Avaya TAS FreiTel-IP Supplement

Avaya Solution & Interoperability Test Lab
Application Notes for TAS FreiTel-IP Handsfree Telephone
with Avaya Communication Manager Avaya SIP
Enablement Services – Issue 1.0
Abstract
These Application Notes describe the configuration of TAS FreiTel-IP handsfree telephone for
operation with Avaya Communication Manager. TAS FreiTel-IP is a SIP telephone intended
for use in environments where a handset and keypad are not required, such as elevators. These
Application Notes contain an extensive description of the configurations for both FreiTel-IP
and Avaya Communication Manager which were used for testing.
Information in these Application Notes has been obtained through 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
The FreiTel-IP handsfree telephone system consists of a SIP endpoint which can control up to
three speakerphones. This product is intended for use in environments where a handset and
keypad are not required, such as elevators.
The speakerphone is a surface-mounted unit which has a user interface which consists of a
button, two LEDs (green and yellow), and a microphone/speaker. Pressing the speakerphone
button causes a call to be made to a preconfigured telephone number, which can be terminated
only by the called party. The green LED on the speakerphone remains on while a call is active.
The FreiTel-IP can be configured to accept calls automatically or manually. If configured for
automatic call acceptance, every incoming call is accepted automatically without delay, and
remains active (indicated by the green LED being on) until terminated by the calling party. If the
unit is configured for manual call acceptance, an incoming call is signaled by the yellow light
and can be answered by pushing and holding the speakerphone button, which turns the yellow
LED off and the green LED on. A manually accepted call remains active until terminated by
releasing the speakerphone button, which turns off the green LED.
FreiTel-IP can be configured to filter incoming calls so unwanted calls are ignored.
The control unit has an Ethernet connector and a single multicolored LED. If attached to an
Ethernet network, the LED turns red until the controller registers with a SIP proxy, at which time
the LED turns green.
The configuration used for testing consisted of a TAS FreiTel-IP control unit and a single
speakerphone together with a pair of Avaya S8720 Server running Avaya Communication
Manager, Avaya G650 Media Gateway, Avaya SIP Enablement Services (SES) server, and
various Avaya telephones. Note that the SES server is a combined home/edge configuration
Avaya 4610
Telephone (
SIP)
192.168.60.X 192.168.150.X
D
PSTN
F
Avaya G650
Media Gateway
Avaya S8720 Server
PRI
TAS
FreiTel-IP
Avaya 2410
Digital Telephone
E
Avaya 4610
Telephone
(H.323)
B
A
Avaya 9640
Telephone
(H.323)
C
.202
192.168.200.X
.100
Avaya SES
CLAN: .6
Figure 1: FreiTel-IP Test Configuration
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The TAS FreiTel-IP handsfree telephone is attached to Avaya Communication Manager via SIP
trunk by way of the Avaya SIP Enablement Services (SES).
The following table contains additional information about each of the telephones contained in the
above diagram:
Diagram Ext PSTN Number Endpoint
A 69001 069 9073 9887 69001 TAS FreiTel-IP
B 60116 069 9073 9887 60116 Avaya 4610 IP
C 60093 069 9073 9887 60093 Avaya 9640 one-X
Deskphone
D 60113 069 9073 9887 60113 Avaya 4610 SIP
E 60007 069 9073 9887 60007 Avaya 2410 Digital
F 069 7505 6176 ISDN telephone
Table 1: Extensions Used for Testing
2. Equipment and Software Validated
Equipment Software Version
Avaya S8720 Server Avaya Communication Manager 4.0.1
(R014x.00.1.731.2)
Avaya G650 Media Gateway
TN2312BP IP Server Interface
TN799DP CLAN Interface
TN2302AP IP Media Processor
HW11 FW040
HW01 FW024
HW20 FW117
Avaya SIP Enablement Services Server SES-4.0.0.0-0.33.6.
Avaya 2410 Digital Telephone 5.0
Avaya 4610 H.323 Telephone 2.8
Avaya 4610 SIP Telephone 2.2.2
Avaya 9640 H.323 Telephone 1.5
TAS FreiTel-IP V01.00
Table 2: Hardware/Software Component Versions
3. Configuration
3.1. Configure Avaya Communication Manager
The configuration and verification operations illustrated in this section were performed using the
Avaya Communication Manager System Administration Terminal (SAT) terminal.
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3.1.1. Verify system-parameters customer-options
Use the display system-parameters customer-options command to verify that Avaya
Communication Manager is configured to meet the minimum requirements to support the
configuration used for these tests. Those items shown in bold indicate required values or
minimum capacity requirements. If these are not met in the configuration, please contact an
Avaya representative for further assistance.
Verify that the number of SIP trunks supported by the system is sufficient for the combination of
trunks to the TAS FreiTel-IP and optional SIP endpoints to be supported.
display system-parameters customer-options Page 2 of 11
OPTIONAL FEATURES
IP PORT CAPACITIES USED
Maximum Administered H.323 Trunks: 10 0
Maximum Concurrently Registered IP Stations: 50 10
Maximum Administered Remote Office Trunks: 0 0
Maximum Concurrently Registered Remote Office Stations: 0 0
Maximum Concurrently Registered IP eCons: 0 0
Max Concur Registered Unauthenticated H.323 Stations: 0 0
Maximum Video Capable H.323 Stations: 0 0
Maximum Video Capable IP Softphones: 0 0
Maximum Administered SIP Trunks: 200 20
Maximum Number of DS1 Boards with Echo Cancellation: 0 0
Maximum TN2501 VAL Boards: 1 0
Maximum G250/G350/G700 VAL Sources: 0 0
Maximum TN2602 Boards with 80 VoIP Channels: 0 0
Maximum TN2602 Boards with 320 VoIP Channels: 0 0
Maximum Number of Expanded Meet-me Conference Ports: 0 0
Figure 2: System-Parameters Customer-Options Form
3.1.2. Configure Dial Plan
3.1.2.1 Configure Dial Plan Analysis
Use the change dialplan analysis command to specify that dialed strings which begin with “6”
are extensions. Include the string “*83” to be used as trunk access code for the SIP trunk as
described in section 3.1.3.3.
change dialplan analysis Page 1 of 12
DIAL PLAN ANALYSIS TABLE
Percent Full: 1
Dialed Total Call Dialed Total Call Dialed Total Call
String Length Type String Length Type String Length Type
6 5 ext
*83 3 dac
Figure 3: Dialplan Analysis Form
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3.1.3. Configure Interface to SES
3.1.3.1 Specify IP node names
Use the change node-names ip command to define the address of the “clan” interface and the
Avaya SIP Enablement Services server.
change node-names ip Page 1 of 1
IP NODE NAMES
Name IP Address Name IP Address
clan 192.168.60.6 . . .
default 0 .0 .0 .0 . . .
ipsi 192.168.60.5 . . .
medpro 192.168.60.7 . . .
procr . . . . . .
ses 192.168.200.100 . . .
Figure 4: Node-Names IP Form
3.1.3.2 Configure Signaling Group for the SIP Trunk Interface to SES
Use the add signaling-group <x> command, where <x> is a free signaling group number, to
create a signaling group which is to be used to connect to the SES. Accept defaults for
parameters, except for those which are highlighted.
Parameter Usage
Group Type Enter “sip” to specify a SIP trunk.
Transport Method Enter “tls” to specify that Transport Layer Security should be used to
encode data information flow on this signaling group.
Near-end Node Name Enter “clan” to use the CLAN interface on the S8500, which was
assigned in Figure 4
Near-end Listen Port Accept the default of “5061” to specify the standard TLS listen port.
Far-end Node Name Enter “ses” to specify the SES server name assigned in Figure 4.
Far-end Listen Port Accept the default of “5061” to specify the standard TLS listen port.
Far-end Domain Enter the domain name which is configured for SES, configured in
Figure 13.
Direct IP-IP Audio
Connections Enter “y” to specify that direct IP-IP audio connections should be used.
Table 3: Configuration Signaling Group for SIP Interface to SES
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add signaling-group 83 Page 1 of 1
SIGNALING GROUP
Group Number: 83 Group Type: sip
Transport Method: tls
Near-end Node Name: clan Far-end Node Name: ses
Near-end Listen Port: 5061 Far-end Listen Port: 5061
Far-end Network Region:
Far-end Domain: ffm.com
Bypass If IP Threshold Exceeded? n
DTMF over IP: in-band-g711 Direct IP-IP Audio Connections? y
IP Audio Hairpinning? n
Session Establishment Timer(min): 120
Figure 5: SIP Signaling-Group Form
3.1.3.3 Configure Interface to SIP Trunk
Use the add trunk-group <x> command, where <x> is a free trunk group number, to create a
trunk group which is to be used to connect to the Avaya SES. Accept defaults for parameters,
except for those which are highlighted.
Parameter Usage
Group Type Specify a type of “sip”.
TAC Set the Trunk Access Code to “*83”.
Group Name Specify “SIP” to identify this trunk. Any identifier can be used.
Service Type Specify the trunk is used as a “tie” line to another PBX.
Signaling Group Specify the signaling group which was configured for the sip trunk in
Figure 5.
Number of Members Specify a value sufficient for the maximum number of IP connections to
be allowed via this trunk.
Table 4: Configuration Parameters for Trunk Interface to SIP Trunk
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add trunk-group 83 Page 1 of 21
TRUNK GROUP
Group Number: 83 Group Type: sip CDR Reports: y
Group Name: SIP COR: 1 TN: 1 TAC: *83
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Signaling Group: 83
Number of Members: 5
Figure 6: SIP Trunk-Group Form
3.1.3.4 Configure Network Region
Use the change network-region <x> command, where <x> is the network region used by the
SIP trunk. Enter the following parameters:
Parameter Usage
Location Use a location of “1”, in this example.
Authoritative Domain Use a domain of “ffm.com”, as configured for Avaya SES in Figure 13.
Name Assign a name for identification purposes.
Intra-region IP-IP
Direct Audio Specify “y” to allow direct connections between IP endpoints.
Table 5: Configuration Parameters for Network Region
change ip-network-region 1 Page 1 of 19
IP NETWORK REGION
Region: 1
Location: 1 Authoritative Domain: ffm.com
Name: FFM
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 1 Inter-region IP-IP Direct Audio: no
UDP Port Min: 2048 IP Audio Hairpinning? n
UDP Port Max: 3327
DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y
Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46 Use Default Server Parameters? y
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
Figure 7: IP-Network-Region Form
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3.1.3.5 Configure Codec Set
Use the change ip-codec-set <x> command, where <x> is the codec set assigned to the network
region used by the SIP trunk. Enter the following parameters:
Parameter Usage
Audio Codec Enter “G.711A” to specify the use of the G711 A-Law codec.
Table 6: Configuration Parameters for Trunk Interface to SES
change ip-codec-set 1 Page 1 of 2
IP Codec Set
Codec Set: 1
Audio Silence Frames Packet
Codec Suppression Per Pkt Size(ms)
1: G.711A n 2 20
2:
3:
4:
5:
6:
7:
Figure 8: IP-Codec-Set Form
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3.1.4. Configure Station for FreiTel-IP
Use the add station <x> command to allocate a station for FreiTel-IP, where <x> is the
extension for FreiTel-IP shown in Table 1.
Parameter Usage
Type Enter the model identification of the phone to be used as shown in Table 1. Note that the TAS FreiTel telephone is configured as an
Avaya 4610 IP Telephone.
Name Enter the name of the user which is to be associated with the phone.
Security Code Enter the security code assigned to the extension.
Table 7: Configuration Parameters IP Telephones
change station 69001 Page 1 of 4
STATION
Extension: 69001 Lock Messages? n BCC: 0
Type: 4610 Security Code: xxxxxxx TN: 1
Port: S00009 Coverage Path 1: COR: 1
Name: ext 69001 Coverage Path 2: COS: 1
Hunt-to Station:
STATION OPTIONS
Loss Group: 19 Personalized Ringing Pattern: 1
Message Lamp Ext: 69001
Speakerphone: 2-way Mute Button Enabled? y
Display Language: english
Survivable GK Node Name:
Survivable COR: internal Media Complex Ext:
Survivable Trunk Dest? y IP SoftPhone? n
Customizable Labels? y
Figure 9: Station Form
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3.1.5. Configure Interface to Avaya SES for FreiTel-IP
Use the change off-pbx-telephone station-mapping command to configure an interface to SES
for the FreiTel-IP. Assign values for this command as shown in the following table.
Parameter Usage
Station Extension Enter the extension FreiTel-IP from Table 1.
Application Enter “OPS”.
Phone Number Enter the extension FreiTel-IP from Table 1.
Trunk Selection Enter the number assigned to the SIP trunk group in Figure 6.
Table 8: Parameters for Off-PBX-Telephone Station-Mapping
change off-pbx-telephone station-mapping 69001 Page 1 of 2
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station Application Dial CC Phone Number Trunk Config
Extension Prefix Selection Set
69001 OPS - 69001 83 1
Figure 10: Off-PBX-Telephone Form, Page 1
3.2. Configure Avaya SIP Enablement Services
Log in to the Avaya SES Web-based Integrated Management tool by selecting the IP address of
the Avaya SES server followed by “/admin” from the Web browser. After entering the login ID
and password, select “Launch Administration Web Interface”.
Figure 11: SES Initial Greeting Screen
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The SES Integrated Management top level menu is then displayed.
Figure 12: SES Integrated Management Top Level Menu
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3.2.1. Configure Basic SES Parameters
From the top-level management screen, select “Server Configuration” -> “System Properties”.
Enter the name to be assigned to the “SIP Domain” that was assigned in Figure 5, and the IP
address of the SES server which was assigned in Figure 4 as the IP address of the “License
Host”. Select the “Update” button.
Figure 13: Avaya SES Edit System Properties Screen
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From the top-level management screen, click “Manage Hosts” -> “Add Host”.
Enter the Host IP Address of the Avaya SES Server, a DB password, and a Profile Service
Password that were allocated to the Avaya SES server when it was installed. Leave the other
fields assigned to their respective default values. Select the “Update” button. Note that this is a
combined home/edge configuration.
Figure 14: Avaya SES “Add Host” Screen
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3.2.2. Configure Interface to Avaya Communication Manager
From the top-level management screen, select “Manage Media Servers”-> “Add Media Server”.
Assign a meaningful name to the “Media Server Interface Name”. Select the IP address of the
Avaya SES server from the “Host” drop-down box. Enter the address of the Avaya S8720
CLAN interface as the SIP Trunk IP Address. Select the “Add” button when these parameters
have been entered.
Figure 15: Avaya SES Add Media Server Interface Screen
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3.2.3. Configure SIP Endpoint for FreiTel-IP
From the top level menu, select the “Manage Users” -> “Add User” menu entries. Enter the
extension for FreiTel-IP as both the “Primary Handle” and the “User ID”. This is the same
extension that was configured for the station in Figure 9 and for the off-pbx-telephone station-
mapping in Figure 10. Enter a Password and First/Last name of the user, check the “Add
Media Server Extension” box, and click “Add”.
Figure 16: Avaya SES “Add User” Screen
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Enter the Media Server Extension for the User ID 69001 (the extension of the Avaya SIP
telephone). Select the Media Server (refer to Figure 15) from the drop-down box and click
“Add” to continue.
Figure 17: Avaya SES Add Media Server Extension Screen
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From the main menu, click the “Update” control in the left frame to commit the changes made.
Figure 18: Update from Top SES Screen
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3.3. Configure FreiTel-IP
The TAS FreiTel-IP is configured via a terminal attached to the unit’s serial maintenance
interface. From this terminal, it is also possible to temporarily enable a telnet interface, so that
further configuration steps can be performed via a telnet session. However, the telnet interface
should be disabled at completion of the configuration session for security reasons.
The screen shown below is presented after initiation of a telnet session to FreiTel-IP. Use the key
codes shown in the follow table to configure the parameters which must be changed from the
default values. The “Designation” column shows the designation for each of the parameters
shown in Figure 20.
Key Usage Designation Value
I IP Address IP Ad. Enter the IP address to be assigned to FreiTel-IP, in this
case 192.168.150.202
M IP Mask IP Mask. Enter the IP mask used by the network to which the
FreiTel-IP is attached, in this case “255.255.255.0”.
R Default router Def Router Adr. Enter the default gateway used by the network to which the FreiTel-IP is attached, in this case
“192.168.150.254”.
a Call destination RufNr. Zeil- Taste 1 Enter one of the telephone numbers from Table 1 to designate the destination which is to be called when the
“call” button on the speakerphone is pressed.
3 SIP user SIP User Enter the extension which is to be used by FreiTel-IP.
4 SIP domain SIP Domain Enter the domain which is to be used by FreiTel-IP. This must be the same value as used in Figure 5 and
Figure 7 .
5 Authorized
user Auth. User Enter the extension which is to be used by FreiTel-IP.
6 Authorized
password Auth. Password Enter the password assigned in Figure 9 and Figure 16.
7 SIP proxy SIP Proxy Enter “192.168.200.100” the address of the Avaya SES
server, as shown in Figure 4 and Figure 14.
9 Display
name Displ. Name Enter a descriptive name to be used for calls made by FreiTel-IP.
Table 9: FreiTel-IP Configuration Parameters
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Upon completion, click “k” to show to the configuration screen so that the parameters which
were changed can be verified.
I -IP Adress
M -IP MASK
R -Default Router
a - Zieltaste 1 Rufnummer: 600007
b - Zieltaste 2 Rufnummer:
c - Zieltaste 3 Rufnummer:
O - Option = 0 ( 1=Trenn-Taste, 2=Annahme-Taste)
o - Option Alarmoutput = $10
A - Audio-Modus (Echounterdrueckung)=0
P - Pegelschwelle EIN fuer Audiomode 7
p - Pegelschwelle Aus fuer Audiomode 7
E - Echounterdrⁿckungsfaktor
t - Audio Test Lock
h - Lauth÷rer 1-3 Vol.m - Mic. 1-3 Vol.
1 -Rtp First Port
2 -Rtp Last Port
3 -SIP User
4 -SIP Domain
5 -Auth. User
6 -Auth. Password
7 -SIP Proxy
8 -Stun Server
9 -Displ Name
0 -DNS Server
f -CLI Filter-Nr. je Sp. setzen
F -CLI Filter setzen
k -Show Konfig.
s -Status
D -Debug
T - Temperatur Init
w - Audio Test manuell
u -Update Url
l -Logout
Figure 19: FreiTel-IP Introductory Screen
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----------------- IP ----------------------
IP Adr. 192.168.150.202
IP Mask. 255.255.255.0
Def Router Adr. 192.168.150.254
Update URL: it
---------------- SIP ----------------------
Rtp First Port:
Rtp Last Port:
SIP User: 69001
SIP Domain: ffm.com
Auth. User: 69001
Auth. Password: xxxxx
SIP Proxy: 192.168.200.100
Stun Server:
Displ.Name: 69001
DNSServer:
---------------- Applikation --------------
CLI Filter[1]:
CLI Filter[2]:
CLI Filter[3]:
Filter Nr[1]:
Filter Nr[2]:
Filter Nr[3]:
Filter Nr[4]:
Filter Nr[5]:
Filter Nr[6]:
Filter Nr[7]:
Filter Nr[8]:
RufNr.Ziel-Taste 1 = 60007
RufNr.Ziel-Taste 2 =
RufNr.Ziel-Taste 3 = 0k
---------------- Audio --------------------
Lautspr.1 Vol (0-63)= 60
Lautspr.2 Vol (0-63) = 60
Lautspr.3 Vol (0-63) = 60
Mic.1 Vol (0-63) = 63
Mic.2 Vol (0-63) = 59
Mic.3 Vol (0-63) = 59
Audio Test Lock 1,2,4
Audio Mode 0
Echo Absenkfaktor 0 0 0
Echo Pegelschwelle-Ein in Mode 7: 0 0 0
Echo Pegelschwelle-Aus in Mode 7: 0 0 0
Option: 3 Alarm Option: 10
Figure 20: FreiTel-IP Configuration Screen
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