Application Notes for Configuring the Polycom VVX 500/600 running

Avaya Solution & Interoperability Test Lab
Application Notes for Configuring the Polycom VVX
500/600 running UC software release 5.0.0.7403 with Avaya
Aura® Session Manager and Avaya Aura® Communication
Manager Release 6.3 - Issue 1.0
Abstract
These Application Notes describe a solution for supporting interoperability between the
Polycom VVX 500/600 running UC software release 5.0.0.7403 with Avaya Aura® Session
Manager and Avaya Aura® Communication Manager release 6.3. Emphasis of the testing was
to verify voice calls of VVX 500/600 as a SIP endpoint registered to Avaya Aura® Session
Manager.
Information in these Application Notes has been obtained through DevConnect Compliance
testing and additional technical discussions. Testing was conducted via the DevConnect
Program at the Avaya Solution and Interoperability Test Lab.
RS; Reviewed:
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©2014 Avaya Inc. All Rights Reserved.
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1. Introduction
These Application Notes provide detail configurations of the Polycom VVX 500/600 (hereafter
referred to as VVX 500/600) with a SIP infrastructure consisting of Avaya Aura® Session
Manager (hereafter referred to as Session Manager) and Avaya Aura® Communication Manager
(hereafter referred to as Communication Manager). During compliance testing, VVX 500/600
successfully registered with Session Manager, established calls with other Avaya telephones and
all the applicable telephony features were executed on the VVX 500/600, where applicable, to
ensure the interoperability with Communication Manager.
2. General Test Approach and Test Results
The general test approach was to have the VVX 500/600 register to Session Manager. Calls were
then placed from Avaya telephone clients/users to and from the VVX 500/600. Other telephony
features such as busy, hold, DTMF, transfer, conference, video (where applicable) and codec
negotiation were also verified.
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and standards-based interfaces pertinent to
the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute a full product performance or feature testing performed by
third party vendors, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a third party solution.
2.1. Interoperability Compliance Testing
Interoperability compliance testing covered the following features and functionality:






Registration of VVX 500/600 to Session Manager.
Call establishment of VVX 500/600 with Avaya telephones.
Telephony features: Basic calls, conference, blind and consultative transfer, DTMF (dual
tone multi frequency), leaving and retrieving voicemail message, busy, hold, call
forward busy, call forward unconditional, call forward no answer, MWI (Message
Waiting Indicator), and Do not Disturb (DND).
Codec negotiation – G.711, G.729 and G.722.
Incoming and Outgoing calls to VVX 500/600 from PSTN.
Video call between two VVX 500/600 phones.
Note: Based on the micro-processor type, VVX 500 and VVX 600 belong to the same family
and therefore the test results of VVX 500 also holds good for VVX 600. During compliance
testing, only VVX 500 was tested.
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2.2. Test Results
The objectives outlined in Section 2.1 were verified. VVX 500/600 was registered to Session
Manager successfully. Calls have been made between Communication Manager telephones and
VVX 500/600 with clear voice path. All executed test cases passed with the following
observations,
 On Communication Manager only the option of G.722 – 64 is available and since this
option is not available from the VVX 500/600 codec list, this codec option could not be
tested.
 Call Forward on Busy (CFB) has to be configured on Communication Manager at the set
level and not through the Polycom Web Configuration Utility. However Call Forward
Unconditional (CFU) and Call Forward No Answer (CFNA) can be configured using the
Polycom Web Configuration Utility.
2.3. Support
Technical support for the Polycom VVX 500/600 can be obtained through Polycom global
technical support:
 Phone: 1-888-248-4143 or 1-408-474-2067
 Web: http://support.polycom.com
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3. Reference Configuration
Figure 1 illustrates a sample configuration with an Avaya SIP-based network that includes the
following Avaya products:
 Communication Manager running on an Avaya S8800 Server with a G650 Media
Gateway.
 Session Manager connected to Communication Manager via a SIP trunk and acting as a
Registrar/Proxy for SIP telephones.
 System Manager used to configure Session Manager.
 Avaya Aura® Messaging providing voice mail service for the SIP endpoints.
The VVX 500/600 registers with Session Manager and is configured as an Off-PBX Station
(OPS) on Communication Manager. Polycom Web Configuration Utility is used to manage the
configuration of the VVX 500/600 phone.
Figure 1: Network Configuration Diagram
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4. Equipment and Software Validated
The following equipment and software/firmware were used for the reference configuration:
Equipment/Software
Release/Version
Avaya Aura® Communication Manager
running on Avaya S8800 Server and
G650 Media Gateway
6.3-03.0.124.0
Avaya Aura® System Manager running on an
Avaya S8800 Server
6.3.0-FP2
Avaya Aura® Session Manager running on
S8800 Server.
6.3.2.0.632023
Avaya Aura® Messaging
6.1
Avaya 9620G IP (SIP) Telephone
6.2.0
Avaya 9608 IP ( H.323) Telephone
6.0.2
Polycom UC Software for VVX 500
5.0.0.7403
Polycom Web Configuration Utility
Windows XP Professional OS
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5. Configure Avaya Aura® Communication Manager
This section describes the steps for configuring an Off-PBX Station (OPS) that can be used for
VVX 500/600 and configuring a SIP trunk between Communication Manager and Session
Manager. Section 5.3 covers the station configuration that will be used by VVX 500/600. Use
the System Access Terminal (SAT) to configure Communication Manager and log in with the
appropriate credentials.
5.1. Verify OPS and SIP Trunk Capacity
Using the SAT, verify that the Off-PBX Telephones (OPS) and SIP Trunks features are enabled
on the system-parameters customer-options form. The license file installed on the system
controls these options. If a required feature is not enabled, contact an authorized Avaya sales
representative.
On Page 1, verify that the number of OPS stations allowed in the system is sufficient for the
number of SIP endpoints that will be deployed.
display system-parameters customer-options
Page
1 of
11
OPTIONAL FEATURES
G3 Version: V16
Location: 2
Platform: 28
Software Package: Enterprise
System ID (SID): 1
Module ID (MID): 1
Platform Maximum Ports:
Maximum Stations:
Maximum XMOBILE Stations:
Maximum Off-PBX Telephones - EC500:
Maximum Off-PBX Telephones OPS:
Maximum Off-PBX Telephones - PBFMC:
Maximum Off-PBX Telephones - PVFMC:
Maximum Off-PBX Telephones - SCCAN:
Maximum Survivable Processors:
65000
41000
41000
41000
41000
41000
41000
0
313
USED
213
37
0
4
24
0
0
0
1
(NOTE: You must logoff & login to effect the permission changes.)
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On Page 2 of the system-parameters customer-options form, verify that the number of SIP
trunks supported by the system is sufficient.
display system-parameters customer-options
OPTIONAL FEATURES
Page
IP PORT CAPACITIES
Maximum Administered H.323 Trunks:
Maximum Concurrently Registered IP Stations:
Maximum Administered Remote Office Trunks:
Maximum Concurrently Registered Remote Office Stations:
Maximum Concurrently Registered IP eCons:
Max Concur Registered Unauthenticated H.323 Stations:
Maximum Video Capable Stations:
Maximum Video Capable IP Softphones:
Maximum Administered SIP Trunks:
Maximum Administered Ad-hoc Video Conferencing Ports:
Maximum Number of DS1 Boards with Echo Cancellation:
Maximum TN2501 VAL Boards:
Maximum Media Gateway VAL Sources:
Maximum TN2602 Boards with 80 VoIP Channels:
Maximum TN2602 Boards with 320 VoIP Channels:
Maximum Number of Expanded Meet-me Conference Ports:
12000
18000
12000
18000
414
100
41000
18000
24000
24000
522
128
250
128
128
300
2 of
11
USED
0
6
0
0
0
0
0
1
130
0
0
1
0
0
1
0
(NOTE: You must logoff & login to effect the permission changes.)
5.2. Configure SIP Trunk
In the IP Node Names form, assign an IP address and host name for the S8800 Server processor,
and Session Manager. The host names will be used throughout the other configuration screens of
Communication Manager.
display node-names ip
IP NODE NAMES
Name
AES62
AVAYARDTT
CLAN1
CLAN2
DevCM3
GW
InteropSM62
LSP-1
MedPro1
MedPro2
SM61
Server-1
default
procr
procr6
IP Address
10.10.98.17
10.10.98.68
10.10.97.217
10.10.97.238
10.10.4.9
10.10.97.193
10.10.1.11
10.10.4.22
10.10.97.218
10.10.97.233
10.10.97.198
10.10.97.19
0.0.0.0
10.10.97.201
::
( 15 of 15
administered node-names were displayed )
Use 'list node-names' command to see all the administered node-names
Use 'change node-names ip xxx' to change a node-name 'xxx' or add a node-name
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In the IP Network Region form, the Authoritative Domain field is configured to match the
domain name configured on Session Manager. In this configuration, the domain name is
bvwdev.com. By default, Intra-region IP-IP Direct Audio and Inter-region IP-IP Direct
Audio (shuffling) is enabled to allow audio traffic to be sent directly between IP endpoints
without using media resources in the Avaya G650 Media Gateway. The IP Network Region
form also specifies the Codec Set to be used for calls routed over the SIP trunk to Session
Manager. This codec set is used when its corresponding network region (i.e., IP Network
Region ‘1’) is specified in the SIP signaling group.
display ip-network-region 1
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IP NETWORK REGION
Region: 1
Location: 1
Authoritative Domain: bvwdev.com
Name:
Stub Network Region: n
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio: yes
Codec Set: 1
Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048
IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 46
Audio PHB Value: 46
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5
AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS
RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
In the IP Codec Set form, select the audio codec type supported for calls routed over the SIP
trunk to VVX 500/600. The form is accessed via the change ip-codec-set 1 command. Note
that IP codec set ‘1’ was specified in IP Network Region ‘1’ shown above. The default settings
of the IP Codec Set form are shown below. However, the IP Codec Set form may specify
multiple codecs, including G.711, G.729A, and G.722, which are supported by VVX 500/600.
display ip-codec-set 1
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2
IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
2: G.729
3: G.722-64K
4:
5:
6:
7:
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Silence
Suppression
n
n
Frames
Per Pkt
2
2
2
Packet
Size(ms)
20
20
20
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Prior to configuring a SIP trunk group for communication with Session Manager, a SIP signaling
group must be configured. Configure the Signaling Group form as follows:








Set the Group Type field to sip.
Set the IMS Enabled field to n.
The Transport Method field was set to tcp. Specify the Communication Manager
Processor Interface (procr) and the Session Manager as the two ends of the signaling
group in the Near-end Node Name field and the Far-end Node Name field,
respectively. These field values are taken from the IP Node Names form.
Ensure that the recommended TCP port value of 5060 is configured in the Near-end
Listen Port and the Far-end Listen Port fields.
The preferred codec for the call will be selected from the IP codec set assigned to the IP
network region specified in the Far-end Network Region field.
Enter the domain name of Session Manager in the Far-end Domain field. In this
configuration, the domain name is bvwdev.com.
The Direct IP-IP Audio Connections field was enabled on this form.
The DTMF over IP field should be set to the default value of rtp-payload.
Communication Manager supports DTMF transmission using RFC 2833. Retain default
values for all other fields.
display signaling-group 1
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SIGNALING GROUP
Group Number: 1
Group Type: sip
IMS Enabled? n
Transport Method: tcp
Q-SIP? n
IP Video? y
Priority Video? n
Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: SM
Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? y
Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? n
Near-end Node Name: procr
Near-end Listen Port: 5060
Far-end Node Name: SM61
Far-end Listen Port: 5060
Far-end Network Region: 1
Far-end Domain: bvwdev.com
Incoming Dialog Loopbacks: eliminate
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? y
H.323 Station Outgoing Direct Media? n
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Bypass If IP Threshold Exceeded?
RFC 3389 Comfort Noise?
Direct IP-IP Audio Connections?
IP Audio Hairpinning?
Initial IP-IP Direct Media?
Alternate Route Timer(sec):
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n
y
n
n
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Configure the Trunk Group form as shown below. This trunk group is used for calls to the SIP
phones. Set the Group Type field to sip, set the Service Type field to tie, specify the signaling
group associated with this trunk group in the Group Number field, and specify the Number of
Members supported by this SIP trunk group. Configure an appropriate TAC value. Retain
default values for all other fields.
display trunk-group 1
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TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
1
Private trunk
two-way
n
0
tie
Group Type: sip
COR: 1
Outgoing Display? y
TN: 1
CDR Reports: n
TAC: #001
Night Service:
Auth Code? n
Member Assignment Method: auto
Signaling Group: 1
Number of Members: 15
On Page 3 of the trunk group form, set the Numbering Format field to private. This field
specifies the format of the calling party number sent to the far-end.
display trunk-group 1
TRUNK FEATURES
ACA Assignment? n
Page
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Measured: none
Maintenance Tests? y
Numbering Format: private
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Modify Tandem Calling Number: no
Show ANSWERED BY on Display? y
Configure the Private Numbering Format form to send the calling party number to the far-end.
Add an entry so that local stations with a 5-digit extension beginning with ‘53’ and whose calls
are routed over any trunk group, including SIP trunk group “1”, have the number sent to the farend for display purposes.
display private-numbering 0
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NUMBERING - PRIVATE FORMAT
Ext Ext
Trk
Len Code
Grp(s)
5 5
1
Maximum Entries: 540
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Private
Prefix
Total
Len
5
Total Administered: 1
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5.3. Configure Stations
Use the add station command to add a station for each VVX 500/600 phone to be supported.
Use 9620SIP for the Station Type and include the Coverage Path for voice mail, if applicable.
The Name field is optional. Use the default values for the other fields on Page 1. The SIP
station can also be configured automatically by Session Manager as described in Section 6.7.
display station 53113
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6
STATION
Extension:
Type:
Port:
Name:
53113
9620SIP
S00006
53113, Moto
Lock Messages? n
Security Code:
Coverage Path 1:
Coverage Path 2:
Hunt-to Station:
BCC:
TN:
COR:
COS:
0
1
1
1
STATION OPTIONS
Time of Day Lock Table:
Loss Group: 19
Message Lamp Ext: 53113
Display Language: english
Button Modules: 0
Survivable COR: internal
Survivable Trunk Dest? y
IP SoftPhone? n
IP Video? n
On Page 2, set the MWI Served User Type field to the appropriate value to allow MWI
notifications to be sent to VVX 500/600.
display station 53113
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STATION
FEATURE OPTIONS
LWC Reception: spe
LWC Activation? y
CDR Privacy? n
Per Button Ring Control? n
Bridged Call Alerting? n
Active Station Ringing: single
H.320 Conversion? n
Coverage Msg Retrieval?
Auto Answer:
Data Restriction?
Idle Appearance Preference?
Bridged Idle Line Preference?
Restrict Last Appearance?
y
none
n
n
n
y
Per Station CPN - Send Calling Number?
EC500 State: enabled
MWI Served User Type: qsig-mwi
Coverage After Forwarding? s
Emergency Location Ext: 53113
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Direct IP-IP Audio Connections? y
Always Use? n IP Audio Hairpinning? n
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Use the change off-pbx-telephone station-mapping command to map Communication
Manager extensions (e.g., 53113) to the same extension configured in Session Manager. Enter
the field values shown. For the sample configuration, the Trunk Selection field is set to aar so
that AAR call routing is used to route calls to Session Manager. AAR call routing configuration
is not show in these Application Notes. The Configuration Set value can reference a set that
has the default settings.
change off-pbx-telephone station-mapping 53113
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station
Extension
53113
Application Dial
CC Phone Number
Prefix
OPS
53113
Page
Trunk
Selection
aar
1 of
Config
Set
1
3
Dual
Mode
On Page 2, change the Call Limit to match the number of call-appr entries in the station form.
Also, verify that Mapping Mode is set to both (the default value for a newly added station).
change off-pbx-telephone station-mapping 53113
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station
Extension
53113
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Appl
Name
OPS
Call
Limit
3
Mapping
Mode
both
Calls
Allowed
all
Page
Bridged
Calls
none
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6. Configure Avaya Aura® Session Manager
This section provides the procedures for configuring Session Manager. The procedures include
adding the following items:
 SIP domain.
 Logical/physical Locations that can be occupied by SIP Entities.
 SIP Entities corresponding to Session Manager and Communication Manager.
 Entity Links, which define the SIP trunk parameters used by Session Manager when
routing calls to/from SIP Entities.
 Application Sequence.
 Define Communication Manager as Administrable Entity (i.e., Managed Element).
 Session Manager, to be managed by System Manager.
 Add SIP Users.
Configuration is accomplished by accessing the browser-based GUI of System Manager using
the URL “https://<ip-address>/SMGR”, where <ip-address> is the IP address of System
Manager. Log in with the appropriate credentials and accept the Copyright Notice.
The main screen of System Manager is seen as shown below.
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6.1. Specify SIP Domain
Add the SIP domain for which the communications infrastructure will be authoritative. Navigate
to Routing  Domains on the left and clicking the New button on the right. The following
screen will then be shown. Fill in the following:
 Name: The authoritative domain name (e.g., bvwdev.com).
 Notes: Descriptive text (optional).
Click Commit (not shown).
Since the sample configuration does not deal with any other domains, no additional domains
need to be added.
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6.2. Add Locations
Locations can be used to identify logical and/or physical locations where SIP Entities reside for
purposes of bandwidth management. To add a location, select Locations on the left and click on
the New button on the right (not shown). The following screen will then be shown. Fill in the
following:
Under General:
 Name:
 Notes:
Under Location Pattern:
 IP Address Pattern:
 Notes:
A descriptive name.
Descriptive text (optional).
A pattern used to logically identify the location.
Descriptive text (optional).
The screen below shows addition of the Belleville location, which includes the Communication
Manager and Session Manager. Click Commit to save the Location definition. Retain default
values for all other fields.
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6.3. Add SIP Entities
In the sample configuration, a SIP Entity is added for Session Manager and Communication
Manager.
6.3.1. Avaya Aura® Session Manager
A SIP Entity must be added for Session Manager. To add a SIP Entity, select SIP Entities on the
left and click on the New button on the right (not shown). The following screen is displayed. Fill
in the following:
Under General:
 Name:
 FQDN or IP Address:
 Type:
 Location:
 Time Zone:
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A descriptive name.
IP address of the signaling interface on Session Manager.
Specify Session Manager.
Select one of the locations defined previously.
Time zone for this location.
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Under Port, click Add, and then edit the fields in the resulting new row as shown below:
 Port:
Port number on which the system listens for SIP requests.
 Protocol:
Transport protocol to be used to send SIP requests. During
compliance testing only TCP was used.
 Default Domain:
The domain used for the enterprise (e.g. bvwdev.com).
Defaults can be used for the remaining fields. Click Commit to save each SIP Entity definition.
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6.3.2. Avaya Aura® Communication Manager
A SIP Entity must be added for Communication Manager. To add a SIP Entity, select SIP
Entities on the left and click on the New button on the right (not shown). The following screen
is displayed. Fill in the following:
Under General:
 Name:
 FQDN or IP Address:



Type:
Location:
Time Zone:
A descriptive name.
IP address of the signaling interface (e.g., procr interface)
on the telephony system.
Specify CM.
Select one of the locations defined previously.
Time zone for this location.
Retain default values for all other fields. Click Commit to save each SIP Entity definition.
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6.4. Add Entity Link
The SIP trunk from Session Manager to Communication Manager is described by an Entity link.
To add an Entity Link, select Entity Links on the left and click on the New button on the right
(not shown). Fill in the following fields in the new row that is displayed:




Name:
SIP Entity 1:
Protocol:
Port:


SIP Entity 2:
Port:

Connection Policy
A descriptive name
Select the Session Manager.
Select the appropriate protocol.
Port number to which the other system sends SIP
requests.
Select the name of Communication Manager.
Port number on which the other system receives
SIP requests.
Check this box. Note: If this box is not checked,
calls from the associated SIP Entity specified in
Section Error! Reference source not found. will be denied.
Click Commit to save the Entity Link definition.
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6.5. Define Communication Manager as Managed Element
Before adding SIP users, Communication Manager must be added to System Manager as a
managed element. This action allows System Manager to access Communication Manager over
its administration interface. Using this administration interface, System Manager will notify
Communication Manager when new SIP users are added.
To define Communication Manager as a managed element, under Services (refer to screen shot
in Section 6.3) navigate to Inventory  Manage Elements on the left and click on the New
button on the right (not shown). In the Application Type field that is displayed (not shown),
select Communication Manager.
Screen below shows an already added Communication Manager. Enter the values as follows and
retain default values for all other fields.
Under General Attributes (G):
 Name:
 Description:
 Hostname or IP Address:



Login:
Password:
Confirm Password:
Enter an identifier for Communication Manager.
Enter an appropriate description.
Enter the IP address of the administration interface for
Communication Manager.
A login name.
Enter password.
Confirm above entered password.
Click Commit to save the settings.
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6.6. Add Application Sequence
Define an application for Communication Manager. Under Elements (refer to screen shot in
Section 6.3) navigate to Session Manager  Application Configuration Applications on
the left and click on the New button on the right (not shown). Fill in the following fields:




Name:
SIP Entity:
CM System for SIP Entity:
Description:
An appropriate name.
Select the Communication Manager SIP entity.
Select the Communication Manager managed element.
An appropriate description.
Click Commit to save the Application definition.
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Next, define the Application Sequences for Communication Manager as shown below.
Under Elements (refer to screen shot in Section 6.3) navigate to Session Manager 
Application Configuration Application Sequences on the left and click on the New button
on the right (not shown). Fill in the following fields:
Enter a descriptive name in the Name field.
Click Commit to save the Application Sequence.
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6.7. Add SIP Users
Add a SIP user corresponding to the VVX 500/600 as defined in Section 5.3. Alternatively, use
the option to automatically generate the SIP stations on Communication Manager
Feature/Evolution Server when adding a new SIP user. Under Users (refer to screen shot in
Section 6.3) navigate to User Management  Manage Users (not shown) on the left and click
on the New button on the right (not shown).
Under the Identity tab enter values for the following required attributes for a new SIP user in the
new user form:



Last Name:
First Name:
Login Name:



Authentication Type:
Password:
Confirm Password:
Enter the last name of the user.
Enter the first name of the user.
Enter <extension>@<sip domain> of the
user (e.g., 53113@bvwdev.com).
Select Basic (by default).
Password to be used by the SIP User.
Re-enter the password from above.
The screen below shows the information when adding a new SIP user during compliance testing.
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Click on the Communication Profile tab and select New to define a Communication Profile
for the new SIP user. Enter a password in the Communication Profile Password and Confirm
Password fields. Enter values for the following required fields:


Name:
Default:
Enter name of communication profile.
Select field to indicate that this is the default profile.
Click New to define a Communication Address for the new SIP user. Enter values for the
following required fields:


Type:
Fully Qualified Address:
Select Avaya SIP.
Enter extension number and SIP domain.
The screen below shows the information when adding a new SIP user to the sample
configuration. Click Add.
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In the Session Manager Profile section, enter the following values,
 Under SIP Registration, select the Session Manager from the drop down list for the
Primary Session Manager field.
 Assign the Application Sequence defined in Section 6.6 to the new SIP user as part of
defining the SIP Communication Profile. The Application Sequence can be used for
both the originating and terminating sequence.
 Select the required Home Location value from the drop down menu.
Retain default values for all other fields.
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In the CM Endpoint Profile section, fill in the following fields:

System:



Profile Type:
Extension:
Template:
Select the managed element corresponding to
Communication Manager.
Select Endpoint.
Enter extension number of SIP user.
Select template for type of SIP phone.
Retain default values for all other fields.
The screen below shows the configuration used for compliance testing.
Click Commit (not shown) to save the User Profile.
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6.8. Add Session Manager
To complete the configuration, adding Session Manager will provide the linkage between
System Manager and Session Manager. Under Elements (refer to screen shot in Section 6.3)
navigate to Session Manager  Session Manager Administration (not shown). Then click
New (not shown), and fill in the fields as described below and shown in the following screen:
Under General:
 SIP Entity Name:


Description:
Management Access
Point Host Name/IP:
Under Security Module:
 Network Mask:

Default Gateway:
Select the name of the SIP Entity added for
Session Manager.
Descriptive comment (optional).
Enter the IP address of the Session
Manager management interface.
Enter the network mask corresponding to the IP
address of Session Manager.
Enter the IP address of the default gateway for
Session Manager.
Retain default values for the remaining fields. Click Save to add this Session Manager (not
shown).
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7. Polycom Web Configuration Utility
This section shows how to log in to the home page of Polycom Web Configuration Utility that is
required to configure the VVX 500/600 phone.
Find the IP address assigned to the VVX 500/600 phone and type it into the URL address bar of
a web browser. The web configuration utility login interface will be displayed as shown below.
Select the Admin radio button and type in the default password of 456.
Click Submit, the homepage of the Polycom VVX 500 is shown below.
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7.1. Configure the Lines for Polycom VVX 500/600
This section shows how to configure the VVX 500/600 to register with Session Manager.
On the homepage of configuration screen, click on the Simple Setup menu, the Simple Setup
page appears as shown below. Enter the following values,
- Phone Language: English (internal)
- Time Zone: Select time zone for the region.
- Under SIP Server section, Address: 10.10.97.198 and Port: 5060; configured in Section
6.3.1.
- Under SIP Outbound Proxy section, Address: 10.10.97.198 and Port: 5060; configured
in Section 6.3.1
- Under the SIP Line Identification section, Display Name: an appropriate name,
Address: 53113, Authentication User ID: 53113 and Authentication Password: 1234;
configured in Section 6.7
Click on Save.
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7.2. SIP Settings
This section shows how to set SIP parameters for VVX 500/600.
On the homepage of VVX 500/600, navigate to menu Settings  SIP (not shown), SIP screen is
shown below. Enter the following values and retain rest at default.
- Under the Outbound Proxy section, Address: 10.10.97.198 and Port: 5060; configured
in Section 6.3.1 Transport: UDPOnly.
- Under the Server1 section, Address: 10.10.97.198 and Port: 5060; configured in
Section 6.3.1.Transport: UDPOnly.
Click on Save.
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7.3. Local Call Forward Settings
This section shows how to set up call forward settings for Polycom VVX 500/600.
On the homepage of Polycom VVX 500/600, navigate to menu Settings  Lines (not shown).
Line1 screen is shown below. Enter the following values and retain rest at default.
- Under the Call Diversion section, ensure that the Enforced by Server radio button is
No.
- Always Forward: Enable and configure an appropriate Directory Number (DN) for the
Always Forward To Contact field.
- On No Answer, Forward: Enable and configure an appropriate Directory Number (DN)
for the On No Answer, Forward to Contact field. Configure an appropriate value on
the On No Answer, forward After Rings field.
Click on Save. As mentioned in Section 2.2, If Busy, Forward option does not function if
configured here and has to be configured on the Communication Manager at the set level.
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7.4. Configuring Message Center for Message Waiting Indicator
This section shows how to set up activation of MWI for Polycom VVX 500/600.
On the homepage of Polycom VVX 500/600, navigate to menu Settings  Lines (not shown).
Line1 screen is shown below. Enter the following values and retain rest at default.
- Under the Message Center section, configure an appropriate Directory Number (DN) for
the Subscription Address field. During compliance testing 53113 was the DN
configured.
- Select Registration from the drop down menu for the Callback Mode
Click on Save.
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7.5. Codec Settings
On the homepage of Polycom VVX 500/600, navigate to menu Settings  Audio Codec
Priority (not shown). Select the codec list as shown below. Click Save.
8. Verification Steps
From the main screen of System Manager as shown in Section 6.0, select Session Manager (not
shown).
From the Session Manager screen shown below, navigate to System Status  User
Registrations to see a list of phones registered to the Session Manager.
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From the physical phone display of VVX 500/600 navigate to Menu  Settings  Status 
Lines (not shown). Verify that the Lines information shows the successful registration of the
VVX 500/600 phone to the Session Manager.
Place a call from and to the VVX 500/600 and verify that the call is established with 2-way
speech path. Verify basic telephony features by establishing calls between VVX 500/600 and
phones on Communication Manager.
9. Conclusion
These Application Notes illustrate the procedures necessary for configuring the Polycom VVX
500/600 to interoperate with Avaya Aura® Communication Manager and Avaya Aura® Session
Manager. All feature functionality test cases described in Section 2.1 were passed along with the
observations noted in Section 2.2.
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10. Additional References
Product documentation for the Avaya products may be found at:
https://support.avaya.com
Product documentation for the Polycom VVX family of phones may be found at:
http://support.polycom.com
[1] Administering Avaya Aura® Communication Manager Server Options, July 2012, Release
6.2, Issue 3.0, Document Number 03-603479.
[2] Administering Avaya Aura® Session Manager, July 2012, Release 6.2, Document Number
03-603324.
[3] Polycom VVX 500/600 Documents:
http://support.polycom.com/PolycomService/support/us/support/voice/index.html
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©2014
Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DevConnect Program at devconnect@avaya.com.
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