected.
o
Request
with Start Line and CSeq Method Mismatch (RFC4475 Page 33) are now rejected.
o
200 OK
wrongly accepted when branch parameter not matching request. Now are discarded.
o
200 OK
wrongly accepted when CSeq is lower than in request. Now are discarded.
2.11-
Upgrade
the phone using the web interface and the FTP method is now OK. Path size in
the web interface was increased up to 63 chars. UserID/Password up to 31.
2.12-
To
export the current configuration you can use the command “tftp2 listparms”
2.13-
Phone
does not crash if "TO"
length is up to 485.
2.14-
Fix
provided for the problem:Impossible to auto answer after 2 calls, due to a
conflict between Auto-Answer and Automatic-HangUp features.
2.15-
The retransmission
of the INVITE is now the same as the original one (Session-Expires now appears
and SDP is not increased by 1).
2.16-
There
is no problem if sw_notify_autoprovision
flag is activated, although the rest of necessary params for that purpose are empty.
2.17-
Notify
dialog-info and call pick-up:
Phone supports the two allowed NOTIFY formats for dialog event (the
older format
“draft-ietf-sipping-service-examples” and the newer format “RFC 4235” –
recommended).
2.18-
Record-Route:
if ST2030 is the caller and receives a Re-INV with RR headers, now it copies
the RR headers in the answer.
2.19-
If 2
Remote-Party-ID headers are received in the same request, the incoming call is
now accepted.
2.20-
Strange
Tone during Transfer is now ok (if we dial any number very quick after pressing
Transf softkey but before phone displays Enter a Number, and then we press Back
(because the first number dialed does not appear on "Enter number"
screen), phone reproduces a high tone continuously - fixed).
2.21-
If
multiple DHCP-Servers are running, then the phone takes the first of the IP
offers.
2.22-
Sometimes,
the ST2030 phone stops refreshing its BLF subscriptions - fixed.
2.23-
Backup
SIP server will take correctly its configured local port, instead of the
Primary SIP server local port , which was being used in previous releases.
2.24-
Fixed
Call Conference when Multi-extension is configured (GenBand). St2030 has an
extension configured in Feature Key 2 (FK2). Then it makes a call from main
line and after that initiates a Conference. The (consultative) call to the 3rd
party is made from its own line instead of with From and Contact headers of the
extension.
3
Relevant known limitations in ST2030
SG.080227.1.59.3
3.1-
If you
are going to upgrade the FW
through the Web GUI, please make sure to: restart the phone after the application upgrade, wait
until you are back in the home page again, and then upload the Telephone
Configure and restart again. See ST2030 SIP SW upgrade via Web GUI v8.txt as
reference.
3.2-
Sylantro
SIP-B: in ACD mode, return to “On line” state is not always automatic after a
call has been finalized.
3.3-
Sylantro
SIP-B: in BLA mode, not all supervised extension states are signaled with the
corresponding backlit key.
3.4-
Call
Park with Sylantro. If we
go on-hook after introduce the park extension + ok (so display shows
"parking ..") but before the phone sends "REFER" (because
it takes around 5 secs to send it), phone crashes.
3.5-
Phone
acting as switch doesn't filter traffic properly towards PC.
3.6-
Login-Logout
substitution mode: When the phone tries to register to the dynamic account,
receives a Notify which is a "pushout" of the static account. The
phone is considering it as a "pushout" of the dynamic account and
falls back.
-EOF-