Polycom Video Game Sound System Version 203B User Manual

SIP 2.0 Administrator’s Guide

SoundPoint /SoundStation IP SIP

Version 2.0.3B Addendum

Version 2.1 Addendum

January 2007

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Addendum

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:

• Gains <gain/>

Attribute

Default

voice.handset.rxag.adjust.IP_650

voice.handset.txag.adjust.IP_650

voice.handset.sidetone.adjust.IP_650

-3

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Addendum

Attribute

Default

voice.headset.rxag.adjust.IP_650

voice.headset.txag.adjust.IP_650

voice.headset.sidetone.adjust.IP_650

-3

Important

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

Values

Attribute

Default

Interpretation

up.backlight.onIntensity

0 (off),

1 (low),

This parameter controls the intensity

of the LCD backlight when it turns

on during normal use of the phone.

(medium),

3 (high)

up.backlight.idleIntensity

0 (off),

1 (low),

This parameter controls the intensity

of the LCD backlight when the

phone is idle.

(medium),

3 (high)

Note: If idleIntensity is set higher

than onIntensity, it will be replaced

with the onIntensity value.

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Addendum

1.1.4 Expanded Memory and Expanded Flash Memory

Changes can be found in the following parameters in the sip.cfg configuration file:

• Directory <dir/>

Permitted

Values

Attribute

Default

Interpretation

dir.local.volatile.maxSize

1 to 100

100

Maximum size in Kbytes of

volatile storage that the

directory will be permitted

to consume.

dir.local.volatile.8meg

0, 1

Attribute applies only to

platforms with 8 Mbytes of

flash memory.

If set to 1, use volatile stor-

age for phone-resident copy

of the directory to allow for

larger size.

dir.local.nonVolatile.max-

Size.8meg

1 to 100

100

Attribute applies only to

platforms with 8 Mbytes of

flash memory.

This is the maximum size of

non-volatile storage that the

directory will be permitted

to consume.

• Provisioning <prov/>

Permitted

Values

Attribute

Default

Interpretation

prov.fileSystem.rfs0.minFreeSpace

5-512

Minimum free space in

Kbytes to reserve in the

file system when down-

loading files from the boot

server.

prov.fileSystem.ffs0.8meg.min-

FreeSpace

512

Note: Polycom recom-

mends that you do not

change these parameters.

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Addendum

• RAM Disk <ramdisk/>

Permitted

Values

Attribute

Default

Interpretation

ramdisk.bytesPerBlock 0, 32, 33,

..., 1024

These three parameters use internal

defaults when value is set to 0.

• Finder <finder/>

Permitted

Values

Attribute

Default

Interpretation

res.finder.sizeLimit

positive

integer

300

If 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.minfree

1 to 2048

1200

A resource will not be downloaded to the

phone if the amount of free memory is

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.

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Addendum

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:

Attribute

Permitted Values

Default

Interpretation

mb.limits.nodes

positive integer

256

Limits the number of tags which the

XML parser will handle. This limits

the amount of memory used by com-

plicated 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.cache

positive integer

200

Limits the total size of objects down-

loaded for each page (both XHTML

and images). Once this limit is

reached, no more images are down-

loaded 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.

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Addendum

1.1.6 G.722 Audio Codec

The SoundPoint IP 650 supports the G.722 audio codec.

Changes can be found in the following parameters in the sip.cfg configuration file:

• Codec Preferences <codecPref/>

Permitted

Values

Attribute

Default Interpretation

voice.codecPref.IP_650.G711M Null, 1-3

Specifies the codec preferences

for the SoundPoint IP 650

platform.

1 = highest

3 = lowest

voice.codecPref.IP_650.G711A

voice.codecPref.IP_650.G729A

Null = do not use

voice.codecPref.IP_650.G722

• Audio Profiles <audioProfile/>

Permitted

Values

Attribute

Default Interpretation

voice.audioProfile.G722.pay-

loadSize

10, 20,

30, ... 80

Preferred Tx payload size in mil-

liseconds to be provided in SDP

offers and used in the absence of

ptime negotiations. This is also

the range of supported Rx pay-

load sizes.

voice.audioProfile.G722.jitter-

BufferMin

20, 40,

The smallest jitter buffer depth (in

milliseconds) that must be

50, 60, ...

(multiple

of 10)

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.

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Addendum

Permitted

Attribute

Values

Default Interpretation

500 The absolute minimum duration

voice.audioProfile.G722.jitter-

BufferShrink

10, 20,

30, ...

time (in milliseconds) of RTP

(multiple

of 10)

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 net-

works with large jitter (3000 ms).

voice.audioProfile.G722.jitter-

BufferMax

> jitter-

Buffer-

Min,

multiple

of 10,

160

The largest jitter buffer depth to

be supported (in milliseconds).

Jitter above this size will always

cause lost packets. This parameter

should be set to the smallest pos-

sible value that will support the

expected network jitter.

<=500

for IP

430, 500,

501, and

600,

<= 160

for IP

300 and

301

• Gains <gain/>

Attribute

Default

voice.gain.rx.analog.chassis.IP_650

voice.gain.rx.analog.ringer.IP_650

voice.gain.rx.digital.chassis.IP_650

voice.gain.rx.digital.ringer.IP_650

voice.gain.tx.analog.chassis.IP_650

voice.gain.tx.digital.chassis.IP_650

-9

-21

Important

Polycom recommends that you do not change these values.

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Addendum

• Receive <rxEq/>

Attribute

Default

voice.rxEq.hs.IP_650.preFilter.enable

voice.rxEq.hs.IP_650.postFilter.enable

voice.rxEq.hd.IP_650.preFilter.enable

voice.rxEq.hd.IP_650.postFilter.enable

voice.rxEq.hf.IP_650.preFilter.enable

voice.rxEq.hf.IP_650.postFilter.enable

Important

Polycom recommends that you do not change these values.

• Transmit <txEq/>.

Attribute

Default

voice.txEq.hs.IP_650.preFilter.enable

voice.txEq.hs.IP_650.postFilter.enable

voice.txEq.hd.IP_650.preFilter.enable

voice.txEq.hd.IP_650.postFilter.enable

voice.txEq.hf.IP_650.preFilter.enable

voice.txEq.hf.IP_650.postFilter.enable

Important

Polycom recommends that you do not change these values.

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Addendum

1.1.7 USB Diagnostics

The SoundPoint IP 650 phone has a USB port, which will be supported by future

releases of the SIP application.

USB port parameters can be found in the USB <usb/> parameter in the sip.cfg config-

uration file.

Attribute

Permitted Values

Default

Interpretation

usb.enable

0, 1

This parameter enables or disables the

USB port on the phone.

usb.bulkDrive.enab 0, 1

This parameter enables or disables

support for a USB bulk drive con-

nected to the USB port on the phone.

usb.bulkDrive.nam alphanumeric

usb-

Drive

This parameter is a string which spec-

ifies the name of the mounted USB

drive.

string

Other changes to support a USB port can be found in the following parameter in the

sip.cfg configuration file:

• Basic Logging <log/>

Permitted

Values

Attribute

Default

Interpretation

log.level.change.usb

0-5

Control the logging detail

level for the usb compo-

nent. These are the input fil-

ters into the internal

memory-based log system.

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Addendum

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Addendum

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 out-

going calls. The prompt to signal employees to enter their billing codes has changed.

Refer to the “Technical Bulletin 9268: Billing Code Entry on SoundPoint IP phones

with Sylantro ” at www.polycom.com/support/voip/ .

2.1.3 Syslog

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 IP Phones” at

www.polycom.com/support/voip/ .

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Addendum

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 sup-

ported and where it is not.

Supported Platforms

IP 430

Unsupported Platforms

IP 300, 301

IP 500

IP 501

IP 600, 601, 650

IP 4000

This can be launched by pressing the Services key, or through the Menu key by select-

ing 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 second or third down arrow

key press)

Note

XHTML tables must be properly formatted (should include <tbody> and </tbody> tags). Improperly

formatted tables could cause the phone to reboot.

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Addendum

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

Values

Attribute

Default

Interpretation

msg.mwi.x.callBackMode

contact or

registration

“registra- Disables message

tion”

retrieval and disables

waiting message notifi-

cation for the line.

disabled

If set to “contact”, a call

will be placed to the

contact specified in the

callback attribute when

the user invokes mes-

sage retrieval.

If set to “registration”, a

call will be placed using

this registration to the

contact registered (the

phone will call itself).

If set to “disabled”,

message retrieval is dis-

abled.

2.1.7 Daylight Saving Time Changes for 2007

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-

eters in the sip.cfg configuration file:

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Addendum

Permitted

Values

Attribute

Default

Interpretation

tcpIpApp.sntp.daylightSavings.fixedDay-

Enable

0, 1

If 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.

tcpIpApp.sntp.daylightSavings.start.month

tcpIpApp.sntp.daylightSavings.start.date

1-12

1-31

3 (March) Month to start DST.

Mapping: 1=Jan,

2=Feb, ..., 12=Dec

Day of the month to

start DST.

Mapping (on or after): 1

= the first occurrence of

a given day-of-the-week

in a month, 8 = the sec-

ond 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.time

0-23

Time of day to start

DST, in 24 hour clock.

Mapping: 2=2 am, 14=2

tcpIpApp.sntp.daylightSavings.start.dayOf- 1-7

Week

Day of week to apply

DST.

Mapping: 1=Sun,

2=Mon, ..., 7=Sat

tcpIpApp.sntp.daylightSavings.start.dayOf- 0, 1

Week.lastInMonth

If set to 1 and fixedDay-

Enable is set to 0, DST

starts on the last day

(specified by start.day-

OfWeek) of the week in

the month. The

start.date is ignored.

tcpIpApp.sntp.daylightSavings.stop.month

1-12

Month to stop DST.

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Addendum

Permitted

Values

Attribute

Default

Interpretation

tcpIpApp.sntp.daylightSavings.stop.date

1-31

Day of the month to

stop DST.

tcpIpApp.sntp.daylightSavings.stop.time

0-23

Time of day to stop

DST in 24 hour clock.

tcpIpApp.sntp.daylightSavings.stop.dayOf- 1-7

Week

Day of week to stop

DST.

tcpIpApp.sntp.daylightSavings.stop.dayOf- 0, 1

Week.lastInMonth

If set to 1 and fixedDay-

Enable=0, stop DST on

the last day of the week

(specified by stop.day-

OfWeek) in the month.

The stop.date is

ignored.

2.1.8 Configurable Feature Keys

It has been determined that only some feature keys can be disabled. The exact feature

keys that can be “null”-ified are platform-dependent.

Platform

Key IDs

IP 300, 301

IP 430

5, 7, 16, 23, 29, 31, 32

7, 8, 9, 10, 29, 31, 33, 34

7, 8, 9, 10, 29, 30, 31, 32, 36, 37

7, 8, 9, 30, 32, 36, 37, 40

1, 2, 5, 7, 16, 29

IP 500, 501

IP 600, 601, 650

IP 4000

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Addendum

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

Values

Attribute

Default

Interpretation

voIpProt.SIP.useSendonlyHold

0, 1

If set to 1, the phone will

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 voIp-

Prot.SIP.useRFC2543hol

d is also set to 1 (default

is 0).

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Addendum

Permitted

Values

Attribute

Default

Interpretation

voIpProt.server.x.transport

DNSnaptr or DNSnapt

If set to Null or

DNSnaptr:

If voIp-

TCPpre-

ferred or

UDPOnly or

TLS or

Prot.server.x.address is a

hostname and voIp-

Prot.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 voIp-

TCPOnly

Prot.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.

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Addendum

Permitted

Values

Attribute

Default

Interpretation

voIpProt.SIP.outboundProxy.transport

DNSnaptr or DNSnapt

If set to Null or

DNSnaptr:

If voIpProt.SIP.outbound-

Proxy.address is a host-

name and

TCPpre-

ferred or

UDPOnly or

TLS or

TCPOnly

voIpProt.SIP.outbound-

Proxy.port is 0 or Null, do

NAPTR then SRV look-

ups to try to discover the

transport, ports and serv-

ers, as per RFC 3263. If

voIpProt.SIP.outbound-

Proxy.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_650

voice.handset.sidetone.adjust.IP_430

call.enableOnNotRegistered

Gain setting.

-13

Handset sidetone.

0,1

If set to 1, calls will be

allowed when the phone

is not successfully regis-

tered.

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.

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Addendum

Permitted

Values

Attribute

Default

Interpretation

call.stickyAutoLineSeize.onHookDialing

Null, 0, 1

Null

If call.stickyAutoLineSeize

is set to 1, this parameter

has no effect. The regular

stickyAutoLineSeize

behavior is followed.

If call.stickyAutoLineSeize

is set to 0 or Null and this

parameter is set to 1, this

overrides the stickyAuto-

LineSeize behavior for

hot dial only. (Any New

Call scenario seizes the

next available line.)

If call.stickyAutoLineSeize

is set to 0 or Null and this

parameter is set to 0 or

Null, there is no differ-

ence between hot dial and

New Call scenarios.

2.1.9.2 phone1.cfg

The following changes has also occurred in the phone1.cfg configuration file:

Permitted

Values

Attribute

Default

Interpretation

reg.x.server.y.transport

DNSnaptr or DNSnapt

Refer to Interpretation

of voIp-

Prot.server.x.transport

in 2.1.9.1 sip.cfg on

page 6, the previous

section.

TCPpre-

ferred or

UDPOnly or

TLS or

TCPOnly

If specified, this

attribute may override

the value in sip.cfg.

reg.x.outboundProxy.transport

DNSnaptr or DNSnapt

Refer to Interpretation

of voIpProt.SIP.out-

boundProxy.transport in

2.1.9.1 sip.cfg on

page 6, the previous

section.

TCPpre-

ferred or

UDPOnly or

TLS or

TCPOnly

If specified, this

attribute may override

the value in sip.cfg.

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Addendum

2.1.9.3 device Parameter

The following changes has also occurred in the device parameter:

Permitted

Values

Attribute

Default

Interpretation

device.prov.redunAttemptLimit

10, Null

Refer to the File Trans-

mit Tries parameter in

2.2.1.3.3 Server Menu

on page 11 of the SIP

2.0 Administrator’s

Guide.

device.prov.redunInterAttemptDelay

device.em.power

300, Null

300

Refer to the Retry Wait

parameter in 2.2.1.3.3

Server Menu on page 11

of the SIP 2.0 Adminis-

trator’s Guide.

Enabled,

Disabled

Null

Refer to the EM Power

parameter in 2.2.1.3.1

Main Menu on page 8

of the SIP 2.0 Adminis-

trator’s Guide.

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Technical Bulletin 11572

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

Configuration file:

sip.cfg

Specify impossible match behavior, trailing # behavior,

digit map matching strings, and time out value.

(boot server)

•

For more information, refer to Dial Plan in Application

Configuration File on page 2.

Configuration file:

phone1.cfg

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.

<December, 2006>

3725-17471-001/A

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Technical Bulletin

SoundPoint ^®IP, SIP 2.1

Local

Web 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/> attribute is described below and also includes:

•

Digit Map <digitmap/> on page 3.

The dial plan is not applied against Placed Call List, VoiceMail, last call return, and

remote control dialed numbers.

Note

This configuration attribute is defined as follows:

Permitted

Attribute

Values

Default Interpretation

dialplan.applyToCallListDial

0, 1

This attribute covers dialing from

Received Call List and Missed Call List

including dialing from Edit or Info sub-

menus.

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.applyToDirectoryDial

dialplan.applyToUserDial

0, 1

This 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.

0, 1

This 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.

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SoundPoint ^®IP, SIP 2.1

Permitted

Values

Attribute

dialplan.applyToUserSend

Default Interpretation

0, 1

This attribute covers the case when the

user presses the Send soft key to send

the dialed number.

Value interpretation is the same as for

dialplan.applyToCallListDial.

dialplan.impossibleMatchHandlin 0, 1 or 2

If set to 0, the digits entered up to and

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.

dialplan.removeEndOfDial

0, 1

If set to 1, strip trailing # digit from digits

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—replaces 9 with 604

R999R911R—convert 999 to 911

xxR601R600Rxx—when applied on 1160122 gives 1160022

xR60xR600Rxxxxxxx—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.

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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:

Attribute

Permitted Values

Default

Interpretation

dialplan.digitmap

string 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.

[2-9]11|0T|

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.

+011xxx.T|

0[2-9]xxxxxxxxx|

+1[2-9]xxxxxxxx|

[2-9]xxxxxxxxx|

[2-9]xxxT

Attributes

dialplan.applyToCallListDial

dialplan.applyToDirectoryDial

dialplan.applyToUserDial, and

dialplan.applyToUserSend

control the use of match and

replace in the dialed number in the

different scenarios. Refer to page 2.

dialplan.digitmap.timeOut

string of positive

integers separated by

‘|’

3 | 3 | 3 | 3 | 3 | 3 | 3 Timeout in seconds for each

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.

Dial Plan in Per-Phone Configuration File

Per-registration dial plan configuration is supported.

The <dialplan/> attribute is described below and also includes:

•

Digit Map <digitmap/> on page 3.

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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

Values

Default

Interpretation

dialplan.x.applyToCallListDial

dialplan.x.applyToDirectoryDial

dialplan.x.applyToUserDial

dialplan.x.applyToUserSend

0, 1

When present, and if

dialplan.x.digitmapis not

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.

0, 1

When present, and if

dialplan.x.digitmapis not

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.

When present, and if

dialplan.x.digitmapis not

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.

When present, and if

dialplan.x.digitmapis not

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.

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SoundPoint ^®IP, SIP 2.1

Permitted

Values

Attribute

dialplan.x.impossibleMatchHandling

Default

Interpretation

0, 1 or 2

When present, and if

dialplan.x.digitmapis not

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.removeEndOfDial

0, 1

When present, and if

dialplan.x.digitmapis not

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.

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

Values

Default

Interpretation

dialplan.x.digitmap

A string compatible with

the digit map feature of

MGCP described in

2.1.5 of RFC 3435;

Null

When present, this attribute

overrides the global dial

plan defined in the sip.cfg

configuration file.

string is limited to 768

characaters and 30

segments; a comma is

also allowed; when

For more information, refer

to Digit Map <digitmap/> on

page 3.

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.

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Technical Bulletin

SoundPoint ^®IP, SIP 2.1

Permitted

Values

Attribute

Default

Interpretation

dialplan.x.digitmap.timeOut

string of positive integers

separated by ‘|’

Null

When present, and if

dialplan.x.digitmapis

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.

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.

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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

This feature is only supported on Sylantro call servers.

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:

With the handset on-hook, enter the long-distance number (including

prefix).

You may need to press the Dial soft key to indicate you are finished

entering the number.

Pick up the handset and enter the long-distance number (including

prefix).

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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.

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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 .

The log.render.level maps to syslog severity as follows:

0 -> SeverityDebug (7)

1 -> SeverityDebug (7)

2 -> SeverityInformational (6)

3 -> SeverityInformational (6)

4 -> SeverityError (3)

5 -> SeverityCritical (2)

6 -> SeverityEmergency (0)

7 -> SeverityNotice (5)

For more information on log.render.level, refer to Basic Logging

<level/><change/> and <render/> on page 138 of the SIP 2.1 Administrator’s

Guide.

Network configuration changes required to support this feature are described

in the following section, Network Configuration Changes.

<December, 2006>

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Technical Bulletin

SoundPoint ^®IP, SIP 2.1

Network Configuration Changes

The Network Configuration menu on the SoundPoint IP phone running SIP

2.1 has been modified to include:

•

Syslog Menu

To access the Syslog menu:

1. From the idle display on a SoundPoint IP phone, press the Menu key.

2. Using the Down Arrow key and the Select soft key, select Settings >

Advanced > Admin Settings > Network Configuration.

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:

Name

Possible Values

Description

Server Address

dotted-decimal IP address

domain name string

The syslog server IP address or host name.

The default value is NULL.

Server Type

Facility

None=0,

UDP=1,

TCP=2

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.

0 to 23

1 to 6

A description of what generated the log message.

For more information, refer to section 4.1.1 of RFC

3165.

The default value is 16, which maps to “local 0”.

Render Level

Specifies the lowest class of event that will be

rendered to syslog. It is based on

log.render.level and can be a lower value.

Refer to Basic Logging <level/><change/> and

<render/> on page 138 of the SIP 2.0

Administrator’s Guide.

Prepend MAC

Address

Enabled, Disabled

If enabled, the phone’s MAC address is prepended

to the log message sent to the syslog server.

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Technical Bulletin

SoundPoint ^®IP, SIP 2.1

Flash Parameter Configuration

The global deviceparameter has been modified to include the following:

Name

Possible Values

Description

device.syslog.serverName

dotted-decimal IP address

domain name string

The syslog server IP address or host name.

The default value is NULL.

device.syslog.transport

device.syslog.facility

None=0,

UDP=1,

TCP=2

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.

0 to 23

1 to 6

A description of what generated the log message.

For more information, refer to section 4.1.1 of RFC

3165.

The default value is 16, which maps to “local 0”.

device.syslog.renderLevel

Specifies the lowest class of event that will be

rendered to syslog. It is based on

log.render.level and can be a lower value.

Refer to Basic Logging <level/><change/> and

<render/> on page 138 of the SIP 2.0

Administrator’s Guide.

device.syslog.prependMac

Enabled, Disabled

If enabled, the phone’s MAC address is prepended

to the log message sent to the syslog server.

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.

Note

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.

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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.

•

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.

Note

The server redundancy behavior in SIP2.1 has changed from that implemented in

prior releases. Prior to SIP 2.1, the reg.x.server.y parameters (see section

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

the reg.x.server.y. configuration parameters where y>=2 should take care to

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).

Warning

<January, 2007>

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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 page 8. 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.

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.

Note

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.

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Technical Bulletin

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.

Hosted VoIP Service

Provider

Call Server 1B

Call Server 1A

Internet

DNS Server

VoIP SMB Customer

Premise

SIP Capable Router

Server2

PSTN

PSTN Gateway

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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:

reg.1.server.1.address="voipserver.serviceprovider.com"

•

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

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.

Note

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.

If reg.x.server.y.registeris set to 0, then phone will not register to that server.

However, the INVITE will fail over to that server if all higher priority servers are

down.

Note

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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.

•

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.

Warning

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.

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Technical Bulletin

SoundPoint ^®IP, SIP 2.1

Changes From Previous Phone Behavior (Releases Before SIP 2.1)

Before SIP 2.1

In SIP 2.1

A Line is only capable of maintaining

one server registration.

A Line will maintain registrations with

all servers that are configured as

registrar servers.

If two servers are configured (for

example, reg.1.server.1.address =

"server1"and

If two servers are configured (for

example, reg.1.server.1.address =

"server1"and

reg.1.server.2.address =

"server2", the phone will initially

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.

reg.1.server.2.address =

"server2", the phone will register with

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

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 will continually attempt to

successful, will switch back to using

Server1 as the working server. The

Server2 registration will be maintained.

The phone attempts to maintain full SIP

functionality with each server, but it is

questionable how effective this is.

Only basic SIP registration for INVITE

functions is maintained with servers

other than the primary server.

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.

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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

Configuration file: sip.cfg

Specify global primary and fallback server configuration

parameters.

(boot server)

•

For more information, refer to Protocol in Application

Configuration File on page 7.

Configuration file:

Specify per registration primary and fallback server configuration

phone1.cfg

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/> attribute includes:

•

Server <server/> on page 7.

Server <server/>

This configuration attribute now includes:

Attribute

Permitted Values

Default

Interpretation

voIpProt.server.x.lcs

0, 1

This attribute overrides the

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.

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Technical Bulletin

SoundPoint ^®IP, SIP 2.1

Registration in Per-Phone Configuration File

Per-registration configuration is supported.

The <registration/> attribute now includes:

Permitted

Attribute

Values

Default

Interpretation

reg.x.server.y.lcs

0, 1

This attribute overrides the

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

Phones, August 2006. Go to

http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,

1687,6314,00.pdf

2. RFC3263 - Locating SIP Servers. Go to

http://www.ietf.org/rfc/rfc3263.txt?number=3263

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.

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