Siemens Hipath 4000

Avaya Solution & Interoperability Test Lab
Configuring Siemens HiPath 4000 V5 with Avaya Meeting

Exchange Enterprise Edition R5.2 using Avaya Aura
Communication Manager R6.0.1 and Avaya Aura
Session Manager R6.1 Issue 1.0
Abstract
These Application Notes present a sample configuration for a network consisting of a
Siemens HiPath 4000 V5 and Avaya Meeting Exchange Enterprise Edition R5.2. These
two systems are connected via a SIP trunk using Avaya Aura Communication Manager
R6.0.1 and Avaya Aura Session Manager R6.1.
Testing was conducted via the Internal Interoperability Program at the Avaya Solution and
Interoperability Test Lab.
BD; Reviewed:
SPOC 08/12/2011
Solution & Interoperability Test Lab Application Notes

2011 Avaya Inc. All Rights Reserved.
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Table of Contents
1. INTRODUCTION....................................................................................................................4
1.1.
EquipmentandSoftwareValidated....................................................................................5
2. CONFIGUREAVAYAAURACOMMUNICATIONMANAGER.......................................6
2.1.
2.2.
2.3.
2.4.
2.5.
2.6.
2.7.
2.8.
2.9.
VerifyAvayaAuraCommunicationManagerLicense.........................................................6
AdministerSystemParametersFeatures............................................................................7
AdministerIPNodeNames................................................................................................7
AdministerIPNetworkRegionandCodecSet....................................................................8
AdministerSIPSignalingGroupandTrunkGroup...............................................................9
AdministerRoutePattern................................................................................................11
AdministerPrivateNumbering.........................................................................................11
AdministerDialPlanandAARanalysis.............................................................................11
SaveChanges...................................................................................................................12
3. CONFIGURINGAVAYAAURASESSIONMANAGER...................................................13
3.1.
3.2.
3.3.
3.4.
3.5.
3.6.
3.7.
3.8.
3.9.
3.10.
3.11.
3.12.
LogintoAvayaAura SystemManager...........................................................................13
AdministerSIPDomain....................................................................................................15
AdministerLocations.......................................................................................................15
AdministerAdaptations...................................................................................................16
AdministerSIPEntities.....................................................................................................17
AdministerEntityLinks....................................................................................................20
AdministerTimeRanges..................................................................................................20
AdministerRoutingPolicies.............................................................................................21
AdministerDialPatterns..................................................................................................22
AdministerAvayaAuraSessionManager.......................................................................26
AddAvayaAuraCommunicationManagerasanEvolutionServer..................................27
AddUsersforSIPPhones.................................................................................................32
4. CONFIGURESIEMENSHIPATH4000V5......................................................................36
4.1.
4.2.
4.3.
4.4.
4.5.
4.6.
4.7.
4.8.
4.9.
4.10.
4.11.
4.12.
4.13.
InstallandConfiguretheSiemensHiPath4000V5Assistant.............................................37
ClientPCPreparation.......................................................................................................37
HiPath4000SystemConfiguration...................................................................................43
NetworkDomainConfiguration.......................................................................................45
NetworkSystemConfiguration........................................................................................46
ReviewofBoardConfiguration........................................................................................47
SiemensStationEndPoints.............................................................................................50
HG3500GatewayConfiguration......................................................................................53
SIPTrunkGateway...........................................................................................................55
SIPRegistrarGateway......................................................................................................57
H.323Gateway................................................................................................................58
TrunksandTrunkGroups.................................................................................................59
LeastCostRouting...........................................................................................................61
5. CONFIGUREAVAYAMEETINGEXCHANGEENTERPRISE.....................................68
5.1.
5.2.
5.3.
5.4.
ConfiguringSIPConnectivity............................................................................................68
ConfigureDialout............................................................................................................69
MapDNISEntries.............................................................................................................70
ConfigureAudioPreferencesfile......................................................................................71
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5.5.
5.6.
RestartingtheMeetingExchangeServer..........................................................................71
AvayaBridgeTalk............................................................................................................72
6. VERIFICATION.....................................................................................................................76
6.1.
6.2.
6.3.
VerifySiemensHiPath4000V5........................................................................................77
VerifyAvayaMeetingExchange Enterprise.................................................................79
VerifiedScenarios............................................................................................................81
7. CONCLUSION........................................................................................................................81
8. ADDITIONALREFERENCES..............................................................................................81
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1. Introduction
The purpose of this interoperability application note is to validate Siemens HiPath 4000 V5
with Avaya Meeting Exchange Enterprise Edition (MX). The sample network is shown in
Figure 1, where the Siemens HiPath 4000 V5 supports the Siemens OpenStage and Optipoint
SIP and H.323 devices. A SIP trunk is used to connect Siemens HiPath 4000 V5 and Avaya
Meeting Exchange Enterprise via Avaya Aura Session Manager. All inter-system calls
are carried over this SIP trunk. Siemens devices are registered to Siemens HiPath 4000 SIP
Gateway (HG3500) .
Figure 1: Connection of Siemens HiPath 4000 V5 and Avaya Meeting Exchange

Enterprise Edition via a SIP trunk.
Dial Patterns used during testing
Number
38888 / 638888
38111 / 638111
38222 / 628222
23XXX
810XXX
BD; Reviewed:
SPOC 08/12/2011
Destination
SCAN Conference Access from Avaya / Siemens handsets
FLEX Conference Access from Avaya / Siemens handsets
DIRECT Conference Access from Avaya / Siemens handsets
Avaya handset number range
Siemens handset number range

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1.1. Equipment and Software Validated

The following equipment and software/firmware were used for the sample configuration.
Equipment
Avaya S8800 Media Server
Avaya G650 Media Gateway
Avaya Handset 9621G

Avaya Flare Experience
A175D01A-003
Siemens HiPath 4000
Siemens SIP Gateway HG3500
Siemens SIP Registrar HG3500
Siemens H.323 Gateway HG3500
Siemens OptiPoint 420Std
Siemens OptiPoint 420 Std S
Siemens OptiPoint OpenStage 20 S
BD; Reviewed:
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Software
Avaya Aura Communication Manager R6.0.1
Version R016x.00.1.510.1
Avaya Aura System Manager R6.1
(Build No. 6.1.0.0.7345-6.1.5.106)
Avaya Aura Session Manager R6.1
(Build No. - 6.1.1.0.611004-6.1.2.612005)
TN2312BP HW15FW049
TN2602AP HW08FW049
TN799DP HW01 FW034
(9620SIP)Firmware: V6.0
(SIP) V1.0.3
Siemens HiPath 4000 V5 R0.5.28
L0-T2R.51.000-004 /pzksti40.26.000-004
L0-T2R.51.000-004 /pzksti40.26.000-004
L0-T2R.51.000-004 /pzksti40.26.000-004
Firmware: V5 R6.3.0
Firmware: V7 R5.7.0
Firmware: V2 R1.21.0

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2. Configure Avaya Aura Communication Manager

This section shows the configuration in Communication Manager. All configurations in this
section are administered using the System Access Terminal (SAT). These Application Notes
assumed that the basic configuration has already been administered. For further information
on Communication Manager, please consult with references [1] and [2]. The procedures
include the following areas:
Verify Avaya Aura Communication Manager License
Administer System Parameters Features
Administer IP Node Names
Administer IP Network Region and Codec Set
Administer SIP Signaling Group and Trunk Group
Administer Route Pattern
Administer Private Numbering
Administer Locations
Administer Dial Plan and AAR analysis
Save Changes
2.1. Verify Avaya Aura Communication Manager License

Use the display system-parameter customer options command to compare the Maximum
Administered SIP Trunks field value with the corresponding value in the USED column.
The difference between the two values needs to be greater than or equal to the desired
number of simultaneous SIP trunk connections.
Note: The license file installed on the system controls the maximum features permitted. If
there is insufficient capacity or a required feature is not enabled, contact an authorized Avaya
sales representative to make the appropriate changes.
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
18000
18000
24000
24000
522
128
250
128
128
300
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USED
0
1
0
0
0
0
0
3
10
0
0
0
0
0
1
0
(NOTE: You must logoff & login to effect the permission changes.)
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2.2. Administer System Parameters Features

Use the change system-parameters features command to allow for trunk-to-trunk transfers.
This feature is needed to allow for transferring an incoming/outgoing call from/to a remote
switch back out to the same or different switch. For simplicity, the Trunk-to-Trunk
Transfer field was set to all to enable all trunk-to-trunk transfers on a system wide basis.
Note: This feature poses significant security risk and must be used with caution. As an
alternative, the trunk-to-trunk feature can be implemented using Class Of Restriction or Class
Of Service levels.
change system-parameters features
Page
FEATURE-RELATED SYSTEM PARAMETERS
Self Station Display Enabled? y
Trunk-to-Trunk Transfer: all
Automatic Callback with Called Party Queuing? n
Automatic Callback - No Answer Timeout Interval (rings): 3
Call Park Timeout Interval (minutes): 10
Off-Premises Tone Detect Timeout Interval (seconds): 20
AAR/ARS Dial Tone Required? y
Music (or Silence) on Transferred Trunk Calls?
DID/Tie/ISDN/SIP Intercept Treatment:
Internal Auto-Answer of Attd-Extended/Transferred Calls:
Automatic Circuit Assurance (ACA) Enabled?
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19
no
attd
transferred
n
2.3. Administer IP Node Names

Use the change node-names ip command to add entries for Communication Manager and
Session Manager that will be used for connectivity. In the sample network, clan and
192.168.81.104 are entered as Name and IP Address for the CLAN card in Communication
Manager running on the Avaya S8800 Server. In addition, sm100 and 192.168.81.121 are
entered for Session Manager.
change node-names ip
Page
1 of
IP NODE NAMES
Name
clan
default
gateway
medpro
procr
procr6
sm100b
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IP Address
192.168.81.104
0.0.0.0
192.168.81.254
192.168.81.105
192.168.81.102
::
192.168.81.121

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2.4. Administer IP Network Region and Codec Set

Use the change ip-network-region n command, where n is the network region number to
configure the network region being used. In the sample network, ip-network-region 1 is
used. For the Authoritative Domain field, enter the SIP domain name configured for this
enterprise and a descriptive Name for this ip-network-region. Set Intra-region IP-IP Direct
Audio and Inter-region IP-IP Direct Audio to yes to allow for direct media between
endpoints. Set the Codec Set to 1 to use ip-codec-set 1.
change ip-network-region 1
Page
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20
IP NETWORK REGION
Region: 1
Location: 1
Authoritative Domain: mmsil.local
Name: To ASM61
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: 65535
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
Use the change ip-codec-set n command, where n is the existing codec set number to
configure the desired audio codec.
change ip-codec-set 1
Page
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IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
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Silence
Suppression
n
Frames
Per Pkt
2
Packet
Size(ms)
20

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2.5. Administer SIP Signaling Group and Trunk Group

2.5.1. SIP Signaling Group
In the test configuration, Communication Manager acts as an Evolution Server. An IMS
enabled SIP trunk is not required. Use signaling group 150 along with trunk group 150 to
reach the Session Manager. Use the add signaling-group n command, where n is the
signaling-group number being added to the system. Use the values defined in Section 2.3
and 2.4 for Near-end Node Name, Far-End Node-Name and Far-End Network Region.
The Far-end Domain is left blank so that the signaling group accepts any authoritative
domain. Set IMS Enabled to n and Peer Detection Enabled to y.
change signaling-group 150
Page
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SIGNALING GROUP
Group Number: 150
Group Type:
IMS Enabled? n
Transport Method:
Q-SIP? n
IP Video? y
Priority Video?
Peer Detection Enabled? y Peer Server:
Near-end Node Name: clan

Near-end Listen Port: 5060
sip
tcp
y
SM
SIP Enabled LSP? n

Enforce SIPS URI for SRTP? y
Far-end Node Name: sm100b

Far-end Listen Port: 5060
Far-end Network Region: 1
Far-end Domain:
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
BD; Reviewed:
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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):

n
n
y
n
n
6
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2.5.2. SIP Trunk Group

Use the add trunk-group n command, where n is the new trunk group number being added
to the system. The following screens show the settings used for trunk group 150. Enter the
following:
Group Type
sip
TAC
a numeric value i.e. 150
Service Type
tie
Signaling Group
the signaling group defined in Section 2.5.1, i.e. 150
Number of Members set to a numeric value, i.e. 10
change trunk-group 150
Page
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21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
150
sip tg 150
two-way
n
0
tie
Group Type: sip

CDR Reports: y
COR: 1
TN: 1
TAC: 150
Outgoing Display? y
Night Service:
Auth Code? n
Member Assignment Method: auto
Signaling Group: 150
Number of Members: 10
Navigate to Page 3 and enter private for Numbering Format.

TRUNK FEATURES
ACA Assignment? n
Page
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21
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
Navigate to Page 4 and enter 97 for Telephone Event Payload Type and P-Asserted
Identity for Identity for Calling Party Display. The number shown on the phone callee
display will be the number stated in the SIP PAI header.
Page
PROTOCOL VARIATIONS
Mark Users as Phone? n
Prepend '+' to Calling Number? n
Send Transferring Party Information? y
Network Call Redirection? n
Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type: 97
Convert 180 to 183 for Early Media? n
Always Use re-INVITE for Display Updates? n
Identity for Calling Party Display: P-Asserted-Identity
Enable Q-SIP? n
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2.6. Administer Route Pattern

Configure a route pattern to correspond to the newly added SIP trunk group. Use the change
route-pattern n command, where n is the route pattern number specified in Section 2.9.
Configure this route pattern to route calls to trunk group number 150 configured in Section
2.5.2. Assign the lowest FRL (facility restriction level) to allow all callers to use this route
pattern. Assign 0 to No. Del Dgts.
change route-pattern 150
Page
Pattern Number: 150 Pattern Name: To SessMan
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
No
Mrk Lmt List Del Digits
Dgts
1: 150 0
0
2:
3:
4:
5:
6:
BCC VALUE TSC CA-TSC
0 1 2 M 4 W
Request
1: y y y y y n
2: y y y y y n
n
n
1 of
DCS/
QSIG
Intw
n
n
n
n
n
n
IXC
user
user
user
user
user
user
ITC BCIE Service/Feature PARM
No. Numbering LAR

Dgts Format
Subaddress
none
none
unre
rest
2.7. Administer Private Numbering

Use the change private-numbering command to define the calling party number to be sent
out through the SIP trunk. In the sample network configuration below, all calls originating
from a 5-digit extension beginning with 37 will result in a 5-digit calling number. For an
outbound call, the calling party number will be in the From header of the SIP INVITE,
change private-numbering 0
Page
1 of
NUMBERING - PRIVATE FORMAT

Ext
Len
5
5
5
Ext
Code
23
37
38
Trk
Grp(s)
150
150
150
Private
Prefix
Total
Len
5
Total Administered: 3
5
Maximum Entries: 540
5
2.8. Administer Dial Plan and AAR analysis

Configure the dial plan for dialing a 5-digit conference number. Use the change dialplan
analysis command to define Dialed String 37 as an aar Call Type.
change dialplan analysis
Page
DIAL PLAN ANALYSIS TABLE
Location: all
Dialed
String
1
230
231
Total Call
Length Type
3
dac
5
ext
5
ext
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Dialed
String
Total Call
Length Type
1 of
12
Percent Full: 1
Dialed
String
Total Call
Length Type

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232
233
37
38
81
*
5
5
5
5
6
2
ext
ext
aar
aar
aar
fac
Use the change aar analysis 0 command to configure an aar entry for Dialed String 37 to
use Route Pattern 150. Add an entry for the SIP phone extensions which begin with 230 or
231. Use unku for call type.
change aar analysis 0
Page
AAR DIGIT ANALYSIS TABLE
Location: all
Dialed
String
Total
Min Max
5
5
5
5
7
7
5
5
5
5
7
7
230
231
3
37
38
4
Route
Pattern
150
150
999
150
150
999
Call
Type
unku
unku
aar
unku
unku
aar
Node
Num
1 of
Percent Full: 1
ANI
Reqd
n
n
n
n
n
n
2.9. Save Changes

Use the save translation command to save all changes.
save translation
SAVE TRANSLATION
Command Completion Status
Error Code
Success
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3. Configuring Avaya Aura Session Manager

This section provides the procedures for configuring Session Manager. For further
information on Session Manager, please consult with references [3]and [4]. The procedures
include the following areas:
Log in to Avaya Aura System Manager
Administer SIP domain
Administer Locations
Administer Adaptations
Administer SIP Entities
Administer Entity Links
Administer Time Ranges
Administer Routing Policies
Administer Dial Patterns
Administer Avaya Aura Session Manager
Add Avaya Aura Communication Manager as an Evolution Server
Administer SIP users
3.1. Log in to Avaya Aura System Manager

Access the System Manager using a Web Browser and entering http://<ip-address>/SMGR,
where <ip-address> is the IP address of System Manager. Log in using appropriate
credentials.
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In the next screen under Elements column select Routing.
In the main panel, a short procedure for configuring Network Routing Policy is shown.
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3.2. Administer SIP Domain

Add the SIP domain, for which the communications infrastructure will be authoritative, by
selecting Domains on the left panel menu and clicking the New button (not shown) to create
a new SIP domain entry. Complete the following options:
Name
The authoritative domain name (e.g., mmsil.local)
Type
SIP
Notes
Description for the domain (optional)
Click Commit to save changes.
3.3. Administer Locations

Session Manager uses the origination location to determine which dial patterns to look at
when routing the call if there are dial patterns administered for specific locations. Locations
are also used to limit the number of calls coming out of or going to a location. This is useful
for those locations where the network bandwidth is limited. To add a Location, select
Locations on the left panel menu and then click the New button (not shown). Enter the
following for each Location:
Under General:
Name
An informative name (e.g. Dublin)
Scroll down the screen to the Location Patterns area
Under Location Pattern:
IP Address Pattern An IP address pattern for this location
Select Add to add more locations. Click Commit to save changes.
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Location Patterns
3.4. Administer Adaptations

Create an adaptation entry for a call to Siemens adaptation, enter the following information:
Adaptation name
An informative name for the adaptation
Module name
Select DigitConversionAdapter
Digit Conversion for Incoming Calls to SM
Matching Pattern 6 with a minimum and maximum of 6
digits long, which is the dial pattern for a station registered
with Avaya which Siemens will be dialing. Delete Digits has
value 1 to indicate the first digit should be removed before
incoming call dials the relevant Avaya extension (.i.e.
623202 becomes 23202, a registered SIP device on the
Avaya).
Digit Conversion for Outgoing Calls from SM
Similarly, when an Avaya extension (which starts with 2 and
is 5 digits long) calls a device on the Siemens, a 6 is inserted
in front of the extension number before being connected with
the Siemens.
Click Commit to save changes.
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3.5. Administer SIP Entities

A SIP Entity must be added for Session Manager and for each SIP-based telephony system
supported by a SIP Trunk. To add a SIP Entity, select SIP Entities on the left panel menu
and then click on the New button (not shown). Enter the following for each SIP Entity:
Under General:
Name
An informative name (e.g., SessionManager)
FQDN or IP Address IP address of the signaling interface on the Session Manager,
Communication Manager, Meeting Exchange or Siemens
HiPath 4000 V5.
Type
Session Manager for Session Manager, CM for
Communication Manager and SIP Trunk for Siemens
HiPath 4000 V5 and Meeting Exchange.
Location
Galway
Time Zone
Time zone for this location
The following screen shows the SIP Entity for Session Manager.
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Scroll down the screen to the area Port. Under Port, click Add, and then edit the fields in the
resulting new row.
Port
Port number on which the system listens for SIP requests
Protocol
Transport protocol to be used to send SIP requests
The following screen shows the Port definitions for the Session Manager SIP Entity. Click
Commit to save changes.
The following screen shows the SIP Entity for Communication Manager.
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The following screen shows the SIP Entity for Siemens HiPath 4000 V5.
The following screen shows the SIP Entity for Meeting Exchange 5.2.
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3.6. Administer Entity Links

A SIP trunk between a Session Manager and a telephony system is described by an Entity
Link. To add an Entity Link, select Entity Links on the left panel menu and click on the New
button (not shown). Fill in the following fields in the new row that is displayed.
Name
An informative name
SIP Entity 1 Select Session Manager
Port
Port number to which the other system sends its SIP requests
SIP Entity 2 The other SIP Entity for this link, created in Section 3.5
Port
Port number to which the other system expects to receive SIP requests
Trusted
Whether to trust the other system
Protocol
Transport protocol to be used to send SIP requests
Click Commit to save changes. The following screen shows the Entity Links used in the
sample network.
Entity Links are also required for Siemens HiPath 4000 V5 / Avaya Aura Session Manager
and Avaya Aura Meeting Exchange. SIP Communication from the Siemens HiPath 4000
V5 is only available via UDP.
3.7. Administer Time Ranges

Before adding routing policies (see next step), time ranges must be defined during which the
policies will be active. To add this time range, select Time Ranges from the left panel menu
and then click New on the right. Fill in the following fields.
Name
An informative name (e.g. always)
Mo through Su Check the box under each day of the week for inclusion
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Start Time
End Time
Enter start time (e.g. 00:00 for start of day)

Enter end time (e.g. 23:59 for end of day)
In Session Manager, a default policy (24/7) is available that would allow routing to occur at
anytime. This time range was used in the sample network.
3.8. Administer Routing Policies

Create routing policies to direct how calls will be routed to a system. Three routing policies
must be added, for Meeting Exchange, Siemens HiPath 4000 V5 and Communication
Manager. To add a routing policy, select Routing Policies on the left panel menu and then
click on the New button (not shown).
Under General:
Enter an informative Name
Under SIP Entity as Destination:
Click Select, and then select the appropriate SIP entity to which this routing policy
applies
Under Time of Day:
Click Add, and then select the time range configured in the Section 3.7.
Click Commit to save changes. The following screen shows the Routing Policy Details for
Meeting Exchange.
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The following is screen shows the Routing Policy Details for Siemens HiPath 4000.
The following is screen shows the Routing Policy Details for Communication Manager.
3.9. Administer Dial Patterns

A dial pattern must be defined that will direct calls to the appropriate telephony system. In
the sample network, 6-digit extensions beginning with 81 reside on Siemens HiPath 4000,
and 5-digit conference numbers begin with 37 Extensions for Digital or H.323 handsets on
the Avaya begin either 230 or 231. To configure the Siemens HiPath 4000 Dial Pattern select
Dial Patterns on the left panel menu and then click on the New button (not shown).
Under General:
Pattern
Dialed number or prefix
Min
Minimum length of dialed number
Max
Maximum length of dialed number
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SIP Domain Select ALL

Notes
Comment on purpose of dial pattern
Navigate to Originating Locations and Routing Policies and select Add (not shown). This
opens the Originating Location and Routing Policy List. Under Originating Location
select Apply The Selected Routing Policies to All Originating Locations and under
Routing Policies select Siemens HiPath 4000. Click Select button to confirm the chosen
options and then be returned to the Dial Pattern screen (shown previously), select Commit
button to save.
A dial pattern must be defined that will direct calls to Meeting Exchange. To configure the
Conferencing Dial Pattern select Dial Patterns on the left panel menu and then click on the
New button (not shown).
Under General:
Pattern
Min
Max
SIP Domain
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Dialed number or prefix

Minimum length of dialed number
Maximum length of dialed number
Select ALL
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Notes
Comment on purpose of dial pattern
Navigate to Originating Locations and Routing Policies and select Add (not shown). This
opens the Originating Location and Routing Policy List. Under Originating Location
select Apply The Selected Routing Policies to All Originating Locations and under
Routing Policies select the Routing Policy created for Conferencing in Section 3.8. Click
Select button to confirm the chosen options and then be returned to the Dial Pattern screen
(shown above), select Commit button to save.
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Screen Shots for Dial Pattern 230 (Digital extensions)
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3.10. Administer Avaya Aura Session Manager

To complete the configuration, adding the Session Manager will provide the linkage between
System Manager and Session Manager. On the SMGR management screen under the
Elements column select Session Manager.
In the left panel on the next screen, select Session Manager Administration and in the right
panel under Session Manager Instances select New (not shown). Fill in the fields as
described below and shown in the following screen:
Under General:
SIP Entity Name Select the name of the SIP Entity added for Session Manager
Description
Descriptive comment (optional)
Management Access Point Host Name/IP
Enter the IP address of the Session Manager management
interface
Under Security Module:
Network Mask
Enter the network mask corresponding to the IP address of
Session Manager
Default Gateway Enter the IP address of the default gateway for Session Manager
Use default values for the remaining fields. Click Commit to add this Session Manager.
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3.11. Add Avaya Aura Communication Manager as an Evolution

Server
In order for Communication Manager to provide configuration and Evolution Server support
to SIP phones when they register to Session Manager, Communication Manager must be
added as an application.
3.11.1.
Create an Avaya Aura Communication Manager Instance
On the SMGR management screen under the Elements column select Inventory.
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Select Manage Elements on the left. Click on New.
On the next screen (not shown), for Type select CM.

Click on the Applications tab and enter the following fields. Use defaults for the remaining
fields:
Name
A descriptive name
Description
A description of the CM instance
Node
Enter the IP address for CM SAT access
Click on the Attributes tab and enter the following:

Login
Login used for SAT access
Password
Password used for SAT access
Confirm Password Password used for SAT access
Click on Commit to save.
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3.11.2. Create an Avaya Aura Communication Manager Evolution

Server Application
On the SMGR management screen under the Elements column select Session Manager.
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Select Application Configuration Applications on the left menu. Click on New (not
shown). Enter the following fields and use defaults for the remaining fields. Click on
Commit to save.
Name
A descriptive name
SIP Entity
Select the CM SIP Entity defined in Section 3.5
CM System for SIP Entity Select the CM instance created in Section 3.11.1
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3.11.3. Create an Avaya Aura Communication Manager Evolution

Server Application Sequence
Select Application Configuration Application Sequences on the left menu. Click on
New (not shown). Enter a descriptive Name. Click on the + sign next to the appropriate
Available Applications and they will move up to the Applications in this Sequence section.
Click on Commit to save.
3.11.4.
Synchronize Avaya Aura Communication Manager Data
On the SMGR management screen under the Elements column select Inventory.
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Select Synchronization Communication System on the left. Select the appropriate

Element Name. Select Initialize data for selected devices. Then click on Now button (not
shown). Use the Refresh to monitor the Sync Status. This may take some time.
3.12. Add Users for SIP Phones

SIP users must be added via Session Manager and the details will be updated on
Communication Manaager. On the SMGR management screen under the Users column
select User Management.
Select Manage Users on the left. Then click on New.
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Click on the Identity tab and enter the following and use defaults for other fields:
Last Name
A desired last name
First Name
A desired first name
Login Name The desired phone extension [email protected] where domain
was defined in Section 3.2
Password
Password for user to log into System Manager (SMGR)
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Click on the Communication Profile tab. Enter the following and defaults for the remaining
fields:
Shared Communication Profile Password
Password to be entered by the user when logging into
the phone
Type
Select Avaya SIP
Fully Qualified Address
Enter the extension number and select the domain
Click on Add.
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Navigate to the Session Manager Profile and Endpoint Profile sections. Select the
appropriate Session Manager server for Primary Session Manager. For Origination
Application Sequence and Termination Application Sequence select the application
sequence created in Section 3.11.3. Choose the Home Location created in Section 3.3. Click
on Endpoint Profile to expand that section. Enter the following fields and use defaults for
the remaining fields:
System
Select the CM Entity
Extension
Enter a desired extension number
Template
Select a telephone type template
Port
Select IP
Click on Commit to save (not shown).
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4. Configure Siemens HiPath 4000 V5

This section shows the configuration in Siemens HiPath 4000 V5. All configurations in this
section are administered using HiPath Assistant. These Application Notes assumed that the
basic configuration has already been administered. For further information on Siemens
HiPath 4000 V5 , please consult with references [10]. The procedures include the following
areas:
Install and Configure of the Siemens HiPath V5 Assistant
Review Network Domain
Review System Configuration
Review Siemens Station Endpoints
Configuration of HG3500 Gateways
Confiuguration of SIP Trunk
Configuration of SIP Registrar Gateway
Configuration of H.323 Gateway
Configuration of Trunks and Trunk Groups
Configuration of Least Cost Routing.
Configure Dial Codes
Configure Digit Patterns
Configure LCR Route
Configure LCR Route Element
Configure Outdial Rule
Configure LCR Authorisations
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4.1. Install and Configure the Siemens HiPath 4000 V5 Assistant

The Siemens HiPath configuration was verified using the web interface HiPath 4000
Assistant V5. Before the web interface can be used, the client PC must be prepared. To
access the web interface use Internet Explorer to connect by entering http://<HiPath-PBXIPaddress> i.e. http://192.168.1.1
Note: This is an overview of the Siemens HiPath 4000 V5 Configuration and is intended for
guidance only. Always consult with your Siemens Qualified representative for exact
configuration regarding your site.
4.2. Client PC Preparation

To access and program the HiPath 4000 Assistant via the Internet browser, the user must first
prepare the client PC. Click on Client Preparation.
The following page is displayed, listing the steps to prepare the Client PC. Click on Next.
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Installation of Internet Browser [1/6]

If an error is displayed for the version of the Internet Browser, follow the steps listed under
Installation to install the correct level of Browser. Otherwise click Next (not shown).
Installation of Java Runtime Environment (Java Plug-In) [2/6]

Verify the client PC has a suitable version of Java Runtime Environment plug-in (JRE). In
the image below, a warning is displayed because the Java version does not match one of the
recommended versions available. Although you can ignore this warning if you have a newer
version of Java, should you experience problems with loading HiPath Assistant screens,
please clear all Java caches and uninstall your existing Java, delete the Applet folder (c:\
Appletcache) and reboot your PC. Return to these client preparation screen and install a
suitable level of Java. Once the Java has installed, close and re-open the browser and go to
the Client Preparation screens again. Click Next through the steps until you reach Step 3.
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Installation of Siemens I&C Security Certificate for JRE [3/6]

Verify the Siemens I&C Security Certificate. If an error message is displayed, click on
Installation to install the Security Certificate. Click Next.
It is strongly recommended that the browser be closed and re-opened after certificate
installation. Return to the Client Preparation process once the browser has re-opened.
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[Step 4a/6 and Step 4b/6]

The next two screens are information level notices regarding the installation of the certificate.
Click Next to proceed.
[Step 4b/6]
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Configuration of the Internet Browser [5/6]

Is also an Informational screen, but offers a diagnostic of the configuration so far.
Installation of the Applet Cache Manager [6/6].

If the Applet Cache Manager is not installed, click on the Installation button at the bottom of
the screen.
The Client PC will be prompted to install InstACM.exe. This is an Applet Cache manager
plug-in which is required to run HiPath Assistant Web Interface. Click on Run.
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The browser may need to be restarted after installation. Return to the Client Preparation and
step through until Step 6. Verify the Applet Cache Manager is installed.
Click Next to finish the Client Preparation and to begin Logon.
The second time of logging on to HiPath Assistant, you will not need to run the Client
Preparation again. Instead, from the public area, choose https://<PBX-IPaddress>
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4.3. HiPath 4000 System Configuration

Once the Client PC is prepared, access to the administration application is available. Return
to the main login page and click on the secure HiPath link.
A login window will appear. Log in using the engineering logon. (Refer to Siemens
Installation engineers or support team for client access details.)
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Once you are successfully logged in, the Assistant will be displayed in a tree structure on the
left. Once the section and subsections of the tree are expanded and the final level selected, a
new screen will open.
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4.4. Network Domain Configuration

A domain is a group of systems with the same tie line (closed numbering). To check or edit
the Network Domain Configuration, expand Configuration Management Network
Domain.
Click on Domain and a new window will open, all fields will be blank. Click on Search
button to retrieve all the Domain records. A search on blank fields usually returns up to 1000
records.
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The domain name is entered in the Domain field. The associated system name is listed under
the Systems tab.
4.5. Network System Configuration

To check/edit the System Configuration which contains the details of the version number,
upload status and the system number of the HiPath 4000. Expand Configuration
Management Network System. Wait for the screen to open in a new window.
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Leaving all fields empty, click on the Search button to return all records found.
Alternatively, click on the down arrow beside the field System to choose from a list. Select
the System name as recorded from the previous step and click Search. One or more fields
may used as search criteria.
4.6. Review of Board Configuration

The HiPath 4000 Communications Server uses ISDN cable connection to the cabinets
containing the Siemens HG3500 IP Gateways. These cards or boards are multifunctional and
can be programmed in different ways, depending on the requirements. The IP gateways also
communicate via IP network across the LAN. This sample configuration includes a SIP
Trunk Gateway (X.X.81.6), a SIP Registrar Gateway (X.X.81.3) for SIP Devices and a
H.323 (IP)Gateway (X.X.81.4) for IP devices. The Gateways are collectively described as
HG3500 or HG35XX boards and are universal cards. It is during configuration of these cards
that the role is defined as whether the boards are SIP or H.323.
To access the configuration of the HG3500 via HiPath Assistant.
Expand Configuration ManagementSystem DataBoardBoard. Wait for the screen
to load. Click on Search to retrieve all the boards programmed.
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Each Board has a specific location code, indicated by LTG (Line Trunk Group), LTU (Line
Trunk Unit) and SLOT(Physical position within the LTU where the card has been inserted).
In the image below, the board is positioned at 1-1-4. In the lab environment, this is the SIP
Trunk Gateway. The board location is important to know when creating Stations for
programming telephony end points.
The field CGW Function Block defines whether the board is SIP, H323 or a combination of
services and the number of channels used. For a full list of the boards available on the
system, click on the radio button Object List. This will display all the boards in a list
format. To look at an individual board, highlight it once in the list and click on Object. This
returns the screen to a single record view.
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The tab STMI2-IGW Board Data shows the IP address of the board , which will be used for
cross referencing with the Avaya configuration and is required as the Gateway or Registrar
address when configuring Siemens handsets.
(Screen shots for boards for SIP Phones and H323 Phones are not shown, but would show
similar information).
Board
Location /
PEN
1-1
1-1-1
1-1-2
1-1-4
1-1-14
1-2
1-2-1
1-2-2
1-2-13
IP Address
Function
NA
X.X.81.3
NA
X.X.81.6
NA
NA
X.X.81.4
NA
NA
1-2-14
X.X.81.5
LTU1 ISDN Admin Link

SIP Gateway Phone Registrar
TDM Interface [Siemens 500 Digital Phones]
SIP GW SIP Trunks
Q-Sig
LTU2 ISDN Admin Link
H.323 GW [Siemens 410 H.323 Phones]
24 Port Analog [Fax]
8 Ch. Voice Card. Interface to Siemens Expressions
Server
SIP Q trunking not currently in use
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4.7. Siemens Station End-Points

The Siemens stations (or end-points) can be managed from the HiPath Assistant. To check or
edit the Stations, expand Configuration ManagementStationStation. Wait for the
screen to display. Click on Search to return all records, alternatively if you know the Station
Number, enter it in the field Station No and then click Search.
The image below illustrates the configuration of a TDM Station:
Station No.
Selected from drop down list
PEN
Must match the Card location described in previous section.
(1-1-2 is the TDM Interface Card)
Device Combination Select from Drop down list
Device Family
Selected from Drop down list
Connection Type
Direct (TDM), IP2 (H.323) SIPSEC (SIP)
Display Name
Enter a suitable display name
COS 1 and 2
Class of Service must be assigned. Default shown
LCRCOS 1 and 2
Least Cost Routing COS must be assigned. Default shown
Way to Display
Set to yes to display Caller Name and ID
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The image below illustrates the configuration of a SIP Station Endpoint.

Station No
PEN
Must match the Card location described in previous section (11-1 is the SIP Registrar Interface Card)
Device Family
Display Name
COS 1 and 2
LCRCOS 1 and 2
Way to Display
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The image below illustrates the configuration of a H.323 Station Endpoint.

Station No
PEN
Must match the Card location described in previous section.
(1-2-1 is the H.323 GW Interface Card)
Device Family
Display Name
COS 1 and 2
LCRCOS 1 and 2
Way to Display
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4.8. HG 3500 Gateway Configuration

To access the configuration of the HG3500 Gateways, use the Web Console interface
https://IPAddress-Of-HGGW/ and Login screen will appear.
Alternatively the configuration of a card may be reached via HiPath Assistant by accessing
Expert Mode HG35XX Web Based Management.
Once you have selected this screen a pop up window will appear offering a list of boards
accessible via this method.
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Click on the connect link of the board you wish to review. On initial load of the screens, Java
is used and can take some time before the screens are available to use. Main menus of use
are:
Explorers Offers access to configure SIP parameters
Save
Save any changes made to the board via these screens
Reset
Some changes require the board to be restarted.
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4.9. SIP Trunk Gateway

The image below shows the SIP Trunk Gateway configuration, listing the SIP Parameters
used in these sample notes. Select Explorers Voice Gateway SIP Parameters. To edit
any of these settings, right click on SIP Parameters and choose Edit SIP Parameters. Not
all fields are configurable. Only the following fields are configurable via these screens.
Section
SIP Server (Registrar / Redirect)
RFC 3261 Timer Values
SIP Transport Protocol
SIP Session Timer
SIP Session Timer
SIP Session Timer
DNS-SRV Records
Outgoing Call Supervision
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Field Name
Period of registration (sec)
Transaction Timeout (msec)
SIP via UDP
RFC 4028 support
Session Expires (sec)
Minimal SE (sec)
Blocking time for unreachable destination(sec)
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The image below shows the SIP Trunk Profile configuration created for Avaya. Select
Explorers Voice Gateway SIP Trunk Profiles Avaya. Under the section Proxy
verify that the Session Manager SM100 SIP Signaling IP Address is entered. To edit the
settings, right click the folder and select edit. Make the necessary changes and click Apply
button at the bottom of the screen (not shown).Click the SAVE disk icon if it goes Red The
SIP Trunk Profile must then be activated, as show in the image below. To activate a SIP
Trunk Profile, right click on the profile and choose Activate. (Not shown). The folder will
then go Green to indicate it is the active Trunk Profile.
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4.10. SIP Registrar Gateway

This is the card used by the handsets when they register. The image below shows the SIP
Trunk Gateway configuration, listing the SIP Parameters used in these sample notes. Select
Explorer Voice Gateway SIP Parameters.
To view registered SIP clients select Explorers Voice Gateway Clients SIP.
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4.11. H.323 Gateway

The image below shows the H.323 GW configuration listing the H.323 Parameters used in
these sample notes. Select Explorers Voice Gateway H.323 Parameters.
To view registered H.323 clients select Explorers Voice Gateway Clients HFA.
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4.12. Trunks and Trunk Groups

Trunks and trunk groups used for SIP connectivity should also be present. The image below
shows the configuration of the SIP trunk used for connectivity to Avaya Aura Session
Manager. Configuration Management System Data Trunk Trunk.
PEN
Search for the PEN number of the board (i.e. 1-1-4-0 or use the
drop down list.)
B-Channel Grp. Usually set to 1
Trunk Type
Digital
Device
HG3550CO indicating a SIP enabled board for Central Office
Trunk Group
211 A trunk may only be assigned to one group.
Trunk Name
Logical Name for the Trunk Group
Node Number
A fictitious number to represent another switch
COT Number
Class of trunk parameters
COP Number
Class of Parameters for Device handlers
COS, LCRCOS
Class of Service and Least Cost Routing Class of Service.
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The Trunk Group Screens shows the Configuration of the Trunk group. Configuration
ManagementSystem DataTrunkTrunk Group.
Trunk Group Info Tab
Trunk Group
Trunk Group Number.
Name
Logical Name for the Trunk Group
Max. No of Lines
Number of lines available to the trunk group
Trunk Group Members Tab
Shows the PEN(s) assigned to the trunk group. In this example the trunk 1-1-4-0 for the SIP
Gateway is assigned.
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4.13. Least Cost Routing

A dialable routing from the Siemens HiPath 4000 to the Avaya Aura Communication
Manager must be inserted into the Siemens LCR tables before a device can contact an Avaya
handset. Programming a LCR route on the Siemens can be quite complex. The steps and
screen shots below are an overview.
4.13.1.
Configure Dial Codes
First the digit range must be assigned in the Siemens HiPath 4000, indicating what the
number is to be used for i.e. Stations, feature codes, TIE lines etc., Configuration
Management Tables Dial Plan Dial Codes.
Dial Code
Initial Digits in the range to be dialed.
Code Type
Set to TIE for TIE line
DPLN
Enter for Dial Plan independent or can be assigned to dial plans (015)
CPS (0-22)
Call Progress State. Various states to determine how the call is
handled.
4.13.2.
Configure Digit Pattern
The HiPath 4000 must also be programmed with the Dial Pattern which determines the
routing the call will take to reach its destination amongst other settings. Configuration
ManagementLeast Cost RoutingDigit Pattern
Digit Pattern
Digit Pattern of the number being dialled. The Z indicates
random number of digits followed by a timeout or # (=end of
dialing)
Dial Plan Number Default is 0 meaning the number is assigned to all dial plans
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Digit Counter
DPLN
Route
Auth.Code (1-64)
The values of the digit counter assigned to the digit pattern

displayed.
The number of the dial plan group for the neighboring entries
in the Route and LCR Auth fields. The value "0-15" is visible
in the DPLN field if an assignment of this type applies "for all
DPLN groups".
Contains the number of the LCR ROUTE.
Assign an LCR authorization of the corresponding
authorization code (1...64) to a DPLN group. A subscriber from
the DPLN group displayed needs the LCR authorization for
dialing the digit pattern. The LCR authorizations are used to
disable or enable the digit pattern for specific subscribers. The
Station COS settings will show which Authorisations the
handset has and therefore which numbers it has permission to
dial.
For further information regarding LCR Authorisation, refer to Section 2.13.6.
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4.13.3.
Configure LCR Route
The LCR Route determines the trunk group used. Configuration ManagementLeast Cost
RoutingLCR Route.
Dest. Node
Fictitious Number representing the far end PBX
Route
Assign a Route number
Trunk Group Trunk Group number from Section 2.12
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4.13.4.
Configure LCR Route Element
The LCR Route Element carries further settings regarding the LCR Route used.
Configuration ManagementLeast Cost RoutingLCR Route Element.
Route
Route Number
Name
Route Name
Outdial Rule Rule for how the number is dialed
Tr.Grp.No Trunk Group Number
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4.13.5.
Configure Outdial Rule
The outdial rule details how the number is dialled. Configuration Management Least
Cost Routing Outdial Rule.
Outdial Rule
Identifying Number for the rule
Outdial Rule Elements
Sequential series of steps determing how the number is
dialed i.e. NPI/TON is checking for a Network Plan
Identifier in conjunction with a Local Subscriber
(TON). The ECHO will outpulse all the digits of a
specified field. The Parameter indicates which field to
apply this to. END is mandatory on all Outdial rules.
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4.13.6.
Configure LCR Authorisations
When entering the Digit Pattern (Section 5.13.2), authorisations were assigned to the Digit
Pattern. Configuration Management Least Cost Routing Digit Pattern. Each digit
pattern created can belong to between 1-64 Authorisation codes. (Sample screenshot below).
Each Subscriber is assigned a COS, LCOSV and LCOSD. Configuration Management

Station Station.
COS (Class of Service)
Controls the features the handset may access i.e. Call
Forwarding, DND, MakeBusy, Trunk Access etc
LCRCOS Voice(1/2)
Least Cost Routing Class of Service (Voice)
LCRCOS Data (1/2)
Least Cost Routing Class of Service (Data)
The digits 1 and 2 indicate a day/night setting, whereby a different COS/LCRCOS may be
used. Each LCRCOS is assigned LCR Authorisation Code(s), controlling the numbers a user
can dial.
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The LCR COS is accessed via Configuration Management Least Cost Routing LCR
Authorizations LCR Class of Service Voice.
LCOS Voice
Idenitfying number for LCOS voice
Auth. Code (1-64) Assign relevant Auth.Code(s) to LCOS Voice
The LCR Class of Service-Voice screen indicates the Authority Codes assigned. When the
digit pattern 623-Z was created, it was assigned Authority Code 1. (See first screen shot in
Section 5.13.6). Any subscriber that uses a LCR COS Voice, with Authority Code 1 enabled
is now permitted to dial number starting 623.
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5. Configure Avaya Meeting Exchange Enterprise

This section describes the steps for configuring the Meeting Exchange to interoperate with
Siemens HiPath 4000 V5 via SIP trunking. It is assumed that the Meeting Exchange is
installed and licensed as described in the product documentation (see reference [5] and [6]).
The following steps describe the administrative procedures for configuring Meeting
Exchange:
Configure SIP Connectivity
Configure Dialout
Map DNIS Entries
Configure Audio Preferences
Restarting the Meeting Exchange server
Configure Bridge Talk
The following instructions require logging in to the Meeting Exchange console using an ssh
connection to access the Command Line Interface (CLI) with the appropriate credentials.
5.1. Configuring SIP Connectivity

Log in to the Meeting Exchange server console using an ssh Client to access the Command
Line Interface (CLI) with the appropriate credentials. Configure settings that enable SIP
connectivity between the Meeting Exchange server and other devices by editing the
system.cfg file as follows:
Edit /usr/ipcb/config/system.cfg
Add Meeting Exchange server IP address
o IPAddress=(10.10.21.51)
Depending on the SIP signalling protocol, TCP or UDP, add one of the following
lines to populate the From Header Field in SIP INVITE messages:
o MyListener=<sip:[email protected]:5060;transport=tcp>
o MyListener=<sip:[email protected]:5060;transport=udp>
Note: The user field 6000, defined for this SIP URI must conform to RFC 3261. For
consistency, it is selected to match the user field provisioned for the respContact
entry (see below).
Depending on the SIP signalling protocol, TCP or UDP , add one of the following
lines to provide SIP Device Contact address to use for acknowledging SIP messages
from the Meeting Exchange server:
o respContact=<sip:[email protected]:5060;transport=tcp>
o respContact=<sip:[email protected]:5060;transport=udp>
Add the following lines to set the Min-SE timer to 900 seconds in SIP INVITE
messages from the Meeting Exchange server:
o sessionRefreshTimerValue= 900
o minSETimerValue= 900
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[sroot@MXAPP2 ~]# vi /usr/ipcb/config/system.cfg

# ip address of the server
IPAddress=10.10.21.51
# request we will be listening to
MyListener=<sip:[email protected]:5060;transport=tcp>
# if this setting is populated will Overwrite the contact field in
responses
respContact=<sip:[email protected]:5060;transport=tcp>
MaxChannelCount=3200
# diff serv values that will appear on the TOS field of the IP packet
DiffServSignallingTOSValue=46
DiffServMediaTOSValue=24
# vlan values
EthernetSignallingVlanValue=0
EthernetMediaVlanValue=0
# SIP settings
sessionRefreshTimer=900
minSETimerValue=900
#outboundProxyRoute=
#allRouteTo=
# Not used for Unixware MeetingXchange
MaxMeetingCount=400
MaxConferenceCount=500
MaxOperatorCount=10
MaxNumReconnectPerSecond=200
# video port licensing
MaxVideoChannelsAllowed=0
BillingDirectory=/usr/ipcb/billing
# initipcb process keep-alive time (seconds)
processKeepAlivePollTime=11
5.2. Configure Dialout

To enable Dial-Out from the Meeting Exchange to HiPath 4000, edit the telnumToUri.tab
file as follows:
Edit /usr/ipcb/config/telnumToUri.tab file with a text editor
Add the following line to the file to route outbound calls from the Meeting Exchange
to the Siemens HiPath 4000 via the Session Manager SM100 ip address.
*
sip:[email protected]:5060;transport=tcp
#
#
#
#
#
#
#
#
#
telnum to uri conversion table

This file is for dialing out from the Bridge to an external party. The
digits that are dialed are converted into the Request URI in the SIP INVITE.
For example, if the digits dialed were 936543 and one of the patterns was
"93????" a match would take place. If the conversion for that match was
$1 then the Request URI for the SIP INVITE would be sip:[email protected]
THE COMMENT COLUMN OR ANY OF THE COLUMNS SHOULD HAVE NO SPACES
TelnumPattern
*
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TelnumConversion
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5.3. Map DNIS Entries

The DNIS entry is the number dialled by Siemens or Avaya subscribers to access a
conference service (SCAN, FLEX or DIRECT) on Meeting Exchange. The DNIS entry needs
to be mapped on Meeting Exchange to enable access to a conference. To map DNIS entries,
run the cbutil utility on Meeting Exchange. Log in to the Meeting Exchange with a ssh
connection with the appropriate credentials. Enable Dial-In access (via passcode) to
conferences provisioned on the Meeting Exchange as follows.
Add a DNIS entry for a scan call function corresponding to DID 38888 by entering the
following command at the command prompt:
cbutil add <dnis> <rg> <msg> <ps> <ucps> <func> [-o <of> -l <ln> -c <cn> -crs <n> cre <n> -cc <code>]
where the variables for add command is defined as follows:
<dnis>
DNIS
<rg>
Reservation Group
<msg>
Annunciator message number
<ps>
Prompt Set number (0-20)
<ucps>
Use Conference Prompt Set (y/n)
<func>
One of: DIRECT/SCAN/ENTER/HANGUP/AUTOVL/FLEX
o <of>
Optional On-failure function one of: ENTER/HANGUP or
DEFAULT.
l <"ln">
Optional line name to associate with caller
c <"cn"> Optional company name to associate with caller
crs <n>
Optional conference room start number
cre <n>
Optional conference room end number
cc<n>
Optional setting only used with DIRECT
In this sample configuration, the DNIS entry for a FLEX call function was added
corresponding to DNIS 38111 by entering the following command at the command prompt:
[sroot@MXAPP2 ~]# cbutil add 38111 0 247 1 N FLEX
cbutil
Copyright 2004 Avaya, Inc. All rights reserved.
At the command prompt, enter cbutil list to verify the DNIS entries provisioned.
[sroot@MXAPP2 ~]# cbutil list
cbutil
Copyright 2004 Avaya, Inc. All rights reserved.
DNIS
Grp Msg PS CP Function On Failure Line Name
Company Name
---------------- --- ---- --- -- --------- --------- -------------------- ---38111
0
247 1
N FLEX
DEFAULT
38222
301
DIRECT
DEFAULT
38888
247
SCAN
DEFAULT
ScanConf
For further details of the parameter fields used, see Reference [5].
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5.4. Configure Audio Preferences file

The audioPreferences.cfg file located at /usr/ipcb/config/ specifies the order in which
codecs are offered in the Session Description Protocol. Set the telephone-event value to
payloadType of 97.
# audioPreferences.cfg
# This table is an ordered list of MIME subtypes specifying the codecs supported
# by this media server. The list is specified in the order in which an SDP offer
# will list the various MIME subtypes on the m=audio line.
# For static payload type numbers (i.e. numbers between 0 - 96) please use the
# iana registered numbering scheme.
# See: http://www.iana.org/assignments/rtp-parameters
mimeSubtype
payloadType
Performance
PCMU
0
10
#PCMA
8
10
#G722
9
85
G729
18
118
#iLBC30
97
140
#iLBC20
98
140
#wbPCMU
102
10
#wbPCMA
103
10
telephone-event
97
10
#iSAC
104
150
#G726_16
105
54
#G726_24
106
54
#G726_32
107
54
#G726_40
108
54
5.5. Restarting the Meeting Exchange Server

After the configuration changes are made, restart the services issuing the command service
mx-bridge restart.
# service mx-bridge restart
/etc/init.d/mx-bridge: Restarting bridge
/etc/init.d/mx-bridge: Server type is DCB
/etc/init.d/mx-bridge: Stopping DCB conferencing server bridge via uninitdcb.sh
Stopping notificationCtrlServer service:
killproc notificationCtrlServer
[ OK ]
Sending CMD_SHUTDOWN level 3 message to the INIT_KEY queue.
Waiting for 6 processes to stop
destroy.
/etc/init.d/mx-bridge: mx-bridge startup
/etc/init.d/mx-bridge: Server type is DCB
Add Process Key 145 IP address 10.10.6.20

Add Process Key 146 IP address 10.10.6.20
key ID 101
key ID 102
key ID 110
=========================================== INITDCB ==============================
FirstMusic = 3199.
FirstLink = 3199.
FirstRP = 3198.
FirstOper = 3195.
numUserLCNs = 3195.
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5.6. Avaya Bridge Talk

The following steps utilize the Bridge Talk application to provision a sample conference on
the Meeting Exchange. This sample conference enables both Dial-In and Dial-Out access to
audio conferencing for endpoints on the Public Switched Telephone Network.
Notes: If any of the features displayed in the Avaya Bridge Talk screen captures are not
present, contact an authorized Avaya Sales representative to make the appropriate changes.
5.6.1. Initializing Bridge Talk

Invoke the Avaya Bridge Talk application as follows:
Double-click on the desktop icon from a Personal Computer loaded with the Avaya
Bridge Talk application and with network connectivity to the Meeting Exchange (Not
shown).
Enter the appropriate credentials in the Sign-In and Password fields.
Enter the IP address of the Meeting Exchange server (10.10.21.51 for this sample
configuration) in the Bridge field as shown below.
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5.6.2. Creating a Dial Out list

Provision a dial list that is utilized for Dial-Out (e.g., Blast dial and Fast dial) from the
Meeting Exchange. From the Avaya Bridge Talk Menu Bar, click Fast Dial New.
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5.6.3. Creating a Dial List

From the Dial List Editor window that is displayed below.
Enter a descriptive label in the Name field.
Enable conference participants on the dial list to enter the conference without a
passcode by selecting the Directly to Conf box as displayed.
Add entries to the dial list by clicking on the Add button and enter Name, Company
and Telephone number for dial out for each participant. [Optional] Moderator
privileges may be granted to a conference participant by checking the Moderator
box.
When finished, click on the Save button on the bottom of the screen.
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5.6.4. Conference Scheduler

From the Avaya Bridge Talk menu bar, click View Conference Scheduler to provision a
conference.
5.6.5. Scheduling a Conference

From the Conference Scheduler window, click File Schedule Conference.
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5.6.6. Provision a Conference

From the Schedule Conference window that is displayed, provision a conference as follows:
Choose Conference Type and set to be available at the Weekend
Enter a Start Date and End Date. Leave End Date blank for open ended conference.
Enter a unique Conferee Code to allow participants access to this conference.
Enter a unique Moderator Code to allow participants access to this conference with
moderator privileges.
Enter a descriptive label in the Conference Name field.
Administer settings to enable an Auto Blast dial by setting Auto/Manual as desired.
Set Start Time and End Time to determine when the conference can be activated.
Set the Maximum Lines to be used in the conference
Select a dial list by clicking on the Dial List button. Either select a preconfigured list, by
clicking on the item and clicking select, Edit an existing list, or Create a new list. (Screens
not shown). When finished, click on the Save button on the bottom of the screen.
6. Verification
This section provides the verification tests that can be performed on Siemens HiPath 4000 V5
and Meeting Exchange to verify their proper configuration.
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6.1. Verify Siemens HiPath 4000 V5

Using HiPath Assistant, a limited number of tools are available for testing purposes. Logon to
HiPath Assistant using the engineering logon. Open Diagnostics Switch Diagnosis
Support Real Time Diagnosis System. Wait for the screen to load.
User Name Enter a user name
Description Enter a brief description
Click Logon to start the Diagnosis system.
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Once Logon has been clicked, select Monitor Trunk/Line/Data Line/ISDN trace from the
list on the right hand side. When this window has opened, set the Station number or trunk
number (PEN) and the Type of monitor. Click on the relevant Start buttons and then click
Monitor Display.
Monitor Display will open a new window which will show results once a call has been made.
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To exit the Real Time Diagnosis

1. Close the Monitor Display (Alt F4)
2. Click Stop ALL on the Monitor Trace Setup and then close the screen (Alt F4)
3. Click Shutdown RDS on the main RealTime Diagnosis Home Screen. Once the
message RDS has been terminated. All RDS servers have been shut down is
displayed, close this screen as well (Alt F4).
6.2. Verify Avaya Meeting Exchange Enterprise

Verify all conferencing related processes are running on the Meeting Exchange as follows:
Log in to the Meeting Exchange server console to access the CLI with the appropriate
credentials.
cd to /usr/dcb/bin
At the command prompt, run the script service mx-bridge status and confirm all
processes are running by verifying an associated 4-digit Process ID (PID) for each
process.
[sroot@MXAPP2 ~]# service mx-bridge status
7404 ?
00:00:00 initdcb
7439 ?
00:00:00 log
7442 ?
00:00:00 bridgeTranslato
7443 ?
00:00:00 netservices
7473 ?
00:00:00 timer
7474 ?
00:00:00 traffic
7475 ?
00:00:00 chdbased
7476 ?
00:00:00 startd
7477 ?
00:00:00 cdr
7478 ?
00:00:00 modapid
7479 ?
00:00:00 schapid
7480 ?
00:00:00 callhand
7481 ?
00:00:00 initipcb
7492 ?
00:00:00 sipagent
7499 ?
00:00:01 msdispatcher
7501 ?
00:00:00 serverComms
7502 ?
00:00:00 softms
7503 ?
00:00:00 softms
7504 ?
00:00:00 softms
7506 ?
00:00:00 softms
7507 ?
00:00:00 softms
7508 ?
00:00:00 softms
11434 ?
00:00:00 schapid
3663 ?
00:00:00 postmaster with 27 children
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6.2.1. Verify Call Routing

Verify end to end signalling/media connectivity between the Meeting Exchange and Siemens
HiPath 4000 V5. This is accomplished by placing calls from Siemens end points to the
Meeting Exchange. This step utilizes the Avaya Bridge Talk application to verify calls to and
from the Meeting Exchange are managed correctly, e.g., callers are added/removed from
conferences. This step will also verify the conferencing applications provisioned.
Configure a conference with Auto Blast enabled and provision a dial list. From an
Siemens endpoint, dial a number that corresponds to DNIS 38888 to enter a
conference as Moderator (with passcode) and blast dial is invoked automatically.
When answered these callers enter the conference.
If not already logged on, log in to the Avaya Bridge Talk application with the
appropriate credentials
Double-Click on the highlighted Conf # to open a Conference Room window
Verify conference participants are added/removed from conferences by observing the
Conference Navigator and/or Conference Room windows.
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6.3. Verified Scenarios

The following scenarios have been verified for the configuration described in these
Application Notes.
Conference calls including various telephone types (see Figure 1) on the Siemens
HiPath 4000 V5 can be made using G.711mu/A-law and G.729.
Scan, Flex, and Direct Conference modes.
Name Recording, Conference recording and Playback (NRP).
RFC 2833 DTMF support for all moderator and conferee commands.
Manual and automatic blast dial-out to conference participants.
Network outage failure and recovery.
Bridge process (softms) failure and recovery.
Session timers on Meeting Exchange.
Line and Conference transfer by Operator.
7. Conclusion
As illustrated in these Application Notes, Siemens HiPath 4000 V5 can interoperate with
Avaya Meeting Exchange Enterprise Edition using a SIP trunk using Avaya Aura
Communication Manager and Avaya Aura Session Manager with Avaya Aura System
Manager. No issues were found during testing with Siemens HiPath 4000 V5 using OptiPoint
and OpenStage SIP and IP devices.
8. Additional References
Product documentation for Avaya products may be found at http://support.avaya.com
[1] Administering Avaya Aura Communication Manager 03-300509 Release 6.0 Issue
6.0
[2] Administering Avaya Aura Communication Manager Server Options 03-603479
Release 6.0.1, Issue 2.2
[3] Administering Avaya Aura Session Manager 03-603324 Release 6.1 Issue 1.0
[4] Maintaining and Troubleshooting Avaya Aura Session Manager 03-603325
Release 6.1 Issue 4.1
[5] Using Meeting Exchange Release 5.2.2 04-603698 Issue 1
[6] Administering Meeting Exchange Servers Release 5.2.2 04-603708 Issue 1
[7] Avaya BridgeTalk v5.2.2.0.9 integrated program Help.
Product documentation for Siemens products may be found at the following websites and
within service manuals.
[8] Siemens Wiki Website http://wiki.siemens-enterprise.com/index.php/Main_Page
[9] Siemens eTAC (a support website with limited information available to
customer/end user) http://siemenstac.custhelp.com/app/home
[10] HiPath 4000 V5 IP Solutions, SIP Connectivity Service Documentation ref:
A31003-H3150-S104-2-7620 Dec 2008.
[11] Other Service Documentation may be obtained via your Service Support Provider or
Account Manager within Siemens Enterprise Communications.
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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 Solution & Interoperability Test Lab at [email protected]
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Avaya Solution & Interoperability Test Lab

Configuring Siemens HiPath 4000 V5 with Avaya Meeting

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Figure 1: Connection of Siemens HiPath 4000 V5 and Avaya Meeting Exchange

Solution & Interoperability Test Lab Application Notes

1.1. Equipment and Software Validated

Avaya Handset 9621G

Solution & Interoperability Test Lab Application Notes

2. Configure Avaya Aura Communication Manager

2.1. Verify Avaya Aura Communication Manager License

Solution & Interoperability Test Lab Application Notes

2.2. Administer System Parameters Features

2.3. Administer IP Node Names

Solution & Interoperability Test Lab Application Notes

2.4. Administer IP Network Region and Codec Set

Solution & Interoperability Test Lab Application Notes

2.5. Administer SIP Signaling Group and Trunk Group

Near-end Node Name: clan

SIP Enabled LSP? n

Far-end Node Name: sm100b

Bypass If IP Threshold Exceeded?

Solution & Interoperability Test Lab Application Notes

2.5.2. SIP Trunk Group

Group Type: sip

Navigate to Page 3 and enter private for Numbering Format.

Numbering Format: private

Modify Tandem Calling Number: no

Show ANSWERED BY on Display? y

Solution & Interoperability Test Lab Application Notes

2.6. Administer Route Pattern

ITC BCIE Service/Feature PARM

No. Numbering LAR

2.7. Administer Private Numbering

NUMBERING - PRIVATE FORMAT

2.8. Administer Dial Plan and AAR analysis

Solution & Interoperability Test Lab Application Notes

2.9. Save Changes

Solution & Interoperability Test Lab Application Notes

3. Configuring Avaya Aura Session Manager

3.1. Log in to Avaya Aura System Manager

Solution & Interoperability Test Lab Application Notes

In the next screen under Elements column select Routing.

Solution & Interoperability Test Lab Application Notes

3.2. Administer SIP Domain

3.3. Administer Locations

Solution & Interoperability Test Lab Application Notes

3.4. Administer Adaptations

Solution & Interoperability Test Lab Application Notes

3.5. Administer SIP Entities

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

3.6. Administer Entity Links

3.7. Administer Time Ranges

Solution & Interoperability Test Lab Application Notes

Enter start time (e.g. 00:00 for start of day)

3.8. Administer Routing Policies

Solution & Interoperability Test Lab Application Notes