Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
1Addendum
This addendum addresses changes to the SoundPoint IP / SoundStation IP SIP 2.0
Administrator’s Guide made by the release of the SoundPoint IP 650 phone.
The SoundPoint IP 650 phone behaves in a similar manner to the SoundPoint IP 601
(supports the SoundPoint IP Expansion Module) unless otherwise specified.
For more information, refer to the Release Notes for the SIP Application, Version
2.0.3 B.
Note
The various .hd. parameters in sip.cfg (such as voice.aec.hd.enable, voice.ns.hd.enable, and
voice.agc.hd.enable) are headset parameters. There are not connected to high definition or HD voice.
1.1 Added or Changed Features
1.1.1 Configurable Feature Keys
The SoundPoint IP 650 phone’s default SIP key layouts is the same as the
SoundPoint IP 600 and 601. Refer to 3.1.7 Configurable Feature keys on page 29.
1.1.2 Handset, Headset, and Speakerphone
The SoundPoint IP 650 phones are full-duplex speakerphones.
Changes can be found in the following parameters in the sip.cfg configuration file:
Polycom recommends that you do not change these values.
1.1.3 LCD Backlight
Backlight intensity on the SoundPoint IP 650 phone has three modes:
• Backlight On
• Backlight Idle
• Dim
You can modify the Backlight On intensity and the Backlight Idle intensity separately.
You can select high, medium, low, and off levels for both. Dim mode intensity is
determined by the Backlight On intensity and the Backlight Idle intensity together.
Backlight settings can be found in the User Preferences <up/> parameter in the sip.cfg
configuration file.
Permitted
Attribute
up.backlight.onIntensity0 (off),
up.backlight.idleIntensity0 (off),
Values
1 (low),
2
(medium),
3 (high)
1 (low),
2
(medium),
3 (high)
DefaultInterpretation
3This parameter controls the intensity
of the LCD backlight when it turns
on during normal use of the phone.
1This parameter controls the intensity
of the LCD backlight when the
phone is idle.
Note: If idleIntensity is set higher
than onIntensity, it will be replaced
with the onIntensity value.
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
• RAM Disk <ramdisk/>
Permitted
Attribute
Values
DefaultInterpretation
ramdisk.bytesPerBlock0, 32, 33,
..., 1024
0
These three parameters use internal
defaults when value is set to 0.
• Finder <finder/>
Permitted
Attribute
res.finder.sizeLimitpositive
Values
ger
inte
DefaultInterpretation
300If a resource that is being downloaded to
the phone is larger than this value * 1000
bytes (= the maximum size), the resource
will be automatically truncated to the
maximum size defined.
res.finder.minfree1 to 20481200A resource will not be downloaded to the
on
e if the amount of free memory is
ph
less than this value * 1000 bytes (= the
minimum size). This parameter is used
for 16MB SDRAM platforms and scaled
up for platforms with more SDRAM.
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
1.1.5 MicroBrowser
The SoundPoint IP 650 phones support an XHTML microbrowser. This can be
launched by pressing the Services key.
MicroBrowser parameter changes in the sip.cfg configuration file are as follows:
AttributePermitted ValuesDefaultInterpretation
mb.limits.nodespositive integer256Limits the number of tags which the
XML parser will handle. This limits
the amount of memory used by complicated pages. A maximum total of
500 (256 each) is recommended. This
value is used as referent values for
16MB of SDRAM.
Note: Increasing this value may have
a detrimental effect on performance
of the phone.
mb.limits.cachepositive integer200Limits the total size of objects down-
loaded for each page (both XHTML
and images). Once this limit is
reached, no more images are downloaded until the next page is
requested. Units = kBytes. This value
is used as referent values for 16MB of
SDRAM.
Note: Increasing this value may have
a detrimental effect on performance
of the phone.
offe
ptime negotiations. This is also
the range of supported Rx payload sizes.
milliseconds) that must be
achieved before play out begins
for the first time. Once this depth
has been achieved initially, the
depth may fall below this point
and play out will still continue.
This parameter should be set to
the smallest possible value which
is at least two packet payloads,
and larger than the expected short
term average jitter.
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
Permitted
Attribute
Values
DefaultInterpretation
voice.audioProfile.G722.jitterBufferShrink
voice.audioProfile.G722.jitterBufferMax
10, 20,
30, ...
(multiple
10)
of
> jitterBufferMin,
multiple
,
of 10
<=500
for IP
430, 500,
501, and
600,
<= 160
for IP
0 an
d
30
301
500The absolute minimum duration
time (in milliseconds) of RTP
packet Rx with no packet loss
between jitter buffer size shrinks.
Use smaller values (1000 ms) to
minimize the delay on known
good networks. Use larger values
to minimize packet loss on networks with large jitter (3000 ms).
160The largest jitter buffer depth to
be supported (in mi
lliseconds).
Jitter above this size will always
ca
use lost packets. This parameter
should be set to the smallest possible value that will support the
expected network jitter.
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
2 Addendum
This addendum addresses changes to the SoundPoint IP / SoundStation IP SIP 2.0
Administrator’s Guide made by the release of the SIP 2.1 application.
For more information, refer to the Release Notes for the SIP Application, Version 2.1 .
Note
The various .hd. parameters in sip.cfg (such as voice.aec.hd.enable, voice.ns.hd.enable, and
voice.agc.hd.enable) are headset parameters. They are not connected to high definition or HD voice.
2.1 Added or Changed Features
2.1.1 Digit Map
Enhancements have been made to the local digit maps that can eliminate the need for
using the Dial or Send soft key when making outgoing calls. Refer to the “Technical
Bulletin 11572: Changes to Local Digit Maps on SoundPoint® IP Phones ” at
www.polycom.com/support/voip/ .
2.1.2 Billing Code
Billing codes let administrators assign specific codes to all of their organization’s outgoing calls. The prompt to signal employees to enter their billing codes has changed.
Refer to the “Technical Bulletin 9268: Billing Code Entry on SoundPoint
with Sylantro ” at www.polycom.com/support/voip/ .
2.1.3 Syslog
®
IP phones
Syslog is a de facto standard for forwarding log messages in an IP network. The
SIP application has been enhanced to support logging system level messages and error
conditions with communications networks to a centralized location. Refer to the
“Technical Bulletin 17124: Syslog on SoundPoint
www.polycom.com/support/voip/ .
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
2.1.4 Server Redundancy
Server redundancy enhancements provides backup to other SIP server(s) by providing
basic registration and redirection services. Refer to the “Technical Bulletin 5844: SIP
Server Fallback Enhancements on SoundPoint® IP Phones” at www.polycom.com/
support/voip/ .
2.1.5 MicroBrowser
An XHTML microBrowser is now supported on the SoundPoint IP 430 and 501
phones. The tables shows the platforms where the XHTML microBrowser is supported and where it is not.
:
Supported PlatformsUnsupported Platforms
IP 430IP 300, 301
IP 501IP 500
IP 600, 601, 650IP 4000
This can be launched by pressing the Services key , or through the Menu key by selecting Features, and then Services, if there is no Services key on the phone.
The microBrowser auto-navigates to the first visible, selectable item on the web page
(a hyperlink, for example):
• after initial page load
• after scrolling further down page (after the se
key press)
cond or third down arrow
Note
XHTML tables must be properly formatted (should include <tbody> and </tbody> tags). Improperly
formatted tables could cause the phone to reboot.
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
2.1.6 Disable Message Waiting Indicator by
Registration
The SIP application has been enhanced to allow the message waiting indicator to be
disabled by registration.
• Changes can be found in the following parameters in the phone1.cfg
configuration file:
Permitted
Attribute
Values
DefaultInterpretation
msg.mwi.x.callBackModecontact or
registration
or
disabled
“registration”
Disables message
rieval and disables
ret
waiting message notification for the line.
If set to “contact”, a call
will be placed to
contact specified in the
callback attribute when
the user invokes message retrieval.
If set to “registration”, a
call will
this registration to the
contact registered (the
phone will call itself).
If set to “disabled”,
ssage retrieval is dis-
me
abled.
2.1.7 Daylight Saving Time Changes for 2007
be placed
the
using
Daylight saving time dates will be changing in North America in 2007. Refer to the
“Technical Bulletin 17803: Daylight Savings Time Changes for 2007 on SoundPoint
®
IP Phones” at www.polycom.com/support/voip/ .
Changes to default values and the Interpretations can be found in the following param-
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
Permitted
Attribute
Values
DefaultInterpretation
tcpIpApp.sntp.daylightSavings.fixedDayEnable
tcpIpApp.sntp.daylightSavings.start.month1-123 (March)Month to start DST.
tcpIpApp.sntp.daylightSavings.start.date1-318Day of the month to
0, 10If set to 0, month, date,
and dayOfWeek are
used in DST start date
calculation.
If set to 1, then only
month and date are
used.
Mapping: 1=Jan,
2=Feb, ..., 12=Dec
start DST.
Mapping (on or after): 1
= the first occurrence of
a given day-of-the-week
in a month, 8 = the second occurrence of a
given day-of-the-week
in a month, 15 = the
third occurrence of a
given day-of-the-week
in a month, 22 = the
fourth occurrence of a
given day-of-the-week
in a month
tcpIpApp.sntp.daylightSavings.start.time0-232Time of day to start
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
2.1.9 Miscellaneous Configuration File Changes
2.1.9.1 sip.cfg
The following changes have also occurred in the sip.cfg configuration file:
Permitted
Attribute
voIpProt.SIP.useSendonlyHold0, 11If set to 1, the phone will
Values
DefaultInterpretation
send a reinvite with a
stream mode attribute of
“sendonly” when a call is
put on hold. This is the
same as the previous
behavior.
If set to 0, the phone will
send a reinvite with a
stream mode attribute of
“inactive” when a call is
put on hold.
NOTE: The phone will
ignore the value of this
parameter if set to 1 when
the parameter voIpProt.SIP.useRFC2543hol
d is also set to 1 (default
is 0).
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
Permitted
Attribute
Values
DefaultInterpretation
voIpProt.server.x.transportDNSnaptr or
TCPpreferred or
UDPOnly or
TLS or
TCPOnly
DNSnaptrIf set to Null or
DNSnaptr:
If voIpProt.server.x.address is a
hostname and voIpProt.server.x.port is 0 or
Null, do NAPTR then
SRV look-ups to try to
discover the transport,
ports and servers, as per
RFC 3263. If voIpProt.server.x.address is an
IP address, or a port is
given, then UDP is used.
If set to TCPpreferred:
TCP is the preferred
transport, UDP is used if
TCP fails.
If set to UDPOnly:
Only UDP will be used.
If set to TLS:
If TLS fails, transport
fails. Leave port field
empty (will default to
5061) or set to 5061.
If set to TCPOnly:
Only TCP will be used.
NOTE: TLS is not sup-
ported on SoundPoint IP
300 and 500 phones.
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
Permitted
Attribute
Values
DefaultInterpretation
voIpProt.SIP.outboundProxy.transportDNSnaptr or
TCPpreferred or
UDPOnly or
TLS or
TCPOnly
DNSnaptrIf set to Null or
DNSnaptr:
If voIpProt.SIP.outboundProxy.address is a hostname and
voIpProt.SIP.outboundProxy.port is 0 or Null, do
NAPTR then SRV lookups to try to discover the
transport, ports and servers, as per RFC 3263. If
voIpProt.SIP.outboundProxy.address is an IP
address, or a port is given,
then UDP is used.
If set to TCPpreferred:
TCP is the preferred
transport, UDP is used if
TCP fails.
If set to UDPOnly:
Only UDP will be used.
If set to TLS:
If TLS fails, transport
fails. Leave port field
empty (will default to
5061) or set to 5061.
If set to TCPOnly:
Only TCP will be used.
NOTE: TLS is not sup-
ported on SoundPoint IP
300 and 500 phones.
voice.gain.rx.analog.chassis.IP_6500Gain setting.
voice.handset.sidetone.adjust.IP_430-13Handset sidetone.
call.enableOnNotRegistered0,11If set to 1, calls will be
allowed when the phone
is not successfully registered.
If set to 0, calls will not
be permitted without a
valid registration. If a
user picks up handset,
presses the New Call soft
key, or presses the
speaker phone, speed dial
or the line keys to get a
dial tone, “Service
unavailable” is displayed.
Administrator’s Guide - SoundPoint® IP / SoundStation® IPAddendum
call.stickyAutoLineSeize
is set to 1, this parameter
has no effect. The regular
stickyAutoLineSeize
behavior is followed.
call.stickyAutoLineSeize
If
is set to 0 or Null and this
parameter is set to 1, this
overrides the stickyAutoLineSeize behavior for
hot dial only. (Any New Call scenario seizes the
next available line.)
call.stickyAutoLineSeize
If
is set to 0 or Null and this
parameter is set to 0 or
Null, there is no difference between hot dial and
New Call scenarios.
The following changes has also occurred in the phone1.cfg configuration file:
Permitted
Attribute
reg.x.server.y.transpo rtDNSnaptr or
reg.x.outboundProxy.transportDNSnaptr or
Values
TCPpreferred or
UDPOnly or
TLS or
TCPOnly
TCPpreferred or
UDPOnly or
TLS or
TCPOnly
DefaultInterpretation
DNSnaptrRefer to Interpretation
of voIpProt.server.x.transport
in 2.1.9.1 sip.cfg on
page 6, the previous
section.
If specified, this
attribute may override
the value in sip.cfg.
DNSnaptrRefer to Interpretation
of voIpProt.SIP.outboundProxy.transport in
2.1.9.1 sip.cfg on
page 6, the previous
section.
If specified, this
attribute may override
the value in sip.cfg.
Changes to Local Digit Maps on SoundPoint® IP Phones
This technical bulletin provides detailed information on how to modify the
configuration files to automate the setup phase of number-only calls.
This information applies to SoundPoint IP phones running SIP application
version 2.1 or later.
Introduction
Enhancements have been made to this feature that can eliminate the need for
using the Dial or Send soft key when making outgoing calls. For example, it
can match the behavior of removing the 9 or 0 from a string of dialed digits or
adding the area code before dialed digits when a switch to 10 digit phone
numbers occurs.
As soon as a digit pattern matching the digit map is found, the call setup
process will complete automatically. The configuration syntax is the same as
that specified in 2.1.5 of RFC 3435. The phone’s behavior when the user dials
digits that do not match the digit map is configurable. It is also possible to strip
a trailing ‘#’ from the digits sent, prepend a ‘+’ to digits, or to replace certain
matched digits with the introduction of ‘R’ to the digit map.
Configuration File Changes
If a single dial plan is used for the entire company, the dial plan is best
specified in the application configuration file (sip.cfg). You can also create
multiple dial plans and specify which phones are to use which in the
phone-specific configuration file (phone1.cfg).
Configuration changes can performed centrally at the boot server or locally:
Central
(boot server)
<December, 2006>
3725-17471-001/A
Configuration file:
sip.cfg
Configuration file:
phone1.cfg
Specify impossible match behavior, trailing # behavior,
digit map matching strings, and time out value.
•For more information, refer to Dial Plan in Application
Configuration File on page 2.
Specify per-registration impossible match behavior,
trailing # behavior, digit map matching strings, and time
out values that override those in sip.cfg.
•For more information, refer to Dial Plan in Per-Phone
Configuration File on page 4.
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
LocalWeb Server
(if enabled)
Specify impossible match behavior, trailing # behavior,
digit map matching strings, and time out value.
Navigate to: http://<phoneIPAddress>/appConf.htm#ls
Changes are saved to local flash and backed up to
<Ethernet address>-phone.cfg on the boot server.
Changes will permanently override global settings unless
deleted through the Reset Local Config menu selection.
Dial Plan in Application Configuration File
The
<dialplan/>
•Digit Map <digitmap/> on page 3.
Note
The dial plan is not applied against Placed Call List, VoiceMail, last call return, and
remote control dialed numbers.
This configuration attribute is defined as follows:
Attribute
dialplan.applyToCallListDial0, 10This attribute covers dialing from
attribute is described below and also includes:
Permitted
Values
DefaultInterpretation
Received Call List and Missed Call List
including dialing from Edit or Info submenus.
If set to 0, the dial plan is not applied
against the dialed number.
if set to 1, the dial plan is applied
against the dialed number.
dialplan.applyToDirectoryDial0, 10This attribute covers dialing from
Directory as well as Speed Dial List.
Value interpretation is the same as for
dialplan.applyToCallListDial
Note: An Auto Call Contact number is
considered a dial from directory.
dialplan.applyT oUserDial0, 11This attribute covers the case when the
user presses the Dial soft key to send
dialed number when in idle state
display.
Value interpretation is the same as for
dialplan.applyToCallListDial.
2
.
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Permitted
Attribute
dialplan.applyT oUserSend0, 11This attribute covers the case when the
dialplan.impossibleMatchHandling0, 1 or 20If set to 0, the digits entered up to and
dialplan.removeEndOfDial0, 11If set to 1, strip trailing # digit from digits
ValuesDefaultInterpretation
user presses the Send soft key to send
the dialed number.
Value interpretation is the same as for
dialplan.applyToCallListDial.
including the point where an impossible
match occurred are sent to the server
immediately.
If set to 1, give reorder tone.
If set to 2, allow user to accumulate
digits and dispatch call manually with
the Send soft key.
sent out.
Digit Map <digitmap/>
A digit map is defined either by a “string” or by a list of strings. Each string in
the list is an alternative numbering scheme, specified either as a set of digits or
timers, or as an expression over which the gateway will attempt to find a
shortest possible match.
Digit map extension letter “R” indicates that certain matched strings are
replaced. The following examples shows the semantics of the syntax:
R9RRxxxxxxx
•
—remove 9 at the beginning of the dialed number
— For example, if a customer dials 914539400, the first 9 is removed
when the call is placed.
•
RR604Rxxxxxxx
—prepend 604 to all 7 digit numbers
— For example, if a customer dials 4539400, 604 is added to the front of
the number, so a call 6044539400 is placed.
•
R9R604Rxxxxxxx
•
R999R911R
•
xxR601R600Rxx
•
xR60xR600Rxxxxxxx
—replaces 9 with 604
—convert 999 to 911
—when applied on 1160122 gives 1160022
—any 60x will be replaced with 600 in the middle of
the dialed number that matches
— For example, if a customer dials 16092345678, a call is placed to
16002345678.
3
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
The following guidelines should be noted:
•You must use only *, #, or 0-9 between second and third R
•If a digit map does not comply, it is not included in the digit plan as a valid
one. That is, no matching is done against it.
•There is no limitation on the number of R triplet sets in a digit map.
However, a digitmap that contains less than full number of triplet sets (for
example, a total of 2Rs or 5Rs) is considered an invalid digit map.
•Using T in the left part of RRR syntax is not recommended. For example,
R0TR322R should be avoided.
This configuration attribute is defined as follows:
AttributePermitted ValuesDefaultInterpretation
dialplan.digitmapstring compatible with
the digit map feature of
MGCP described in
2.1.5 of RFC 3435.
String is limited to 768
characters and 30
segments; a comma is
also allowed; when
reached in the digit
map, a comma will
turn dial tone back
on;’+’ is allowed as a
valid digit; extension
letter ‘R’ is used as
defined above.
3 | 3 | 3 | 3 | 3 | 3 | 3Timeout in seconds for each
When this attribute is present,
number-only dialing during the
setup phase of new calls will be
compared against the patterns
therein and if a match is found, the
call will be initiated automatically
eliminating the need to press Send.
control the use of match and
replace in the dialed number in the
different scenarios. Refer to page 2.
segment of digitmap.
Note: If there are more digit maps
than timeout values, the default
value of 3 will be used. If there are
more timeout values than digit
maps, the extra timeout values are
ignored.
,
,
, and
Dial Plan in Per-Phone Configuration File
Per-registration dial plan configuration is supported.
The
<dialplan/>
•Digit Map <digitmap/> on page 3.
attribute is described below and also includes:
4
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
In the following tables, x is the registration number. IP 300, 301, and 430: x=1-2;
IP 500 and 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1
Permitted
Attribute
dialplan.x.applyToCallListDial0, 10When present, and if
dialplan.x.applyToDirectoryDial0, 10When present, and if
Values
DefaultInterpretation
dialplan.x.digitmap
Null, this attribute overrides
the global dial plan defined in
the sip.cfg configuration file.
For interpretation, refer to
Dial Plan in Application
Configuration File on page 2.
dialplan.x.digitmap
Null, this attribute overrides
the global dial plan defined in
the sip.cfg configuration file.
For interpretation, refer to
Dial Plan in Application
Configuration File on page 2.
is not
is not
dialplan.x.applyToUserDial0, 11When present, and if
dialplan.x.digitmap
Null, this attribute overrides
the global dial plan defined in
the sip.cfg configuration file.
For interpretation, refer to
Dial Plan in Application
Configuration File on page 2.
dialplan.x.applyToUserSend0, 11When present, and if
dialplan.x.digitmap
Null, this attribute overrides
the global dial plan defined in
the sip.cfg configuration file.
For interpretation, refer to
Dial Plan in Application
Configuration File on page 2.
is not
is not
5
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Permitted
Attribute
ValuesDefaultInterpretation
dialplan.x.impossibleMatchHandling0, 1 or 20When present, and if
dialplan.x.digitmap
Null, this attribute overrides
the global dial plan defined in
the sip.cfg configuration file.
For interpretation, refer to
Dial Plan in Application
Configuration File on page 2.
dialplan.x.removeEndOfDial0, 11When present, and if
dialplan.x.digitmap
Null, this attribute overrides
the global dial plan defined in
the sip.cfg configuration file.
For interpretation, refer to
Dial Plan in Application
Configuration File on page 2.
is not
is not
Digit Map <digitmap/>
The digit map syntax is the same as for the application configuration file (refer
to Digit Map <digitmap/> on page 3).
This configuration attribute is defined as follows:
Permitted
Attribute
dialplan.x.digitmapA string compatible with
ValuesDefaultInterpretation
the digit map feature of
MGCP described in
2.1.5 of RFC 3435;
string is limited to 768
characaters and 30
segments; a comma is
also allowed; when
reached in the digit map,
a comma will turn dial
tone back on;’+’ is
allowed as a valid digit;
extension letter ‘R’ is
used as defined above.
NullWhen present, this attribute
overrides the global dial
plan defined in the sip.cfg
configuration file.
For more information, refer
to Digit Map <digitmap/> on
page 3.
6
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Attribute
Permitted
ValuesDefaultInterpretation
dialplan.x.digitmap.timeOutstring of positive integers
separated by ‘|’
NullWhen present, and if
dialplan.x.digitmap
not Null, this attribute
overrides the global dial
plan defined in the sip.cfg
configuration file.
For more information, refer
to Digit Map <digitmap/> on
page 3.
is
Trademark Information
Polycom®, SoundPoint®, and the Polycom logo design are registered trademarks of Polycom, Inc. in the U.S. and
various countries. All other trademarks are the property of their respective companies.
7
Technical Bulletin 9268
Billing Code Entry on SoundPoint® IP Phones with Sylantro
This technical bulletin provides detailed information on how the SIP
application has been modified for billing code entry when managed by a
Sylantro call server.
This information applies to SoundPoint IP phones running SIP application
version 2.1 or later.
Introduction
Note
Billing codes let administrators assign specific codes to all of their
organization’s outgoing calls.
When a SoundPoint IP phone managed by a Sylantro call server is configured
to require billing codes, calls are not connected until the a valid billing code is
entered.
The modified user interface on a SoundPoint IP phone running SIP 2.1 is
described in the following section, Billing Code Entry.
Billing Code Entry
This section describes the steps the user must perform to enter a billing code.
To enter a billing code when placing a call:
1. Do one of the following to a place a call:
This feature is only supported on Sylantro call servers.
aWith the handset on-hook, enter the long-distance number (including
prefix).
<December, 2006>
3725-17483-001/A
You may need to press the Dial soft key to indicate you are finished
entering the number.
bPick up the handset and enter the long-distance number (including
prefix).
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
You may need to press the Send soft key to indicate you are finished
entering the number.
The cursor pauses after the last digit has been entered. The call is not
placed at this time.
A secondary dial tone is played and the text “Enter more digits” appears
on the display just above the soft keys.
2. Enter the billing code.
If the billing code is accepted, the call is placed at this time.
If the billing code is not accepted, you will hear a fast busy tone and the
call is not placed.
Trademark Information
Polycom®, SoundPoint®, and the Polycom logo design are registered trademarks of Polycom, Inc. in the U.S. and
various countries. All other trademarks are the property of their respective companies.
2
Technical Bulletin 17124
Syslog on SoundPoint® IP Phones
This technical bulletin provides detailed information on how the SIP
application has been modified to support logging system level messages and
error conditions with communications networks to a centralized location.
This information applies to SoundPoint IP phones running SIP application
version 2.1 or later.
Introduction
Syslog is a de facto standard for forwarding log messages in an IP network.
The term "syslog" is often used for both the actual syslog protocol, as well as
the application or library sending syslog messages.
The syslog protocol is a very simplistic protocol: the syslog sender sends a
small textual message (less than 1024 bytes) to the syslog receiver. The receiver
is commonly called "syslogd", "syslog daemon" or "syslog server". Syslog
messages can be sent through UDP or TCP. The data is sent in cleartext.
Syslog is supported by a wide variety of devices and receivers. Because of this,
syslog can be used to integrate log data from many different types of systems
into a central repository.
The syslog protocol is defined in RFC 3164. For more information on syslog,
go to http://www.ietf.org/rfc.html .
You must enter the administrative password to access this menu. The
default value is “456”.
3. Using the Down Arrow key and the Select soft key, scroll down to Syslog
Menu.
Syslog Menu
The following syslog configuration parameters can be modified on the Syslog
menu:
NamePossible ValuesDescription
Server Addressdotted-decimal IP address
OR
domain name string
Server TypeNone=0,
UDP=1,
TCP=2
Facility0 to 23A description of what generated the log message.
Render Level1 to 6Specifies the lowest class of event that will be
Prepend MAC
Address
Enabled, DisabledIf enabled, the phone’s MAC address is prepended
The syslog server IP address or host name.
The default value is NULL.
The protocol that the phone will use to write to the
syslog server.
If set to “None”, transmission is turned off, but the
server address is preserved.
For more information, refer to section 4.1.1 of RFC
3165.
The default value is 16, which maps to “local 0”.
rendered to syslog. It is based on
log.render.level
Refer to Basic Logging <level/><change/> and
<render/> on page 138of the
Administrator’s Guide
to the log message sent to the syslog server.
and can be a lower value.
SIP 2.0
.
2
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Flash Parameter Configuration
The global
device
parameter has been modified to include the following:
NamePossible ValuesDescription
device.syslog.serverNamedotted-decimal IP address
OR
domain name string
device.syslog.transportNone=0,
UDP=1,
TCP=2
device.syslog.facility0 to 23A description of what generated the log message.
device.syslog.renderLevel1 to 6Specifies the lowest class of event that will be
device.syslog.prependMacEnabled, DisabledIf enabled, the phone’s MAC address is prepended
The syslog server IP address or host name.
The default value is NULL.
The protocol that the phone will use to write to the
syslog server.
If set to “None”, transmission is turned off, but the
server address is preserved.
For more information, refer to section 4.1.1 of RFC
3165.
The default value is 16, which maps to “local 0”.
rendered to syslog. It is based on
log.render.level
Refer to Basic Logging <level/><change/> and
<render/> on page 138of the SIP 2.0
Administrator’s Guide
to the log message sent to the syslog server.
and can be a lower value.
.
Note
The parameters for this feature should be put in separate configuration files to
simplify maintenance. Do not add them to existing configuration files (such as
sip.cfg). Create a new configuration file for parameters that should apply to all
phones.
Polycom recommends that you test the new configuration files on two phones
before initializing all phones.
Trademark Information
Polycom®, SoundPoint®, and the Polycom logo design are registered trademarks of Polycom, Inc. in the U.S. and
various countries. All other trademarks are the property of their respective companies.
3
Technical Bulletin 5844
SIP Server Fallback Enhancements on SoundPoint® IP Phones
This technical bulletin provides detailed information on how the SIP
application has been enhanced to support SIP server fallback.
This information applies to SoundPoint IP phones running SIP application
version 2.1 or later.
Introduction
Server redundancy is often required in VoIP deployments to ensure continuity
of phone service for events where the call server needs to be taken offline for
maintenance, the server fails, or the connection from the phone to the server
fails.
Two types of redundancy are possible:
•Fail-over: In this mode, the full phone system functionality is preserved by
having a second equivalent capability call server take over from the one
that has gone down/off-line. This mode of operation should be done
using DNS mechanisms or “IP Address Moving” from the primary to the
back-up server.
Note
Warning
•Fallback: In this mode, a second less featured call server (router or
gateway device) with SIP capability takes over call control to provide basic
calling capability, but without some of the richer features offered by the
primary call server (for example, shared lines, presence, and Message
Waiting Indicator). Polycom phones support configuration of multiple
servers per SIP registration for this purpose.
In some cases, a combination of the two may be deployed.
In SIP 2.1, the fallback behavior has been enhanced and this behavior is
described in this document.
Your SIP server provider should be consulted for recommended methods of
configuring phones and servers for fail-over configuration.
The server redundancy behavior in SIP2.1 has changed from that implemented in
prior releases. Prior to SIP 2.1, the
4.6.2.1 of the SIP 2.0 Administrator's Guide) could be used for fail-over
configuration. The older behavior is no longer supported. Customers that are using
reg.x.server.y
the
ensure that their current deployments are not adversely affected. For example the
phone will only support advanced SIP features such as shared lines, missed calls,
presence with the primary server (y=1).
. configuration parameters where y>=2 should take care to
reg.x.server.y
parameters (see section
<January, 2007>
3725-17472-001/A
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Terminology
Before you read this document, it is important to understand certain
terminology and become familiar with the server/registration configuration
as described in the references listed in the References on page8. The behavior
described in this document supersedes that described in section 3.6.5 of the
SIP 2.0 Administrator's Guide.
SIP Registrations: SoundPoint IP phones support the ability to have multiple
SIP Registrations per phone. This is often used to support multiple “Lines” on
a single phone and normally the SIP server(s) used for each Registration are
the same. However, they could be different.
Primary and Fallback Servers: Each of these SIP Registrations may be
configured for concurrent registration with multiple servers for fallback
purposes. For example, a phone may be configured to have two SIP
Registrations and each SIP Registration may be configured with two separate
servers (a primary server and a fallback server). DNS mechanisms (as
described in RFC3263) may be used such that the servers are capable of
resolving to multiple physical SIP servers for fail-over purposes.
Note
The primary server is the only one that will be used for advanced SIP features such
as shared lines, message waiting indicators, and presence. This is a change in
behavior from software releases before SIP 2.1 All other configured servers are
referred to as fallback servers.
Working Server: The phone maintains a list of all possible servers gained from
both DNS and configuration. The highest priority server which has an active
registration is treated as the working server and will be the first server tried for
call initiation purposes. At any time, there is only one working server
recognized by the phone.
Registrar Server: Servers (both primary and fallback) may be configured with
registration enabled or disabled using the
reg.x.server.y.register
configuration parameter. Servers that have this parameter enabled will
attempt registrations and are termed a registrar server. If a server is not a
registrar server, calls will be attempted on that server if appropriate, but
registration will not be attempted. Only a registrar server can become the
working server.
For the purposes of this document, we will use examples where the phone has
only one SIP Registration.
The sections Server <server/> on page 95 and Registration <reg/> on page 149
of the SIP 2.0 Administrator's Guide describe the parameters that are relevant to
the configuration of the phones for server redundancy and fallback behavior.
Configuration file changes for SIP 2.1 are described in Configuration File
Changes on page 7.
2
Technical Bulletin
`
`
`
`
PSTN Gateway
Internet
PSTN
Hosted VoIP Service
Provider
VoIP SMB Customer
Premise
DNS Server
Call Server 1A
Call Server 1B
SIP Capable Router
Server2
®
SoundPoint
IP, SIP 2.1
SIP 2.1 Server Fallback Implementation
In the SIP 2.1 release, the redundancy behavior of Polycom SoundPoint IP and
SoundStation IP phones has been changed and improved by adding the ability
for a single SIP Registration (Line) to be registered to more than one server
concurrently. In previous releases, the phone would only maintain one active
server registration per SIP Registration (Line). The concurrent server
registration capability adds an ability to do a faster and more efficient
hand-over to an independent call server both for incoming as well as outgoing
calls.
To assist in explaining the redundancy behavior, an illustrative example of
how a system may be deployed is defined in the following section.
Example Deployment
A small medium business (SMB) customer uses a hosted IP-Centrex service
from a Service provider. The Service provider has two redundant call servers
at their network operations center (NOC) and uses a DNS server to resolve the
IP addresses of these servers. The SMB customer also has an on-premise router
which has the ability to handle SIP call traffic and has a connection to an
on-site PSTN gateway. This gateway is intended to be used in conditions in
which the Internet connection to the service provider is not working.
3
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Phone Configuration
The phones at the customer site are configured as follows:
•Server 1 (the primary server) will be configured with the address of the
service provider call server. The IP address of the server(s) to be used will
be provided by the DNS server. For example:
•Server 2 (the fallback server) will be configured to the address of the
router/gateway that provides the fallback telephony support and is
on-site. For example:
reg.1.server.2.address=172.23.0.1
Note
It is possible to configure the phone for more than two servers per registration, but
you need to exercise caution when doing this to ensure that the phone and network
load generated by registration refresh of multiple registrations do not become
excessive. This would be of particularly concern if a phone had multiple
registrations with multiple servers per registration and it is expected that some of
these servers will be unavailable.
Phone Operation for Registration
After the phone has booted up, it will register to all the servers that are
configured.
Server 1 is the primary server and supports greater SIP functionality than any
of servers. For example, SUBSCRIBE/NOTIFY services (used for features such
as shared lines, presence, and BLF) will only be established with Server 1.
Upon registration timer expiry of each server registration, the phone will
attempt to re-register. If this is unsuccessful, normal SIP re-registration
behavior (typically at intervals of 30 to 60 seconds) will proceed and continue
until the registration is successful (for example, when the Internet link is once
again operational). While the primary server registration is unavailable, the
next highest priority server in the list will serve as the working server. As soon
as the primary server registration succeeds, it will return to being the working
server.
Note
If
reg.x.server.y.register
However, the INVITE will fail over to that server if all higher priority servers are
down.
is set to 0, then phone will not register to that server.
4
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Behavior When the Primary Server Connection Fails
For Outgoing Calls (INVITE Fallback)
When the user initiates a call, the phone will go through the following steps to
connect the call:
1. Try to make the call using the working server.
2. If the working server does not respond correctly to the INVITE, then try
and make a call using the next server in the list (even if there is no current
registration with these servers). This could be the case if the Internet
connection has gone down, but the registration to the working server has
not yet expired.
3. If the second server is also unavailable, the phone will try all possible
servers (even those not currently registered) until it either succeeds in
making a call or exhausts the list at which point the call will fail.
At the start of a call, server availability is determined by SIP signaling failure.
SIP signaling failure depends on the SIP protocol being used as described
below.
Warning
•If TCP is used, then the signaling fails if the connection fails or the Send
fails.
•If UDP is used, then the signaling fails if ICMP is detected or if the signal
times out. If the signaling has been attempted with all servers in the list
and this is the last server then the signaling fails after the complete UDP
timeout defined in RFC 3261. If it is not the last server in the list, the
maximum number of retries using the configurable retry timeout is used.
For more information, refer to Server <server/> on page 95 and
Registration <reg/> on page 149 of the SIP 2.0 Administrator's Guide.
If DNS is used to resolve the address for Servers, the DNS server is unavailable,
and the TTL for the DNS records has expired, the phone will attempt to contact the
DNS server to resolve the address of all servers in its list before initiating a call.
These attempts will timeout, but the timeout mechanism can cause long delays (for
example, two minutes) before the phone call proceeds “using the working server”.
To mitigate this issue, long TTLs should be used. It is strongly recommended that
an on-site DNS server is deployed as part of the redundancy solution.
For Incoming Calls (Incoming Call Fallback)
The primary call server can use mechanisms for detecting that the Internet
connection is down and route incoming calls through the PSTN link to the
back-up gateway/router on-site. Since the phone is simultaneously registered
to both servers, it will receive calls through the gateway even if the primary
registration has not expired. This is a key advantage of the new behavior
introduced in SIP 2.1.
5
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Changes From Previous Phone Behavior (Releases Before SIP 2.1)
Before SIP 2.1In SIP 2.1
A Line is only capable of maintaining
one server registration.
If two servers are configured (for
example,
"server1"
reg.1.server.2.address =
"server2"
register with Server1 as the working
server. Phone calls will be placed and
received through Server1 only. If the
registration to Server1 fails or expires,
then the phone will attempt to register
with Server2. If this registration
succeeds, then incoming calls will be
received using this server. At this point,
Server2 takes over as the working
server.
The phone will continually attempt
registration using SIP registration
protocols with Server1. At the point that
this succeeds, the registration with
Server2 will expire and Server1 will
resume as the working server.
The phone attempts to maintain full SIP
functionality with each server, but it is
questionable how effective this is.
reg.1.server.1.address =
and
, the phone will initially
A Line will maintain registrations with
all servers that are configured as
registrar servers.
If two servers are configured (for
example,
"server1"
reg.1.server.2.address =
"server2"
both Server1 as the working and
Server2. Phone calls will be placed
through Server1, but may be received
through either Server1 or Server2. If
the registration to Server1 fails or
expires, then the Server2 will become
the working server.
The phone will continually attempt to
register with Server1 and, when this is
successful, will switch back to using
Server1 as the working server. The
Server2 registration will be maintained.
Only basic SIP registration for INVITE
functions is maintained with servers
other than the primary server.
reg.1.server.1.address =
and
, the phone will register with
Recommended Practices for Server Fallback Deployments
The best method for ensuring optimum server redundancy is to deploy two
identical call servers and use either DNS methods or “IP Address Moving”
together with call server recommended practices for maintaining
synchronization of records between the redundant servers. This is termed
fail-over (refer to Terminology on page 2). Deployment varies dependent on
the SIP call server being used. Consult your SIP call server supplier for
recommended practices for fail-over configuration.
In situations where server redundancy for fall-back purpose is used, the
following measures should be taken to optimize the effectiveness of the
solution:
1. Deploy an on-site DNS server to avoid long call initiation delays that can
result if the DNS server records expire.
6
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
2. Do not use OutBoundProxy configurations on the phone if the
OutBoundProxy could be unreachable when the fallback occurs.
SoundPoint IP phones can only be configured with one OutBoundProxy
per registration and all traffic for that registration will be routed through
this proxy for all servers attached to that registration. If Server 2 is not
accessible through the configured proxy, call signaling with Server 2 will
fail.
3. Avoid using too many servers as part of the redundancy configuration as
each registration will generate more traffic.
4. Educate users as to the features that will not be available when in
“fallback” operating mode.
Configuration File Changes
Configuration changes can performed centrally at the boot server:
Central
(boot server)
Configuration file: sip.cfgSpecify global primary and fallback server configuration
parameters.
•For more information, refer to Protocol in Application
Configuration File on page 7.
Configuration file:
phone1.cfg
Specify per registration primary and fallback server configuration
parameters values that override those in sip.cfg.
•For more information, refer to Registration in Per-Phone
Configuration File on page 8.
Protocol in Application Configuration File
The
<voIpProt/>
•Server <server/> on page 7.
Server <server/>
This configuration attribute now includes:
AttributePermitted ValuesDefaultInterpretation
voIpProt.server.x.lcs0, 10This attribute overrides the
attribute includes:
voIpProt.SIP.lcs
If set to 1, the proprietary “epid”
parameter is added to the From field
of all requests to support Microsoft
Live Communications Server.
.
7
Technical Bulletin
®
SoundPoint
IP, SIP 2.1
Registration in Per-Phone Configuration File
Per-registration configuration is supported.
The
<registration/>
Permitted
Attribute
reg.x.server.y.lcs0, 10This attribute overrides the
Values
attribute now includes:
DefaultInterpretation
reg.x.lcs .
If set to 1, the Microsoft Live
Communications Server is
supported for registration x.
References
1. SIP 2.0 Administrator’s Guide for the SoundPoint IP and SoundStation IP
Polycom®, SoundPoint®, and the Polycom logo design are registered trademarks of Polycom, Inc. in the U.S. and
various countries. All other trademarks are the property of their respective companies.
8
Loading...
+ hidden pages
You need points to download manuals.
1 point = 1 manual.
You can buy points or you can get point for every manual you upload.