Application Notes for Bittel UNO Voice SIP Telephone with Avaya

Application Notes for Bittel UNO Voice SIP Telephone with Avaya
Avaya Solution & Interoperability Test Lab
Application Notes for Bittel UNO Voice SIP Telephone with
Avaya Aura® Communication Manager and Avaya Aura®
Session Manager - Issue 1.0
Abstract
These Application Notes describe the steps required to integrate Bittel UNO Voice SIP
Telephones with Avaya Aura® Communication Manager and Avaya Aura® Session Manager.
Bittel UNO Voice SIP Telephones are hotel guest phones that provide the following features:
speakerphone, hold, redial, message waiting indicator (MWI), and programmable buttons. In
the compliance test, Bittel UNO Voice SIP Telephones successfully registered with Avaya
Aura® Session Manager, established calls with other Avaya SIP and H.323 telephones, and
executed telephony and hospitality features using Avaya Aura® Communication Manager
Feature Name Extensions (FNEs).
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.
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1. Introduction
These Application Notes describe the steps required to integrate Bittel UNO Voice SIP
Telephones with Avaya Aura® Communication Manager and Avaya Aura® Session Manager.
Bittel UNO Voice SIP Telephones are hotel guest phones that provide the following features:
speakerphone, hold, redial, message waiting indicator (MWI), and programmable buttons. In the
compliance test, Bittel UNO Voice SIP Telephones successfully registered with Avaya Aura®
Session Manager, established calls with other Avaya SIP and H.323 telephones, and executed
telephony and hospitality features using Avaya Aura® Communication Manager Feature Name
Extensions (FNEs).
2. General Test Approach and Test Results
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent
to the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute full product performance or feature testing performed by
DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a DevConnect member’s solution.
The interoperability compliance test included feature and serviceability testing. The feature
testing focused on establishing calls between Bittel UNO Voice and Avaya SIP and H.323
telephones and exercising basic telephony features, such as hold and mute, and hospitality
features, including wake up calls and updating housekeeping status for a guest’s room. In
addition, other extended telephony features, such as call forwarding and call pickup were also
exercised using FNEs.
The serviceability testing focused on verifying that the Bittel UNO Voice SIP telephone comes
back into service after re-connecting the Ethernet connection or rebooting the SIP phone.
2.1. Interoperability Compliance Testing
Interoperability compliance testing covered the following features and functionality:
SIP registration of Bittel UNO Voice with Session Manager.
Calls between Bittel UNO Voice and Avaya SIP and H.323 telephones with Direct IP-IP
Media (Shuffling) enabled and disabled.
G.711 and G.729 codec support.
Proper recognition of DTMF tones.
Basic telephony features including Hold, Mute, and Redial.
Long call duration and long hold duration.
Extended telephony features using Communication Manager Feature Name Extensions
(FNEs) such as Hospitality Wakeup calls, Housekeeping Status Access Codes, Call
Forwarding and Call Pickup.
Voicemail coverage, MWI support, and logging into voicemail system to retrieve
voicemail messages.
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Use of programmable buttons on Bittel UNO Voice.
Proper system recovery after a restart of the Bittel UNO Voice and loss of IP
connectivity.
2.2. Test Results
All test cases passed with the following observations noted:
The Bittel UNO Voice SIP Telephone Model HA9888 (67) TSD-IP, which was used for
the compliance test, does not support multiple calls, caller ID display, and initiating a call
transfer or conference call.
While Bittel UNO Voice SIP Telephone is dialing a call, DTMF tones are heard by the
caller. After the call is established, DTMF tones are not heard when the keypad is
pressed. For example, when logging into voicemail, DTMF tones are not heard by the
caller, but the DTMF tones are sent to the voicemail system and the user is able to log in.
Bittel UNO Voice does not provide an indication to the caller when the Long Hold Recall
Timer expires. When Communication Manager sends the re-INVITE message to Bittel
UNO Voice when the Long Hold Recall Timer expires, Bittel UNO responds to the
message but doesn’t provide a reminder to the caller that there is a call on hold.
If a call attempt fails for whatever reason, such as calling a busy telephone, dialing
invalid number, or using an unsupported codec, Bittel UNO Voice plays busy tone for 30
seconds and then hangs up or the caller may hang up.
2.3. Support
For technical support on the Bittel UNO Voice SIP Telephone, contact Bittel Customer Service
and Support via phone, email, or website.
Phone: 86-633-2212103
Email: info@bittelgroup.com
Web: http://www.bittelphone.com/en/products/support.asp
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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:
Avaya Aura® Communication Manager running on an Avaya S8800 Server with a G650
Media Gateway. Communication Manager was configured as an Evolution Server.
Avaya Aura® Session Manager connected to Communication Manager via a SIP trunk
and acting as a Registrar/Proxy for SIP telephones.
Avaya Aura® System Manager used to configure Session Manager.
Avaya Aura® Messaging serving as the voicemail system.
Avaya 9600 Series SIP and H.323 Telephones.
Bittel UNO Voice SIP Telephones.
Bittel UNO Voice SIP Telephones registered with Session Manager and were configured as OffPBX Stations (OPS) on Communication Manager.
Figure 1: Avaya SIP Network with Bittel UNO Voice SIP Telephones
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3.1. SIP Call Flows
Bittel UNO Voice SIP Telephones originate a call by sending a call request (SIP INVITE
message) to Session Manager, which then routes the call over a SIP trunk to Communication
Manager for origination services. If the call is destined for another local SIP phone,
Communication Manager routes the call back over the SIP trunk to Session Manager for delivery
to the destination SIP phone. If the call is destined for a H.323 telephone, Communication
Manager routes the call directly to the H.323 endpoint.
For a call arriving at Communication Manager that is destined for a Bittel UNO Voice SIP
Telephone, Communication Manager routes the call over the SIP trunk to Session Manager for
delivery to the Bittel UNO Voice SIP Telephone.
4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Equipment/Software
Release/Version
Avaya Aura® Communication Manager
running on an Avaya S8800 Server with
and G650 Media Gateway
6.3 SP 1 (R016x.03.0.124.0 w/Patch 20850)
Avaya Aura® Session Manager
6.3 SP 3 (6.3.3.0.633004)
Avaya Aura® System Manager
6.3.3
(Build No. – 6.3.0.8.5682-6.3.8.1814,
Software Update Revision No: 6.3.3.5.1719)
Avaya Aura® Messaging
6.2 SP 2
Avaya 9600 Series IP Telephones
3.2 (H.323)
2.6.9.1 (SIP)
Bittel UNO Voice SIP Telephone
10.19_0.94.003
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5. Configure Avaya Aura® Communication Manager
This section provides the procedures for configuring Communication Manager. The procedures
include the following areas:
Verify Communication Manager license
Administer IP Network Region and IP Codec Set
Administer Hospitality Features
Use the System Access Terminal (SAT) to configure Communication Manager and log in with
the appropriate credentials.
Note: It is assumed that basic configuration of the Communication Manager has already been
completed, such as the SIP trunk to Session Manager. The SIP station configuration for Bittel
UNO Voice is configured through Avaya Aura® System Manager in Section 6.2.
5.1. Verify License
Using the SAT, verify that the Off-PBX Telephones (OPS) option is enabled on the systemparameters 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
OPTIONAL FEATURES
G3 Version: V16
Location: 2
Platform: 28
Page
1 of
11
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
118
21
0
0
9
0
0
0
0
(NOTE: You must logoff & login to effect the permission changes.)
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5.2. Administer IP Network Region and IP Codec Set
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
avaya.com. By default, 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 IP Codec Set to be used for calls
routed over the SIP trunk to Session Manager.
change ip-network-region 1
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20
IP NETWORK REGION
Region: 1
Location: 1
Authoritative Domain: avaya.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? y
UDP Port Max: 65535
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 34
Audio PHB Value: 46
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 7
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 the Bittel UNO Voice SIP Telephone. 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. The Bittel UNO Voice SIP
Telephone supports G.711 and G.729 codecs.
change ip-codec-set 1
Page
1 of
2
IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
2:
3:
4:
5:
6:
7:
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Silence
Suppression
n
Frames
Per Pkt
2
Packet
Size(ms)
20
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5.3. Administer Hospitality Features
This section covers the configuration of two hospitality features: wakeup calls and housekeeping
status. A hotel guest may enter the wake up feature access code (FAC) followed by the time for
the wakeup call in hhmm format, where hh is the hour and mm is the minute. The housekeeping
status of a hotel room may be changed by dialing the housekeeping status access code from the
hotel room phone.
5.3.1. Administer Features Access Codes (FACs)
In the Feature Access Code (FAC) form configure the Automatic Wakeup Call Access Code
and the Housekeeping Status (Client Room) Access Codes, as needed, as shown below. The
FACs should comply with the dial plan.
change feature-access-codes
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10
FEATURE ACCESS CODE (FAC)
Hospitality Features
Automatic Wakeup Call
Housekeeping Status (Client Room)
Housekeeping Status (Client Room)
Housekeeping Status (Client Room)
Housekeeping Status (Client Room)
Housekeeping Status (Client Room)
Housekeeping Status (Client Room)
Housekeeping Status (Station)
Housekeeping Status (Station)
Housekeeping Status (Station)
Housekeeping Status (Station)
Verify Wakeup Announcement
Voice Do Not Disturb
Access
Access
Access
Access
Access
Access
Access
Access
Access
Access
Access
Access
Access
Code:
Code:
Code:
Code:
Code:
Code:
Code:
Code:
Code:
Code:
Code:
Code:
Code:
*70
*71
*72
*73
*74
*75
*76
*77
*78
*79
*80
*81
5.3.2. Administer Feature Name Extensions (FNEs)
Prior to dialing the wakeup call or housekeeping status access codes, the SIP user must first
receive dial tone from Communication Manager. This is achieved by first dialing the Idle
Appearance Select FNE configured as shown below. Afterwards, the wakeup or housekeeping
status access code may be dialed.
change off-pbx-telephone feature-name-extensions set 1
EXTENSIONS TO CALL WHICH ACTIVATE FEATURES BY NAME
Exclusion (Toggle On/Off):
Extended Group Call Pickup:
Held Appearance Select:
Idle Appearance Select:
Last Number Dialed:
Malicious Call Trace:
Malicious Call Trace Cancel:
Off-Pbx Call Enable:
Off-Pbx Call Disable:
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78117
78118
78119
78120
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5.3.3. Allow Wake-up Calls
In the Hospitality form, enable Room Activated Wakeup With Tones. Communication
Manager will prompt the user with tones when enabling a wakeup call. For example, a 3-burst
confirmation tone will be played to prompt the user to enter the wakeup time.
change system-parameters hospitality
HOSPITALITY
Dual Wakeups? n
Page
Daily Wakeup? n
2 of
3
VIP Wakeup? n
Room Activated Wakeup With Tones? y
Time of Scheduled Wakeup Activity Report:
Time of Scheduled Wakeup Summary Report:
Time of Scheduled Emergency Access Summary Report:
Announcement Type: silence
Length of Time to Remain Connected to Announcement: 30
Extension to Receive Failed Wakeup LWC Messages:
Routing Extension on Unavailable Voice Synthesis:
Display Room Information in Call Display? n
Automatic Selection of DID Numbers? n
Custom Selection of VIP DID Numbers? n
Number of Digits from PMS:
PMS Sends Prefix? n
Number of Digits in PMS Coverage Path: 3
Digit to Insert/Delete:
5.3.4. Allow Housekeeping Status Updates
To allow housekeeping to change the housekeeping status of a guests room by dialing the
appropriate access code, Client Room must be enabled on the COS assigned to the SIP phone.
In this example, Client Room was enabled for COS 1, which was assigned to the Bittel SIP
phone.
change cos-group 1
CLASS OF SERVICE
Page
COS Group: 1
Auto Callback
Call Fwd-All Calls
Data Privacy
Priority Calling
Console Permissions
Off-hook Alert
Client Room
Restrict Call Fwd-Off Net
Call Forwarding Busy/DA
Personal Station Access (PSA)
Extended Forwarding All
Extended Forwarding B/DA
Trk-to-Trk Transfer Override
QSIG Call Offer Originations
Contact Closure Activation
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0
y
n
n
n
y
y
y
y
y
n
n
n
n
n
n
1
y
y
n
y
n
n
y
n
y
n
n
n
n
n
n
1 of
2
COS Name:
2
y
n
n
n
y
n
n
y
n
n
n
n
n
n
n
3
n
y
n
n
y
n
n
y
n
n
n
n
n
n
n
4
y
y
n
n
y
n
n
y
n
n
n
n
n
n
n
5
n
n
y
n
y
n
n
y
n
n
n
n
n
n
n
6
y
n
y
n
y
n
n
y
n
n
n
n
n
n
n
7
n
y
y
n
y
n
n
y
n
n
n
n
n
n
n
8
y
y
y
n
y
n
n
y
n
n
n
n
n
n
n
9 10 11 12 13 14 15
n y n y n y n
n n y y n n y
n n n n y y y
y y y y y y y
y n n n y y y
n n n n n n n
n n n n n n n
y y y y y y y
n n n n n n n
n n n n n n n
n n n n n n n
n n n n n n n
n n n n n n n
n n n n n n n
n y y y n n n
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6. Configure Avaya Aura® Session Manager
This section provides the procedures for configuring Session Manager. The procedures include
the following areas:
Launch System Manager
Administer SIP User
Note: It is assumed that basic configuration of Session Manager has already been performed.
This section will focus on the configuration of a SIP user for the Bittel UNO Voice SIP
Telephone.
6.1. Launch System Manager
Access the System Manager Web interface by using the URL “https://ip-address” in an Internet
browser window, where “ip-address” is the IP address of the System Manager server. Log in
using the appropriate credentials.
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6.2. Administer SIP User
In the subsequent screen (not shown), select Users User Management Manage Users to
display the User Management screen below. Click New to add a user.
6.2.1. Identity
The New User Profile screen is displayed. Enter desired Last Name and First Name. For
Login Name, enter “<ext>@<domain>”, where “<ext>” is the desired Bittel UNO Voice SIP
extension and “<domain>” is the applicable SIP domain name from Section 5.2. For Password
and Confirm Password, enter the appropriate credentials for System Manager. Retain the
default values in the remaining fields.
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6.2.2. Communication Profile
Select the Communication Profile tab. For Communication Profile Password and Confirm
Password, enter the desired password for the SIP user to use for registration.
In the Communication Address sub-section, click New to add a new entry. The
Communication Address sub-section is updated with additional fields, as shown below. For
Type, retain “Avaya SIP”. For Fully Qualified Address, enter and select the SIP user extension
and domain name to match the login name from Section 6.2.1. Click Add.
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Scroll down to check and expand Session Manager Profile. For Primary Session Manager,
Origination Application Sequence, Termination Application Sequence, and Home Location,
select the values corresponding to the applicable Session Manager and Communication Manager.
Retain the default values in the remaining fields.
Scroll down to check and expand CM Endpoint Profile. For System, select the value
corresponding to the applicable Communication Manager. For Extension, enter the SIP user
extension from Section 6.2.1. For Template, select DEFAULT_9620SIP_CM_6_3. For Port,
click and select IP. Retain the default values in the remaining fields. Click Commit to save the
configuration (not shown).
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In the CM Endpoint Profile sub-section, click the Endpoint Editor button to display the page
below. In the General Options tab, specify that coverage path that points to the voicemail
system in the Coverage Path 1 field. This provides voicemail coverage for the SIP user. In this
example, coverage path 20 was used.
In the Feature Options tab, set the MWI Served User Type field to sip-adjunct. This allows
MWI to be enabled for the SIP user. The voicemail system was connected via SIP to Session
Manager. Click Done when complete, followed by Commit on the previous page.
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7. Configure Bittel UNO Voice SIP Telephone
Access the Bittel UNO Voice web interface by using the URL “https://ip-address” in an Internet
browser window, where “ip-address” is the IP address of the SIP phone. Log in using the
appropriate credentials and then click Enter.
Note 1: To access the Bittel UNO Voice web interface initially, hold down the Mute button on
the SIP phone to hear the the phone’s IP address. Use this IP address to first access the
configuration screens. Afterwards, the IP address may be changed.
Note 2: When configuration changes are made to the Bittel UNO Voice SIP Telephone, the
phone will reboot.
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Select Network in the left pane and configure the SIP phone’s network settings as shown below.
In this example, a static IP is configured. Click OK.
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Navigate to the SIP Proxy webpage as shown below. Under the Basic SIP Proxy Settings
section, configure the following parameters.
SIP Registration:
SIP Proxy:
SIP Server Port:
SIP User ID:
SIP Authentication ID:
Set to Yes.
Set to the Session Manager IP address (e.g., 10.32.24.235).
Set to the SIP port the phone listens on (e.g., 5060).
Specify the User ID (e.g., 78010, the SIP extension).
Specify the SIP extension used to register with Session
Manager from Section 6.2.1 (e.g., 78010).
SIP Authentication PIN:Specify the Communication Profile Password configured
in Session Manager in Section 6.2.2. This password is
used for the SIP phone to register with Session Manager.
User Name:
Specify any user name for the SIP phone.
Under the Advanced SIP Settings section, enable Subscribe for MWI and set the Message
Service Number to the voicemail pilot number (e.g., 26000). Click OK.
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Navigate to the Voice webpage as shown below and configure the desired codecs to be supported
in the Voice Codec Settings section. In this example, PCMU was allowed. In the Dial Rules
Settings section, specify The time wait to dial out. In this example, the timer was set to 1
second. Using this setting, Bittel UNO Voice will wait 1 second for more digits and if not
received, it will dial the call. If a higher inter-digit timeout is required, increase this field. Click
OK.
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Navigate to the Phonebook webpage as shown below. This page is used to configure the
programmable button. In this example, the following programmable buttons were configured.
M9 (Operator button):
M2 (Wake Up button):
M1 (Messages):
This button was set to the operator extension number (e.g., 77301).
This button was configured to allow a guest to set the wake up
feature. In this example, 78119 is the FNE for Idle Appearance
Select. Dialing this FNE allows the guest to receive dial tone. The
‘p’ instructs the SIP phone to wait 1 seconds prior to dialing the
*70, which is the Automatic Wakeup Call access code configured
in Section 5.3.1. Recall that The time wait to dial out was to 1
second in the webpage above. After the Automatic Wakeup Call
access code is dialed, 3-burst confirmation is played, and the guest
should then specify the wakeup time in hhmm format, where hh is
the hour and mm are the minutes.
Specify the voicemail system pilot number (e.g., 26000). When
this button is pressed, the voicemail system will be dialed to allow
the guest to retrieve their messages.
Click OK when done.
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8. Verification Steps
This section provides the tests that can be performed to verify proper configuration of the Bittel
UNO Voice SIP Telephone with Avaya Aura® Communication Manager and Avaya Aura®
Session Manager.
1. Verify that the Bittel UNO Voice SIP Telephone has successfully registered with Session
Manager. In System Manager, navigate to Elements Session Manager System Status
User Registrations to check the registration status.
2. Verify basic telephony features by establishing calls between a Bittel UNO Voice SIP
Telephone with and another phone.
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9. Conclusion
These Application Notes have described the administration steps required to integrate the Bittel
UNO Voice SIP Telephone with Avaya Aura® Communication Manager and Avaya Aura®
Session Manager. The Bittel UNO Voice SIP Telephone with successfully registered with
Session Manager and basic telephony and hospitality features were verified. All test cases
passed with observations noted in Section 2.2.
10. References
This section references the Avaya documentation relevant to these Application Notes. The
following Avaya product documentation is available at http://support.avaya.com.
[1] Administering Avaya Aura® Communication Manager, Release 6.3, Issue 9, October 2013,
Document Number 03-300509.
[2] Administering Avaya Aura® Session Manager, Release 6.3, Issue 3, October 2013.
[3] Bittel User Manual Cover Models HA9888(66) TSD-IP, HA9888(67) TSD-IP, HS9888(68)
TSD-IP, HS9888(69) TSD-IP, Version V1.
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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.
JAO; Reviewed:
SPOC 1/17/2014
Solution & Interoperability Test Lab Application Notes
©2014 Avaya Inc. All Rights Reserved.
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