Polycom 3804-11530-222 Personal Computer User Manual


 
Release Notes - SIP Application Changes
Page 44 Copyright © 2007 Polycom, Inc.
2.17.2 Removed Features
None.
2.17.3 Corrections
The following issues have been resolved with this release:
11658: Phone continues to append to log file on FTP boot server after that file
has reached its configured size limit
12613: SoundPoint IP600 and 601 phones may establish a call with no audio
after holding, resuming and ending multiple calls
12949: If the phone’s first line is a shared line and cannot obtain dial tone,
pressing the “NewCall” soft key does not activate the first available line
14673: Special characters such as ‘@’, ‘:’ and ‘?’ are not accepted as part of
the FTP or HTTP password
14968: If the phone reboots, the app.log size can increase past the size limit
15002: If the phone’s first line is unregistered, pressing the “NewCall” soft key
does not activate another line
15127: Phone may have one-way audio in a call after multiple transfers have
been done
15218: If multiple contact header fields contain multiple expire values, the
phone does not always pick the lowest non-zero value
15235: Phone will freeze if the SAS-VP server becomes unavailable when the
phone application is starting
15339: ACK lacks the same authorization credentials as the INVITE which is a
failure to comply with RFC 3261
15419: Blind transfer doesn't work for URL calling
15568: A comma in quotes in SIP address headers should be interpreted
correctly
15596: Remote phone can force local conference host to resume call
unexpectedly in specific scenario
15615: When a shared line call is on hold, lifting the handset seizes the last
used line instead of the first available line
14939: Shared line user must press “Answer” soft key twice to answer an
incoming call in some scenarios
15907: After a reboot, a phone may show "1 new missed call" which can't be
cleared until another call is missed
15982: The SDP session identifier should not be changed on each re-INVITE
16021: FTP downloads may fail because incorrect timeouts are used
16141: Phone with a shared line loses hot dialed digits when remote shared
line changes state, such as placing an active call on hold