Polycom SoundStation IP 4000, IP 5000, IP 6000, IP 7000

Transcription

T P P
A P P
N O T E
TPP-10011
Date: April, 2010
Product: Polycom® SoundStation® IP 4000,
System version: ShoreTel 9.2
5000, 6000, and 7000
Abstract
Clear conference room conversations are critical for business communications. Full Duplex conference
phones running on the IP network allow successful conferencing while leveraging the power and cost
effectiveness of IP communications. This application note provides the details on adding the Polycom
SoundStation IP 4000, 5000, 6000, and 7000 SIP conference phones to the ShoreTel® IP Phone system.
Table of Contents
Overview............................................................... 1
Features and Benefits............................................ 2
Polycom Overview and Contact........................... 2
Polycom Product Information ............................... 3
Polycom SoundStation IP 7000 .................................3
Polycom Soundstation IP5000 ..................................4
Architecture Overview ......................................... 6
Requirements, Certification and Limitations ........ 6
Polycom IP Phone Requirements .............................6
Version Support .................................................... 7
Certification Testing Results Summary ................ 7
Table 1: Basic Test Cases ........................................7
Table 2: Extended Feature Test Cases ...................8
Configuration Overview ....................................... 9
ShoreTel Configuration ........................................ 9
ShoreTel System Settings - General ........................9
Call Control Settings ..............................................10
Figure 2 – Administration Call Control/Options....10
Figure 3 – Call Control/Options Screen ................11
Switch Settings .......................................................11
Figure 4 – Administration/Switches .......................12
Figure 5 – Edit Switches .........................................13
Figure 6 – ShoreGear Switch Built-in Capacity......14
Site Settings ............................................................14
Figure 7 – Administration/Sites .............................14
Figure 8 – Site Screen SIP Proxies..........................15
Creating SIP Extension ..........................................16
Figure 9 – Trunk Groups Settings ..........................16
Figure 10 – Adding/Editing Users .........................17
Figure 11 – Individual User SIP Settings ................18
SIP Profiles..............................................................18
Figure 12 – SIP Profiles...........................................19
Figure 13 – Edit SIP Profile .....................................20
Polycom SoundStation IP Configuration ............ 21
SoundStation IP Phone Settings – Phone User
Interface .................................................................21
SoundStation IP Phone Settings – Web User Interface
................................................................................22
Polycom Troubleshooting .................................. 24
Polycom Technical Support................................ 24
Document and Software Copyrights .................. 25
Trademarks......................................................... 25
Disclaimer ........................................................... 25
Company Information......................................... 25
Overview
This document provides the details on the Polycom SoundStation IP 4000, 5000, 6000, and 7000 Full
Duplex IP conference phones and describes how to integrate this conference room phone into the
ShoreTel IP Phone system. The document focuses on the configuration procedures needed to set up the
Polycom IP Conference phones for the ShoreTel system and the configuration needed on the ShoreTel
system to support the Polycom SoundStation IP conference phones.
Features and Benefits
Quality conference room phones provide clear, full duplex, hands-free communications for all
participants within a conference room as well as for the connected parties. Conference room phones on
the ShoreTel IP phone system take advantage of this effective communications path while reaping the
benefits of the power and cost effectiveness, through reduced costs of operation and maintenance, of
ShoreTel’s VoIP system.
Polycom Overview and Contact
Polycom, Inc. (NASDAQ: PLCM) is the global leader in telepresence, video, and voice solutions and
a visionary in communications that empower people to connect and collaborate everywhere.
Companies choose Polycom solutions because they allow geographically dispersed workforces to
communicate more effectively and productively over distances. Using Polycom voice, video, and
telepresence products and services, people connect and collaborate with one another from their
desktops, meeting rooms, class rooms, and mobile settings rather than travelling to one place to solve
problems.
Information regarding the Polycom SoundStation IP 4000, 5000, 6000, and 7000 conference phones
can be found through the following contact information:
Polycom Headquarters
4750 Willow Road
Pleasanton, CA 94588-2708
USA
Phone:
1.800.POLYCOM
(in North America)
or 1.925.924.6000
Fax: 1.925.924.6100
Sales:
Lori.Schimpf, Alliance Manager
408-425-7245
[email protected]
Reseller Info:
Resellers who would like to start selling this solution should contact us using contact information at:
[email protected]
960 Stewart Drive Sunnyvale, CA 94085 USA Phone +1.408.331.3300 +1.877.80SHORE Fax +1.408.331.3333 www.ShoreTel.com
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Polycom Product Information
Polycom SoundStation IP 7000
Product Image
Product Description
The SoundStation IP 7000 is a breakthrough conference phone that delivers outstanding
performance and a robust feature set for SIP-based VoIP platforms. It is the most advanced
conference phone ever developed, and is ideal for executive offices, conference rooms, and board
rooms.
The SoundStation IP 7000 features Polycom HD Voice technology, boosting productivity and
reducing listener fatigue by turning ordinary conference calls into crystal-clear interactive
conversations. It delivers high-fidelity audio from 160 Hz to 22 kHz, capturing both the deeper
lows and higher frequencies of the human voice for conference calls that sound as natural as
being there.
Datasheet
(http://www.polycom.com/global/documents/support/sales_marketing/products/voice/plcm_ss_ip7000.
pdf)
North America Product Code: 2230-40300-001
Approximate List price: $1449 US MSR
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Product Image
Product Description
The SoundStation IP 6000 is an advanced IP conference phone that delivers superior performance
for small to midsize conference rooms. With advanced features, full SIP interoperability and
remarkable voice quality, the SoundStation IP 6000 is a price/performance breakthrough for SIPenabled IP environments.
The SoundStation IP 6000 features Polycom HD Voice technology, boosting productivity and
reducing listener fatigue by turning ordinary conference calls into crystal-clear interactive
conversations. It delivers high-fidelity audio from 220 Hz to 14 kHz, capturing both the deeper
lows and higher frequencies of the human voice for conference calls that sound as natural as
being there
Datasheet
(http://www.polycom.com/global/documents/support/sales_marketing/products/voice/plcm_ss_ip
6000.pdf)
Polycom Soundstation IP5000
Product Image
The Polycom® SoundStation® IP 5000 conference phone delivers remarkably clear conference
calls for small conference rooms and executive offices. It features Polycom HD Voice™
technology, broad SIP interoperability, and a modern design that is ideal for smaller rooms—all at
an affordable price.
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The SoundStation IP 5000 conference phone delivers advanced audio performance that is
designed for executive offices and smaller conference rooms with up to 6 participants. From fullduplex technology that eliminates distracting drop-outs to the latest echo cancellation
advancements, only Polycom can deliver a conference phone experience with no compromises.
Data Sheet
http://www.polycom.com/global/documents/products/voice/datasheets/soundstationip5000-datasheet.pdf
North America Product Code:
2200-30900-025, IP5000 – POE Only
2200-43240-001 – AC Power Kit Accessory
Approximate List price: $599 US MSRP
Product Image
Product Description
With superlative audio quality the SoundStation IP 4000 voice conferencing unit is the answer for
organizations that are ready for the benefits and versatility of a SIP enabled business. Designed for
offices or small to medium sized conferences rooms, the SoundStation IP 4000 provides complete
room coverage. You can speak naturally from up to 10 feet away from a microphone and be heard
clearly on the far end of the call. The optional extension microphones offer increased pickup for larger
rooms. With gated microphone technology, echo and background noise is almost entirely eliminated.
The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu driven
user interface, viewable on a high resolution backlit LCD, offers convenient access to business
telephony functions you use most including transfer, hold, redial, and conference.
The SoundStation IP 4000 is currently being replaced by the SoundStation IP 6000, although Polycom
will continue to support the SoundStation IP 4000.
Datasheet
(http://www.polycom.com/global/documents/support/sales_marketing/products/voice/soundstation_ip4
000_datasheet.pdf)
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Architecture Overview
The following is a diagram of the solution architecture showing the integration between the Polycom
SoundStation IP conference phones and the ShoreTel IP System:
Figure 1 – Testing Environment
SIP devices are being implemented on the ShoreTel System with SIP Extensions.
Requirements, Certification and Limitations
The following requirements are necessary to integrate the Polycom SoundStation IP conference phones
to the ShoreTel IP Phone system as described in this Application Note.
Polycom IP Phone Requirements
The Polycom SoundStation IP Conference Phones – The devices should be running the latest
firmware (see version support table below).
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Version Support
SoundStation
IP 4000
SIP 3.1.2
ShoreTel
Release
SoundStation SoundStation
IP 5000
IP 6000
SoundStation
IP 7000
SIP 3.1.2
SIP 3.1.2
SIP 3.1.2
6.1

7.0

7.5

8.0




8.1




9.2




Certification Testing Results Summary
Table 1: Basic Test Cases
ID
Name
Description
1.1
Verify successful startup and initialization of the device
up to a READY/IDLE state using a static IP address
Verify successful re-initialization of device after power
loss while device is idle
Verify successful startup and initialization of the device
up to a READY/IDLE state using DHCP
Verify successful re-initialization of device after power
loss while device is idle
Verify the ability to set Diffserv Code Point from SIP
DUT (device under test)
Verify setting of Date and Time Update on SIP DUT
1.7
Device initialization with
static IP address
Device reset – idle (for
static configurations)
Device initialization with
DHCP
Device reset – idle (for
dynamic configurations)
Verify Diffserv Code
Point support
Verify Date and Time
Update support
Place call
1.8
Receive call
1.9
CODEC support (DUT to
ShoreTel Phone)
1.2
1.3
1.4
1.5
1.6
Verify successful call placement with normal dialing to a
variety of terminating phones
Verify successful call placement with normal dialing to a
variety of terminating phones
Verify successful call connection and audio path using all
supported CODECs (G.711-Ulaw and G.729)
Results
Pass
Pass
Pass
Pass
Not Tested
Pass
Pass
Pass
Pass
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ID
Name
Description
Results
1.10
CODEC support (DUT to
SIP reference)
CODEC negotiation
Verify successful call connection and audio path using all
supported CODECs (G.711-Ulaw and G.729)
Verify successful negotiation between devices configured
with different default CODECs (G.711-Ulaw and G.729)
Verify successful hold and resume of connected call
Pass
Verify successful hold and resume of connected call
Verify successful forwarding of incoming calls
Verify successful forwarding of incoming calls
Verify successful transmission of in-band and out-ofband digits (RFC2833) for calls placed to and from the
DUT with a variety of other devices
Pass
Pass
Pass
Pass
RFC2833
only
1.11
1.12
1.13
1.14
1.15
1.16
Hold DUT to SIP
reference
Hold DUT to ShoreTel
Forward
Forward from SIP DUT
Dual-tone multifrequency (DTMF)
transmission
Pass
Pass
Table 2: Extended Feature Test Cases
ID
Name
Description
Notes
2.1
Call waiting
Pass
2.2
2.3
2.4
2.5
2.6
Park
Transfer – blind
Transfer – monitored
Conference – ad hoc
Caller ID
Verify appropriate notification and successful connection
of incoming call while busy with another party
Verify successful park and retrieval of connected call
Verify successful blind transfer of connected call
Verify successful monitored transfer of connected call
Verify successful ad hoc conference of three parties
Verify that Caller ID name and number is sent and
received from SIP endpoint device
Verify dialing “911” on DUT could connect with “911”
services
Verify that calls are properly terminated on the ShoreTel
Auto Attendant menu and that you can transfer to the
desired extension.
desired extension using the “Dial by Name” feature.
Voice Mail Login Extension.
Not Tested
2.7
911
2.8
Auto Attendant Menu
2.9
Auto Attendant Menu
“Dial by Name”
2.10
Auto Attendant Menu
checking Voice Mail
mailbox
Pass
Pass
Pass
Pass
Pass,
displays
Caller
Name only
on LCD, but
it does have
the Caller
ID in
history.
Pass
Pass
Pass
-8-
ID
Name
Description
2.11
Initiate call to a Hunt
Group
2.12
Initiate call to a
Workgroup
2.13
Hunt Group Member
2.14
Workgroup Agent
2.15
Call Forward – “FindMe”
2.16
ShoreTel Converged
Conferencing Server
2.17
Bridged Call Appearance
(BCA) extension
Initiate a call from DUT and verify that calls route to the
proper Hunt Group and are answered by an available
hunt group member with audio in both directions using
G.729 and G.711 codecs.
Initiate a call from DUT and verify that calls route to the
proper Workgroup and are answered successfully by an
available workgroup agent with audio in both directions
using G.729 and G.711 codecs.
Verify that incoming calls to a hunt group can be
answered properly when DUT is a member of the hunt
group.
Verify that incoming calls to a workgroup can be
answered properly when DUT is an agent of the
workgroup.
Verify that calls are forwarded to DUT’s “FindMe”
destination.
Verify that calls are properly forwarded to the ShoreTel
Converged Conferencing Server and it properly accepts
the access code and you’re able to participate in the
conference bridge.
Verify that calls are properly presented to all of the
phones that have BCA configured and that the call can be
answered, placed on-hold and then transferred.
Notes
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Configuration Overview
The following steps are required to configure the Polycom SoundStation IP 4000, 5000, 6000, and
7000 conference phones to work with the ShoreTel system.
ShoreTel Configuration
This section describes the ShoreTel system configuration to support the Polycom SoundStation IP
phones. The section is divided into general system settings and individual user configurations needed to
support the SoundStation IP phones.
ShoreTel System Settings - General
The first settings to address within the ShoreTel system are the general system settings. These
configurations include the call control, the switch, and the site settings. If these items have already been
configured on the system, skip this section and go on to the “ShoreTel System Settings – Individual
Users” section below.
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Call Control Settings
The Call Control Options within ShoreWare® Director may need to be reconfigured. To configure
these settings for the ShoreTel system, log into ShoreWare Director and select Administration, Call
Control, and then Options (Figure 2).
Figure 2 – Administration Call Control/Options
The “Call Control/Options” screen will then appear (Figure 3).
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Figure 3 – Call Control/Options Screen
If this is an upgrade from previous ShoreTel versions, you may see a parameter named
“Always Use Port 5004 for RTP.” If so, you will need to disable this parameter by unchecking the box and saving the setting. When enabled, SIP extension configuration will fail.
It is also important to note that this “one time” setting requires a system restart (all servers first,
then ShoreGear switches followed by IP Phones) to take effect. Once the server has been
restarted, this configuration parameter will no longer be visible, or may be grayed out. The
default for new installations is disabled, thus the parameter is not visible (as shown in Figure
3).
Realm: The realm is used in authenticating all SIP devices. It is typically a description of the
computer or system being accessed. Changing this value will require reboot of switches
serving as SIP extensions. It is not necessary to modify this parameter to get the Polycom
solution functional.
SIP session interval: Session interval value indicates the session (call) “keep alive” period.
There is no need to modify the default value of 3600 seconds.
SIP session refresher: The refresher setting decides if user agent client or user agent server
refreshes the session. Again, there is no need to modify the default value of “Caller (UAC).”
This allows Polycom to be in control of the session timer refresh.
Switch Settings
When allocating Ports for SIP extensions, these changes are modified by selecting “Administration,”
then “Switches” in ShoreWare Director (Figure 4).
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Figure 4 – Administration/Switches
This action brings up the “Switches” screen. From the “Switches” screen, simply select the name of the
switch to configure. The “Edit ShoreGear …Switch” screen will be displayed. Within the “Edit
ShoreGear …Switch” screen, define one of the “Port Type” settings from the available ports to “100
SIP Proxy” (Figure 5), then save the change.
Note: If your installation requires more than 100 SIP extensions, configure the “Port Type” as
“100 SIP Proxy” as necessary (i.e. two ports configured for “100 SIP Proxy” will provide 200 SIP
extensions).
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Figure 5 – Edit Switches
If the ShoreGear switch that you have selected has “built-in” capacity (i.e., ShoreGear 50/90/220T1/E1,
etc.) for IP phones and SIP trunks, you can also remove 5 ports from the total number available to
provide the “100 SIP Proxy” configuration necessary (Figure 6).
Note: Every 5 ports you remove from the total available will result in “100 SIP Proxy” ports
being made available.
One dedicated ShoreGear 120 switch can act as a proxy for the entire site and support up to 2400 SIP
phones.
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Figure 6 – ShoreGear Switch Built-in Capacity
Site Settings
The next settings to address are the administration of sites. These settings are modified under the
ShoreWare Director by selecting “Administration” then “Sites” (Figure 7).
Figure 7 – Administration/Sites
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This selection brings up the “Sites” screen. Within the “Sites” screen, select the name of the site to
configure. The “Edit Site” screen will then appear. Scroll down to the “SIP Proxy” parameters (Figure
8).
Figure 8 – Site Screen SIP Proxies
The “Virtual IP Address” parameter is a new configuration parameter with ShoreTel 8. This “Virtual IP
Address” is an IP address that can be moved to a different switch during a failure. For each site that
supports SIP extensions, one “Virtual IP Address” is defined that will act as the SIP Proxy for the site.
This IP address must be unique and static.
The ShoreTel server will assign this “Virtual IP Address” to the ShoreGear that is configured as SIP
proxy for the site. Two ShoreGear switches can be configured as SIP proxy servers for redundancy and
reliability purposes. If the primary proxy server goes down, the other proxy switch will take over the
“Virtual IP Address.” Due to this “Virtual IP Address” mechanism, SIP phones will not know if the
proxy switch goes off-line.
Note: If you choose not to define a “Virtual IP Address,” you can only define one proxy switch, and
there will be no redundancy or failover capabilities. The switches available in the “Proxy Switch 1 / 2”
will only be shown if proxy resources have been enabled on the switch.
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The Admission Control Bandwidth defines the bandwidth available to and from the site. This is
important as SIP endpoints may be counted against the site bandwidth. See the ShoreTel Planning and
Installation Guide for more information about this.
ShoreTel 8 now adds 10 CODECs by default. These CODECs can be grouped as “Codec Lists” and
defined in the sites page for “Inter-site” and “Intra-site” calls. See the ShoreTel 8 Server Release Notes
for more information. The default settings will work properly with Comdasys.
Creating SIP Extension
You need to create a user extension for a handset on Comdasys. This is accomplished from ShoreWare
Director by selecting “Administration” followed by “Users…,” then “Individual Users.” This action
will bring up the “Individual Users” screen at the top of the page. To the right of “Add new user at
site:,” select the site you wish to create the user in (from the drop down menu), and select “Go” (Figure
9).
Figure 9 – Trunk Groups Settings
This action brings up the “Users” “Edit Users” screen (Figure 10).
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Figure 10 – Adding/Editing Users
Define the “First Name” and “Last Name” as you deem appropriate. ShoreWare Director will auto-assign the next
available “Number” (i.e., extension), but you can modify it to any available extension. Define the “License Type”
as needed; in this example we chose “Extension and Mailbox” although it’s not necessary to have a mailbox.
Define the proper “User Group” and set the “Home Port” to “Any IP Phone.”
Note: If you configured the “License Type” for “Extension-Only,” you cannot select “Any IP Phone”
but instead must set the “Home Port” for the “SoftSwitch” selection. Save your changes, then scroll down to
the “SIP Password:” section (Figure 11).
Figure 11 – Individual User SIP Settings
There is no default “SIP Password”: it is masked with the appearance that there is, but don’t be confused to think
that there’s a default password. You can modify it to any value you wish, but be certain to note what you changed it
to, as you will need it when configuring the SoundStation parameters. Save your changes.
SIP Profiles
ShoreWare Director’s “Call Control…” section contains an “SIP Profiles” option. ShoreTel 8 comes standard with
a “_System” and “_ShorePhoneIP8000” SIP profiles (they cannot be deleted - only disabled). By default, the
SoundStation system utilizes the “_System” profile. In order to optimize the functionality, you will need to add a
custom profile. This is accomplished from ShoreWare Director by selecting “Administration” followed by
“Call Control…,” then “SIP Profiles.” This action brings up the “SIP Profiles” screen. At the top of the
page below the “SIP Profiles List”, select the “New…” radio button, as shown in Figure 12.
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Figure 12 – SIP Profiles
This action brings up the “Edit SIP Profile” screen, Figure 13.
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Figure 13 – Edit SIP Profile
Define a “Name:” for the entry, and be sure to define an appropriate name. For the “User Agent:” option,
enter “ComdasysB2BUA” (without quotes); the “Priority:” defaults to 100, no change is required. Enable the
profile by checking (enabling) the “Enable” option. In the “Custom Parameters:” options, add the following
entries:
MWI=notify
FakeDeclineAsRedirect=1
Save the changes.
Note: Please do not disable any of the default SIP profiles. In case there are issues with the custom profile
defined, disabling the system profiles may cause the Polycom SoundStation phones to not be added to the ShoreTel
system. Refer to the ShoreTel 8 Server Release Notes for more information.
This completes all of the ShoreTel configuration parameters necessary to install the Polycom SoundStation.
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Polycom SoundStation IP Configuration
To setup the Polycom SoundStation IP conference phones with the ShoreTel system, it must first be installed and
operating on the network, please refer to the respective SoundStation IP Quick Start Guide or SoundPoint IP /
SoundStation IP Administrators Guide at http://www.polycom.com/voicedocumentation.
Note: Polycom recommends using configuration files hosted by a centralized server for all SoundStation IP
conference phone configuration and settings. Please see Polycom’s whitepaper on configuration file management in
the troubleshooting section at the end of this document. If a centralized server is unavailable, you may configure the
devices by hand as directed below.
SoundStation IP Phone Settings – Phone User Interface
The following settings will be managed through the phone’s user interface (PhoneUI):
1. Press the “Menu” button on the conference phone to enter the “Main Menu” screen.
2. Scroll down using the down arrow key to item “3. Settings…” and press the “Select” key to enter the
“Settings” screen, or press the number “3” on the dial key pad.
3. Scroll down using the down arrow key to item “2. Advanced…” and press the “Select” key, or press the
number “2” on the dial key pad.
4. At the “Password” display prompt, enter the password (the default is 456) and press the soft key under the
“Enter” text that appears on the LCD display.
5. On the “Advanced” screen, the “1. Admin Settings…” parameter should be highlighted. Press the “Select”
key or press the number “1” on the dial key pad to select the “Admin Settings” menu.
6. Scroll down using the down arrow key to item “2. SIP Configuration…” and press the “Select” key, or
press the number “2” on the dial key pad.
7. On the “SIP Configuration” screen for “Server:” press “Edit” and press the “1/A/a” soft key until the
screen title “SIP Configuration(1/Ascii)” appears.
8. Enter the IP address for the ShoreGear Switch using the keypad (use “*” key to enter the periods in the IP
address), and then press “Ok” to enter.
9. Press the “Back” key to exit the “Server Configuration” menu.
10. The “SIP Configuration” menu is displayed again and the “Outbound Proxy” parameter should be
highlighted. If it is not, use the scroll keys to highlight the “Outbound Proxy” parameter. Now press the
“Select” key.
11. The “Outbound Proxy” configuration menu is displayed with the “Address:” field highlighted, press the
“Edit” key.
12. On the “Outbound Proxy” sub-menu, press the “1/A/a” softkey until the screen title “Outbound
Proxy(1/Ascii)” appears.
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13. Enter the IP address for the ShoreGear Switch using the keypad (use “*” key to enter the periods in the IP
address), and then press “Ok” to enter.
14. Press the “Back” key to exit the “Outbound Proxy” configuration menu.
15. The “SIP Configuration” menu is displayed again and the “RFC2543 Hold” field will be highlighted.
Scroll down using the down arrow key to item “Line 1”, and then press the “Select” key to enter the
“Line1” field.
16. On the “Line 1” screen for the “Display Name:” prompt, press “Edit” to enter the display name.
17. Enter the desired display name (example: 3000) and press “Ok” to input the changes. Use the “1/A/a” soft
key to toggle between text and numbers.
18. Scroll down using the down arrow key to item “Address:” and press the “Edit” key.
19. Enter the desired address (example: 3000) and press “Ok” to input the changes. Use the “1/A/a” soft key to
toggle between text and numbers.
20. Scroll down using the down arrow key to item “Label:” and press the “Edit” key to enter the Label.
21. Enter the desired label (example: 3000) and press “Ok” to input the changes. Use the “1/A/a” soft key to
toggle between text and numbers.
22. Verify that the “Type:” field is configured to a value of “Private”. If it is not, press the “Edit” key, then
press either the up or down scroll key to toggle the parameter to “Private”, and then press the “Ok” key to
accept the change.
23. If you are using an Authorization Id and Password (Optional), scroll down to the prompts “Auth User ID:”
and “Auth Password” and enter a User ID and Password (using same text input procedure as in previous
steps).
24. Press “Back” key twice to enter “Network Configuration” screen
25. Option “2. Save Config” should be highlighted, If it is not, scroll to item “2. Save Config” and press
“Select” or press the number “2” on the dial key pad. The changes made will be saved and the
SoundStation IP conference phone will reboot.
Note: For forwarding to function, the full SIP address must be entered (e.g. sip:[email protected]).
Extended
forwarding features (Forward on busy, no answer, or unconditional) can be found in the HTTP admin
(http://<phone ip address> . Login=admin, Password=456).
SoundStation IP Phone Settings – Web User Interface
In order to adjust settings via the Web User Interface (WebUI), you need the SoundStation IP phone’s IP address.
Use the following steps to retrieve the SoundStation IP phone’s IP address.
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1. Press the “Menu” button on the conference phone to enter the “Main Menu” screen.
2. Scroll down using the down arrow key to item “2. Status…” and press the “Select” key to enter the
“Status” screen, or press the number “2” on the dial key pad.
3. Scroll down using the down arrow key to item “2. Network…” and press the “Select” key, or press the
number “2” on the dial key pad.
4. On the “Network” menu, the “1. TCP/IP Parameters...” parameter will be highlighted. If it is not, use the
scroll keys to highlight the “1. TCP/IP Parameters...”parameter, and then press the “Select” key.
5. On the LCD display, you will see the menu for “TCP/IP Parameters”. The IP address of your SoundStation
IP conference phone will be displayed at “IP:”xxx.xxx.xxx.xxx.
Once you have retrieved the IP address of the SoundStation IP conference phone you can set two additional
parameters.
1. Bring up a web browser on a PC/Laptop that is on the same network as, or with network access to, the
SoundStation IP conference phone.
2. Enter the IP address in the browser’s address bar, for example: http://xxx.xxx.xxx.xxx
3. You will be prompted for a password. At the password prompt, enter “Polycom” for the User Name and
“456” for the Password.
4. Once logged into the WebUI, navigate and click the menu item “SIP”, scroll down to the field “Digitmap
Impossible Match” and change from “0” to “2”.
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5. Click the “Submit” button. The phone will save the changes and reboot.
6. Once the SoundStation IP conference phone has returned to a booted status, Login via the WebUI again,
following steps 1 through 3 from this section.
7. Once logged into the WebUI, navigate and click the menu item “Lines”, scroll down to the section
“Message Center”.
8. Enter the value “102” into the fields for “Subscriber” and “Callback Contact”.
9. Click the “Submit” button. The phone will save the changes and reboot.
10. Upon restart, you will now be ready to use your SoundStation IP conference phone with your ShoreTel
system.
Polycom Troubleshooting
Refer to the following documents on Polycom.com for troubleshooting tips.
Polycom SoundStation IP Administration Guide Chapter 5 Troubleshooting
(http://www.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_Ad
min_Guide_SIP_3_1.pdf)
Technical Bulletins for SoundStation IP Phones
(http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical_Bulletins_pub.html)
White Paper for Configuration File Management for SoundStation IP Phones
(http://www.polycom.com/global/documents/support/technical/products/voice/white_paper_configurati
on_file_management_on_soundpoint_ip_phones.pdf)
Polycom Technical Support
If you have questions, please contact Polycom Support Hotline at (800) POLYCOM. The hotline is open Monday
through Friday, 6 a.m. to 6 p.m. Mountain time.
- 24 -
For Technical Support: [email protected]
For Knowledge Base: http://www.polycom.com/usa/en/support/voice/voice.html
Document and Software Copyrights
Copyright © 2009 by ShoreTel, Inc., Sunnyvale, California, U.S.A. All rights reserved. Printed in the United States
of America. Contents of this publication may not be reproduced or transmitted in any form or by any means,
electronic or mechanical, for any purpose, without prior written authorization of ShoreTel Communications, Inc.
ShoreTel, Inc. reserves the right to make changes without notice to the specifications and materials contained herein
and shall not be responsible for any damage (including consequential) caused by reliance on the materials
presented, including, but not limited to typographical, arithmetic or listing errors.
© 2009, Polycom, Inc. All rights reserved. POLYCOM®, the Polycom "Triangles" logo and the names and marks
associated with Polycom's products are trademarks and/or service marks of Polycom, Inc. and are registered and/or
common law marks in the United States and various other countries. All other trademarks are property of their
respective owners. No portion hereof may be reproduced or transmitted in any form or by any means, for any
purpose other than the recipient's personal use, without the express written permission of Polycom.
Trademarks
The ShoreTel logo, ShoreTel, ShoreCare, ShoreGear, ShoreWare and ControlPoint are registered trademarks of
ShoreTel, Inc. in the United States and/or other countries. ShorePhone is a trademark of ShoreTel, Inc. in the
United States and/or other countries. All other copyrights and trademarks herein are the property of their respective
owners.
Disclaimer
To be “ShoreTel Certified” means that Technology Partner's product will interoperate with the ShoreTel system,
but ShoreTel does not certify that the features or functionality of Technology Partner's product will perform as
specified by Technology Partner nor that Technology Partner's product will meet your specific application needs or
requirements. To inter-operate means that Technology Partner's product is able to exchange, use and share
information with the ShoreTel system.
Company Information
ShoreTel, Inc.
960 Stewart Drive
Sunnyvale, California 94085 USA
+1.408.331.3300
+1.408.331.3333 fax
- 25 -

Polycom SoundStation IP 4000, IP 5000, IP 6000, IP 7000

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