Polycom SOUNDPOINT 3804-11530-222 - Manual (Page 43)

Release Notes - SIP Application Changes
Copyright © 2007 Polycom, Inc. Page 37
• 15679:
Ring Type 12 (Ringback-style) sounds incomplete after the first ring
• 15694: Phone crashes and reboots when 'Exit' is pressed from Network
Configuration menu in Korean Language
• 15730: If a menu is displayed when a call is missed on the SoundPoint IP 300
and 301 phones, the missed call count is not updated on the idle display
• 15766: Display is incorrect after selecting name dialing then entering and
exiting a call list while dial tone is playing
• 15781: After putting a local conference on hold then splitting the calls then
joining them, the first call may remain on hold
• 15855: In the Instant Msg menu of the SoundPoint IP 300 and 301 phones,
"x/Ascii" is not displayed after pressing the "1/A/a" softkey
2.13.4 Configuration File Parameter Changes
.cfg
File Action Parameter Description
sip added voIpProt.server.x.expires.overlap The number of seconds before the
expiration time returned by server ‘x’ at
which the phone should try to re-register.
The phone will try to re-register at half the
expiration time returned by the server if that
value is less than the configured overlap
value.
Default = 60. Minimum = 5, maximum =
65535.
sip added voIpProt.SIP.ms-forking Default = 0. Can be 0 or 1.
0 = Support for MS-forking is disabled.
1 = Support for MS-forking is enabled and
the phone will reject all Instant Message
INVITEs. This parameter is relevant for LCS
server installations.
Note that if any endpoint registered to the
same account has MS-forking
disabled, all other endpoints default back to
non-forking mode. Windows Messenger
does not use MS-forking so be aware of this
behavior if one of the endpoints is Windows
Messenger.
sip added voIpProt.SIP.dialog.usePvalue Default = 0. Can be 0 or 1.
0 = Phone uses “pval” field name in Dialog.
This obeys the draft-ietf-sipping-dialog-
package-06.txt draft.
1 = Phone uses a field name of “pvalue”.
sip added voIpProt.SIP.connectionReuse.useAli
as Default = 0. Can be 0 or 1.
0 = old behaviour
1 = Phone uses the connection reuse draft
which introduces "alias".
sip added se.pat.callProg.15.name="secondary
dial"
se.pat.callProg.15.inst.1.type="chord"
se.pat.callProg.15.inst.1.value="1" Same configuration method as primary dial
tone. Allows a different tone to be
configured for secondary dial tone.