Polycom SoundStation IP 7000 User manual

Category
IP phones
Type
User manual

Polycom SoundStation IP 7000 is a breakthrough conference phone that delivers outstanding performance and a robust feature set for SIP-based VoIP platforms, ideal for executive offices, conference rooms, and board rooms.

With Polycom HD Voice technology, SoundStation IP 7000 turns ordinary conference calls into crystal-clear interactive conversations, boosting productivity and reducing listener fatigue. 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.

Polycom SoundStation IP 7000 is a breakthrough conference phone that delivers outstanding performance and a robust feature set for SIP-based VoIP platforms, ideal for executive offices, conference rooms, and board rooms.

With Polycom HD Voice technology, SoundStation IP 7000 turns ordinary conference calls into crystal-clear interactive conversations, boosting productivity and reducing listener fatigue. 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.

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 IP 6000 ................................. 4
Polycom Soundstation IP5000 .................................. 4
Polycom SoundStation IP 4000 ................................. 5
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.
T P P A PP N O T E
TPP-10011
Date: April, 2010
Product: Polycom® SoundStation® IP 4000,
5000, 6000, and 7000
System version: ShoreTel 9.2
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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
Lori.schimpf@polycom.com
Reseller Info:
Resellers who would like to start selling this solution should contact us using contact information at:
lori.schimpf@polycom.com
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
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 SoundStation IP 6000
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 SIP-
enabled 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)
North America Product Code: 2200-15660-001
Approximate List price: $1029 US MSR
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 roomsall 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 full-
duplex 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/soundstation-
ip5000-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
Polycom SoundStation IP 4000
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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North America Product Code: 2200-06640-001
Approximate List price: $1139 US MSR
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
SoundStation
IP 5000
SoundStation
IP 6000
SIP 3.1.2
SIP 3.1.2
SIP 3.1.2
ShoreTel
Release
6.1
7.0
7.5
8.0
8.1
9.2
Certification Testing Results Summary
Table 1: Basic Test Cases
ID
Name
Description
Results
1.1
Device initialization with
static IP address
Verify successful startup and initialization of the device
up to a READY/IDLE state using a static IP address
Pass
1.2
Device reset idle (for
static configurations)
Verify successful re-initialization of device after power
loss while device is idle
Pass
1.3
Device initialization with
DHCP
Verify successful startup and initialization of the device
up to a READY/IDLE state using DHCP
Pass
1.4
Device reset idle (for
dynamic configurations)
Verify successful re-initialization of device after power
loss while device is idle
Pass
1.5
Verify Diffserv Code
Point support
Verify the ability to set Diffserv Code Point from SIP
DUT (device under test)
Not Tested
1.6
Verify Date and Time
Update support
Verify setting of Date and Time Update on SIP DUT
Pass
1.7
Place call
Verify successful call placement with normal dialing to a
variety of terminating phones
Pass
1.8
Receive call
Verify successful call placement with normal dialing to a
variety of terminating phones
Pass
1.9
CODEC support (DUT to
ShoreTel Phone)
Verify successful call connection and audio path using all
supported CODECs (G.711-Ulaw and G.729)
Pass
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ID
Name
Description
Results
1.10
CODEC support (DUT to
SIP reference)
Verify successful call connection and audio path using all
supported CODECs (G.711-Ulaw and G.729)
Pass
1.11
CODEC negotiation
Verify successful negotiation between devices configured
with different default CODECs (G.711-Ulaw and G.729)
Pass
1.12
Hold DUT to SIP
reference
Verify successful hold and resume of connected call
Pass
1.13
Hold DUT to ShoreTel
Verify successful hold and resume of connected call
Pass
1.14
Forward
Verify successful forwarding of incoming calls
Pass
1.15
Forward from SIP DUT
Verify successful forwarding of incoming calls
Pass
1.16
Dual-tone multi-
frequency (DTMF)
transmission
Verify successful transmission of in-band and out-of-
band digits (RFC2833) for calls placed to and from the
DUT with a variety of other devices
Pass
RFC2833
only
Table 2: Extended Feature Test Cases
ID
Name
Description
Notes
2.1
Call waiting
Verify appropriate notification and successful connection
of incoming call while busy with another party
Pass
2.2
Park
Verify successful park and retrieval of connected call
Pass
2.3
Transfer blind
Verify successful blind transfer of connected call
Pass
2.4
Transfer monitored
Verify successful monitored transfer of connected call
Pass
2.5
Conference ad hoc
Verify successful ad hoc conference of three parties
Pass
2.6
Caller ID
Verify that Caller ID name and number is sent and
received from SIP endpoint device
Pass,
displays
Caller
Name only
on LCD, but
it does have
the Caller
ID in
history.
2.7
911
Verify dialing911” on DUT could connect with “911
services
Not Tested
2.8
Auto Attendant Menu
Verify that calls are properly terminated on the ShoreTel
Auto Attendant menu and that you can transfer to the
desired extension.
Pass
2.9
Auto Attendant Menu
“Dial by Name”
Verify that calls are properly terminated on the ShoreTel
Auto Attendant menu and that you can transfer to the
desired extension using theDial by Name” feature.
Pass
2.10
Auto Attendant Menu
checking Voice Mail
mailbox
Verify that calls are properly terminated on the ShoreTel
Auto Attendant menu and that you can transfer to the
Voice Mail Login Extension.
Pass
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ID
Name
Description
Notes
2.11
Initiate call to a Hunt
Group
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.
Pass
2.12
Initiate call to a
Workgroup
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.
Pass
2.13
Hunt Group Member
Verify that incoming calls to a hunt group can be
answered properly when DUT is a member of the hunt
group.
Pass
2.14
Workgroup Agent
Verify that incoming calls to a workgroup can be
answered properly when DUT is an agent of the
workgroup.
Pass
2.15
Call Forward FindMe”
Verify that calls are forwarded to DUTs “FindMe
destination.
Pass
2.16
ShoreTel Converged
Conferencing Server
Verify that calls are properly forwarded to the ShoreTel
Converged Conferencing Server and it properly accepts
the access code and youre able to participate in the
conference bridge.
Pass
2.17
Bridged Call Appearance
(BCA) extension
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.
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
Userssection below.
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Call Control Settings
The Call Control Options within ShoreWarDirector 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 un-
checking the box and saving the setting. When enabled, SIP extension configuration will fail.
It is also important to note that this “one timesetting 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,
thenSwitches” in ShoreWare Director (Figure 4).
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Figure 4 Administration/Switches
This action brings up the “Switches” screen. From the “Switchesscreen, 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 thePort Type” settings from the available ports to100
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 hasbuilt-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 Proxyconfiguration 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 selectingAdministrationthenSites(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. TheEdit Site” screen will then appear. Scroll down to the “SIP Proxy” parameters (Figure
8).
Figure 8 Site Screen SIP Proxies
The “Virtual IP Addressparameter is a new configuration parameter with ShoreTel 8. This Virtual IP
Addressis an IP address that can be moved to a different switch during a failure. For each site that
supports SIP extensions, oneVirtual IP Addressis 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 Addressto 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 Addressmechanism, SIP phones will not know if the
proxy switch goes off-line.
Note: If you choose not to define aVirtual 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 asCodec Listsand
defined in the sites page for “Inter-site” and “Intra-sitecalls. 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 selectingAdministration” followed byUsers…, then Individual Users. This action
will bring up the “Individual Usersscreen 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).
Figure 10 Adding/Editing Users
Define the “First Name” andLast 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 its not necessary to have a mailbox.
Define the properUser 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).
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Figure 11 Individual User SIP Settings
There is no default “SIP Password: it is masked with the appearance that there is, but dont 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 “_Systemand “_ShorePhoneIP8000SIP profiles (they cannot be deleted - only disabled). By default, the
SoundStation system utilizes the “_Systemprofile. 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 7000 User manual

Category
IP phones
Type
User manual

Polycom SoundStation IP 7000 is a breakthrough conference phone that delivers outstanding performance and a robust feature set for SIP-based VoIP platforms, ideal for executive offices, conference rooms, and board rooms.

With Polycom HD Voice technology, SoundStation IP 7000 turns ordinary conference calls into crystal-clear interactive conversations, boosting productivity and reducing listener fatigue. 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.

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