Avaya & Hipath

Avaya Solution & Interoperability Test Lab
Application Notes for configuring Siemens HiPath 4000

V5with Avaya Aura® System Manager R6.3, Avaya Aura®
Session Manager R6.3 and Avaya Aura® Conferencing 7.0 –
Issue 1.0
Abstract
These Application Notes present a sample configuration for connectivity between Siemens
HiPath 4000 V5 with Avaya Aura® Communication Manager R6.2, Avaya Aura® Session
Manager R6.3 and Avaya Aura® Conferencing 7.0 using SIP Trunking.
Testing was conducted via the Internal Interoperability Program at the Avaya Solution and
Interoperability Test Lab.
BD; Reviewed: Solution & Interoperability Test Lab Application Notes 1 of 117
SPOC 6/24/2013 ©2013 Avaya Inc. All Rights Reserved. AAC7_H4K_SM63
Table of Contents
Table of Contents
1. Introduction ............................................................................................................................. 5
2. Interoperability Testing ........................................................................................................... 6
2.1. Test Description and Coverage ........................................................................................ 6
2.2. Test Results and Observations ......................................................................................... 6
2.2.1. Dial Out functionality ............................................................................................... 6
2.2.2. Unattended Transfer / Cross PBX Unattended Transfer / Attended Transfer by
Flare Devices .......................................................................................................................... 7
2.3. Known Issues and Limitations ......................................................................................... 7
2.4. Interconnectivity between Conferencing and Siemens HiPath 4000 ............................... 7
3. Reference Configuration ......................................................................................................... 8
4. Equipment and Software Validated ........................................................................................ 9
5. Configure Avaya Aura® Communication Manager ............................................................. 10
5.1. Verify Avaya Aura® Communication Manager License .............................................. 11
5.2. Administer System Parameter Features ......................................................................... 11
5.3. Administer IP Node Names............................................................................................ 13
5.4. Administer IP Network Region and Codec Set .............................................................. 13
5.5. Create SIP Signaling Group and Trunk Group .............................................................. 15
5.5.1. SIP Signaling Group ............................................................................................... 15
5.5.2. SIP Trunk Group ..................................................................................................... 16
5.6. Administer Route Pattern ............................................................................................... 17
5.7. Administer Private Numbering ...................................................................................... 18
5.8. Administer Locations ..................................................................................................... 18
5.9. Administer Dial Plan and AAR Analysis ....................................................................... 19
5.10. Save Changes .............................................................................................................. 20
6. Configure Avaya Aura® Session Manager ........................................................................... 21
6.1. Log in to Avaya Aura® System Manager ...................................................................... 21
6.2. Administer SIP Domains ................................................................................................ 23
6.3. Administer Locations ..................................................................................................... 24
6.4. Administer Adaptations.................................................................................................. 26
6.5. Administer SIP Entities .................................................................................................. 26
6.6. Administer SIP Entity Links .......................................................................................... 30
6.7. Administer Time Ranges ................................................................................................ 33
6.8. Administer Routing Policy ............................................................................................. 33
6.9. Administer Dial Pattern .................................................................................................. 37
6.10. Administer Avaya Aura® Session Manager ............................................................... 40
6.11. Add Avaya Aura® Communication Manager as an Evolution Server....................... 41
6.11.1. Create a Avaya Aura® Communication Manager Instance ................................ 41
6.11.2. Create an Evolution Server Application.............................................................. 43
6.12. Synchronize Avaya Aura® Communication Manager Data ...................................... 45
6.13. Configure Avaya Extensions for Conferencing.......................................................... 45
7. Configure Siemens HiPath 4000 V5 ..................................................................................... 46
7.1. Configuring HiPath Assistant ........................................................................................ 47
7.2. HiPath 4000 System Configuration................................................................................ 52
7.2.1. HiPath Network Domain Configuration ................................................................. 53
7.2.2. HiPath System Configuration ................................................................................. 55
7.2.3. Configuring HiPath Expert Access (ComWin)....................................................... 55
7.3. Configuring SIP Trunking.............................................................................................. 61
7.3.1. Configure LEGK ..................................................................................................... 61
7.3.2. Configure the HG3500 Board ................................................................................. 61
7.3.3. Configuring the Gate Keeper .................................................................................. 70
7.4. HG 3500 SIP Gateway Configuration via Web Interface .............................................. 72
7.4.1. SIP Trunk Gateway ................................................................................................. 73
7.4.2. SIP Registrar Gateway ............................................................................................ 74
7.4.3. H.323 Gateway ....................................................................................................... 75
7.5. Least Cost Routing ......................................................................................................... 76
7.5.1. Configure Dial Codes ............................................................................................. 76
7.5.2. Configure Digit Pattern ........................................................................................... 76
7.5.3. Configure LCR Route ............................................................................................. 78
7.5.4. Configure LCR Route Element ............................................................................... 79
7.5.5. Configure Outdial Rule ........................................................................................... 80
7.5.6. Configure LCR Authorizations ............................................................................... 81
7.6. Siemens Station End-Points ........................................................................................... 83
7.6.1. Configuration of H.323 and SIP Endpoints via Web Interface .............................. 86
7.7. Enable Direct Media Connection and BCHANL on Siemens Endpoints ...................... 88
7.8. Class of Parameter .......................................................................................................... 89
7.9. Class of Trunk ................................................................................................................ 90
7.10. Class of Service .......................................................................................................... 91
7.11. Altering Codec Settings .............................................................................................. 93
7.12. Other System Settings ................................................................................................ 94
8. Configure Avaya Aura® Conferencing 7.0 .......................................................................... 96
8.1. Add Meet Me and Adhoc URIs to Provision Client ...................................................... 96
8.2. Add a Location ............................................................................................................. 100
8.3. Assign Media Server Clusters to Locations ................................................................. 101
8.4. Add System Manager Domains to Provisioning Client ............................................... 102
8.5. Configuring a Web Conferencing Server ..................................................................... 104
8.6. Configuring a Web Conferencing Host ........................................................................ 105
8.7. Additional Configuration of Conferencing for Siemens HiPath 4000 Interconnectivity
106
9. Administer Avaya Extensions for Conferencing. ............................................................... 108
9.1. Configure SIP phone with Conference Profile ............................................................. 108
9.2. Configure H.323 phone with Conference Profile......................................................... 111
10. Verification Steps............................................................................................................. 112
10.1. Verification of Siemens HiPath 4000 V5 ................................................................. 112
10.2. Verification of Avaya Endpoints .............................................................................. 113
10.3. Verification of Avaya Conferencing ........................................................................ 115
11. Conclusion ....................................................................................................................... 115
12. Additional References ...................................................................................................... 116
1. Introduction
The purpose of this interoperability Application Note is to validate Siemens HiPath 4000 V5
with Avaya Aura® Communication Manager R6.2, Avaya Aura® Session Manager R6.3 and
Avaya Aura ® conferencing 7.0 which are connected via SIP trunking. The purpose of this
interoperability note is to validate basic telephone calls and features from the Avaya and Siemens
systems with Avaya Aura® Conferencing 7.0.
Avaya Aura Conferencing 7.0 is an Enterprise conferencing and collaboration product providing
planned and on-demand integrated Audio, Web, and Video conferencing and control from a
single point for a seamless Unified Communication experience. Avaya Aura® Conferencing
provides reliable call preservation and redundancy as well as outstanding bandwidth
management and utilization with distributed architecture and dynamic adaptation through media
cascading. A lower total cost of ownership is driven by a unified infrastructure, simplifying
management and lower acquisition, upgrade, and bandwidth costs.
The distributed architecture allows cascading Media Servers to be positioned to optimize WAN
bandwidth and provide high availability redundancy. Avaya Aura® Conferencing supports on-
demand conferencing through MeetMe and Event, and Adhoc conferences with advanced
conference controls.
The Avaya Aura® Conferencing solution has strong integration with the Avaya Aura® core and
includes broad support for Avaya endpoints. Avaya Aura® Conferencing video conferencing
supports high definition resolutions up to 720p through a software video routing technology
based on the H.264 SVC standard.
The Siemens HiPath 4000 V5 consisted of HiPath 4000 Communication Server which is
effectively the processor for the system. Attached via LTU (Line Trunk Units) links are two
AP3700 cabinets, containing the HG3500 boards which provide both SIP and IP connectivity
dependant on the programming configuration. The AP3700 additionally contained a TDM board
and Analog board for non IP/SIP based telephony.
2. Interoperability Testing
The general test approach was to simulate an Avaya enterprise environment consisting of
Communication Manager R6.2, Conferencing 7.0 co-resident Simplex and Siemens HiPath 4000
V5. The separate devices were connected via SIP trunking provided by Session Manager R6.3
and managed using System ManagerR6.3. Calls were made from endpoints on the respective
systems into the Conferencing 7.0 and conferencing features were tested from both Avaya and
Siemens endpoints.
2.1. Test Description and Coverage

Siemens endpoints were tested using SIP and Non SIP (IP and Digital) devices. Testing only
covered Audio as none of the Siemens devices have video capability.
Testing focused on the following:
MeetMe Conferencing using Avaya and Siemens endpoints
G711U/G711A and G729 Codec with IP Shuffle
Moderator / Participant Features
o Dialout
o Silence / Unsilence
o Lecture Mode
o Mute/Unmute
o Other Conference features where applicable
Calls on Hold
Unattended / attended transfer
Longevity Test
Network Outage and Recovery
2.2. Test Results and Observations

Testing was generally successful, with 83% of tests passing. The remaining test failures detailed
in Section 2.2.1 and Section 2.2.2.
2.2.1. Dial Out functionality

Issues were observed when an Avaya SIP device was performing a dial out to another Avaya SIP
destination. This issue did not occur when the dial out to destination was either an Avaya H323
device or an extension on the Siemens PBX (TDM, IP or SIP). This fault is currently under
investigation and has been logged in JIRA Ref AAC-414. In essence, the SIP commands relating
to the call are being received back to the Conference but are not being acted on when received.
Scenario
Avaya SIP 9621 dials Conference Meet me or Adhoc number and enters Moderator PIN.
Avaya is now Moderator in Conference.
Avaya SIP 9621 enters *1 and is prompted to enter an extension number, followed by #.
The call is set up and the destination phone rings
On answer at the destination phone the message “you have been invited to join the
conference, please press 1” should be heard but is not being heard.
This failure only occurs when the call is being made between two Avaya SIP devices.
The issue is NOT seen in the following circumstances:-
Avaya SIP (Moderator) dial out to Avaya H323 – H323 hears message and can press 1 to
join conference
Avaya H323 (Moderator) dial out to Avaya SIP – SIP hears message and can press 1 to
join conference
Avaya SIP (Moderator) dial out to Siemens across SIP trunk via SM to either Siemens
SIP or Siemens non SIP phone – both can join the conference
Siemens ( as Moderator / SIP or non SIP) dial out to other Siemens/Avaya H.323 or
Avaya SIP is also successful
2.2.2. Unattended Transfer / Cross PBX Unattended Transfer / Attended

Transfer by Flare Devices
Neither the ADVD A175 nor Flare for Windows can perform an Unattended Transfer. This is a
known feature but is working as designed.
The ADVD can perform an Attended transfer.
The Avaya Flare for Windows can only perform an “attended transfer” using Adhoc
Conference. The general method is to bring the second party into the spotlight; call the
second party and allow them to join the call, which then becomes a three party
conference. The Flare for Windows device may disconnect from the conference, leaving
the remaining two parties connected for audio by selecting Moderator Controls then
Continuation. The Flare then closes the Moderator Control window and selects the
disconnect call button. The Flare should NOT select the End Conference icon in the
Moderator Controls window.
2.3. Known Issues and Limitations

2.4. Interconnectivity between Conferencing and Siemens HiPath
4000
Whilst observing the conference initially it was seen that the Siemens endpoints were
disconnecting after 5 minutes. A review of WireShark / traceSM showed the following:
A SIP INFO was sent from Conferencing via Session Manager to the Siemens HiPath
which did not respond to the request.
The SIP INFO would then be repeatedly sent by the Session Manager to the Siemens
HiPath before Session Manager finally sent a Request Timeout back to the Conferencing
Server. At this point the Siemens handset would disconnect.
Conferencing then sent a second SIP INFO via Session Manager to the Siemens HiPath
which again did not respond to the request.
Another Request Timeout was then sent back to the Conferencing Server.
A third set of SIP INFO was sent by Conferencing to the Siemens HiPath and Siemens
responded this time to the SIP INFO with 481 Call leg/Transaction Does Not Exist. This
was sent to the Conference Server which responded with a BYE to the Siemens Endpoint.
On receipt of this BYE at the Siemens HiPath, the PBX responded with another 481 Call
leg/Transaction Does Not Exist.
There is virtually no additional SIP configuration available in the Siemens HiPath 4000, other
than to configure the protocol and ports. In order to overcome this issue a recommendation was
made to change the following setting on Conferencing via Element Manager: Feature Server
Elements Application Servers AS1 Configuration Parameters. Select Parm Group
“Long Call” and edit the field Duration and set zero to disable this field. This is also detailed in
Section 8.7 when configuring the Avaya Aura Conferencing 7.0.
3. Reference Configuration
The test configuration consisted of Communication Manager, System Manager, Session
Manager, Conferencing 7.0 and Siemens HiPath 4000 V5. A variety of handsets configured as IP
and SIP devices were used.
Figure 1: Network Diagram of Avaya Aura ® Session Manager R6.3, Avaya Aura ®
Communication Manager R6.2. Avaya Aura® Conferencing 7.0 and Siemens HiPath 4000
V5.
4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Equipment Software
Avaya S8800 Media Server Avaya Aura® Conferencing 7.0 (SP2)
MCP Core Linux 15.1.7
Element Manager Console:
Management Console Version:
MCP_15.0.18.1_2012-12-02-2012
Element Manager OMI Version: 15.0.18
Avaya S8800 Media Server Avaya Aura® System Manager R6.3 GA
Software Update Revision: 6.3.0.8.923
Build 6.3.0.8.5682-6.3.8.818
VSP: 6.2.2.06002.0
Avaya S8800 Media Server Avaya Aura ® Session Manager R6.3
6.3.0.0.630039
Avaya S8800 Media Server Avaya Aura® Communication Manager R6.2
(SP4) Patch 02.0.823.0-20199
VSP: 6.2.2.06002.0
Version: 06.2-02.0.823.0
Avaya Handset 9621G SIP – S96x1_SALBR6_2_1r26_v4r70.tar
Avaya Handset 9650 SIP - SIP96xx_2_6_8_4.bin
Avaya Handset 9650 H.323 - ha96xxua3_1_05_S.bin
Avaya Handset ADVD175 Flare SIP_A715_1_1_1_019004.tar
Avaya Flare Experience for Windows Release 1.1.0.3
Siemens HiPath 4000 Siemens HiPath 4000 V5 R0.5.28
Communications Server
Siemens SIP Gateway HG3500 L0-T2R.51.000-004 / pzksti40.26.000-004
Siemens SIP Registrar HG3500 L0-T2R.51.000-004 / pzksti40.26.000-004
Siemens H.323 Gateway HG3500 L0-T2R.51.000-004 / pzksti40.26.000-004
Siemens OpenStage 15T Firmware:V1 R0.23.0 [TDM]
Siemens OpenStage 40 S Firmware:V2 R2.24.0 [SIP]
Siemens OptiPoint 420 Std Firmware:V7 R6.2.0 [SIP]
Siemens OpenStage 20 S Firmware :V2 R1.21.0 [SIP]
Siemens OptiPoint 420 Std S Firmware:V5 R6.3.0 [IP]
5. Configure Avaya Aura® Communication Manager
This section provides details on the configuration of Communication Manager. All
configurations in this section are administered using the System Access Terminal (SAT). This
section provides the procedures for configuring Communication Manager on 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 Groups
Administer Route Pattern
Administer Private Numbering
Administer Locations
Administer Dial Plan and AAR Analysis
Save Changes
The following assumptions have been made as part of this document:

It is assumed that Communication Manager, System Manager, Session Manager and
Conferencing 7.0 have been installed, received a basic configuration and have been
licensed. Refer to Section 12 Reference [5] and [6] for documentation regarding these
procedures.
Throughout this section, the administration of Communication Manager is performed
using a System Access Terminal (SAT). The commands are entered on the system with
the appropriate administrative permissions. Some administration screens have been
abbreviated for clarity.
The user has experience with administering the Avaya system via both SAT and Web
Based Management systems.
5.1. 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 Page 2 of 11

OPTIONAL FEATURES
IP PORT CAPACITIES USED

Maximum Administered H.323 Trunks: 12000 0
Maximum Concurrently Registered IP Stations: 18000 2
Maximum Administered Remote Office Trunks: 12000 0
Maximum Concurrently Registered Remote Office Stations: 18000 0
Maximum Concurrently Registered IP eCons: 414 0
Max Concur Registered Unauthenticated H.323 Stations: 100 0
Maximum Video Capable Stations: 41000 1
Maximum Video Capable IP Softphones: 18000 0
Maximum Administered SIP Trunks: 24000 40
5.2. Administer System Parameter 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 trunk-to-trunk transfer 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 1 of 19
FEATURE-RELATED SYSTEM PARAMETERS

Self Station Display Enabled? y
Trunk-to-Trunk Transfer: all
Automatic Callback with Called Party Queuing? y
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? no

DID/Tie/ISDN/SIP Intercept Treatment: attendant
Internal Auto-Answer of Attd-Extended/Transferred Calls: transferred
Automatic Circuit Assurance (ACA) Enabled? n
Navigate to page 19 of system-parameters features. Set Direct IP-IP Audio Connections and
SIP Endpoint Managed Transfer to y.
change system-parameters features Page 19 of 19

FEATURE-RELATED SYSTEM PARAMETERS
IP PARAMETERS
Direct IP-IP Audio Connections? y
IP Audio Hairpinning? n
Synchronization over IP? n
SDP Capability Negotiation for SRTP? y
SIP Endpoint Managed Transfer? y
CALL PICKUP
Maximum Number of Digits for Directed Group Call Pickup: 4
Call Pickup on Intercom Calls? y Call Pickup Alerting? n
Temporary Bridged Appearance on Call Pickup? y Directed Call Pickup? y
Extended Group Call Pickup: none
Enhanced Call Pickup Alerting? n
Use the change system-parameters ip-options command to override the ip-codec for SIP
connections. Navigate to page 4 and set Override ip-codec set for SIP direct-media
connections to y.
change system-parameters ip-options Page 4 of 4

IP-OPTIONS SYSTEM PARAMETERS
SYSLOG FROM TN BOARDS

Local Facility #: local4
Dest #1 IP address: Port #: 514

Override ip-codec-set for SIP direct-media connections? y
5.3. 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, procr and
10.10.9.127 are entered as Name and IP Address for the Communication Manager. In addition,
smvl109 and10.10.9.124 are entered for Session Manager.
change node-names ip Page 1 of 2

IP NODE NAMES
Name IP Address
clan 10.10.9.141
default 0.0.0.0
gateway 10.10.9.1
medpro 10.10.9.142
procr 10.10.9.127
procr6 ::
smvl109 10.10.9.124
5.4. 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. This domain name is also referenced in
Session Manager. (See Section 12 References [7] and [8]). Set the 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.
display ip-network-region 1 Page 1 of

20
IP NETWORK REGION
Region: 1
Location: 1 Authoritative Domain: mmsil.local
Name: To Session Manager
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 1 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 0
Audio PHB Value: 0
Video PHB Value: 0
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 0
Audio 802.1p Priority: 0
Video 802.1p Priority: 0 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
Use the change ip-codec-set n command to configure IP Codec Set paramenters where n is the
IP Codec Set number. In these Application Notes, IP Codec Set 1 was used as the main default
codec set. The standard G.711 codecs and G729 codec were selected.
Audio Codec Set for G.711MU, G.711A, G729 and G.729A as required
Silence Suppression: Retain the default value n
Frames Per Pkt: Enter 2
Packet Size (ms): Enter 20
Retain the default values for the remaining fields, and submit these changes.
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.711MU n 2 20
2: G.711A n 2 20
3: G.722-64K 2 20
4: G.729 n 2 20
5: G.729A n 2 20
6:
7:
Media Encryption
1: none
2:
3:
5.5. Create SIP Signaling Group and Trunk Group
5.5.1. SIP Signaling Group
In the test configuration, Communications Manager acts as an Evolution Server. An IMS enabled
SIP trunk is not required. The example uses signal group 151 in conjunction with Trunk Group
151 to reach the Session Manager. Use the add signaling-group n command where n is the
signaling group number being added to the system.
Group Type Set to sip
IMS Enabled Set to n
Transport Method Set to tcp
Peer Detection Enabled Set to y
Near-end Node Name Set to procr (from Section 5.3)
Near-end Listen Port Set to 5060
Far-end Node Name Set to smvl109 (from Section 5.3)
Far-end Listen Port Set to 5060
Far-end Network Region Set to 1 (From Section 5.4)
Far-end Domain Set to mmsil.local [Optional – can be left blank]
DTMF over IP Set to rtp-payload
Direct IP-IP Audio Connections Set to y
IP Audio Hairpinning Set to n
Initial IP-IP Direct Media Set to y
add signaling-group 150 Page 1 of 2

SIGNALING GROUP
Group Number: 150 Group Type: sip

IMS Enabled? n Transport Method: tcp
Q-SIP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: SM
Near-end Node Name: procr Far-end Node Name: smvl109

Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 1
Far-end Domain: mmsil.local

Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y Initial IP-IP Direct Media? y
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6
5.5.2. SIP Trunk Group
Use the command add trunk-group n to add a corresponding trunk group, where n is the trunk
group number.
Group Number Set from the add-trunk-group n command
Group Type Set as sip
Group Name Choose an appropriate name
COR Set Class of Restriction (default 1)
TN Set Tenant Number (default 1)
TAC Choose an integer value
Direction Set to two-way
Outgoing Display Set to y
Service Type Set to tie
Signaling Group Enter the corresponding Signaling group number (from
Section 5.5.1)
Number of Members Enter the number of members
display trunk-group 150 Page 1 of 21

TRUNK GROUP
Group Number: 150 Group Type: sip CDR Reports: y

Group Name: ToSessMan COR: 1 TN: 1 TAC: 150
Direction: two-way Outgoing Display? y
Dial Access? n Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method: auto
Signaling Group: 150
Number of Members: 15
Navigate to Page 3 and set Numbering Format to private.
Add trunk-group 150 Page 3 of 21

TRUNK FEATURES
ACA Assignment? n 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 120 for the Telephone Event Payload Type and P-Asserted-
Identity for Identity for Calling Party Display.
add trunk-group 150 Page 4 of 21

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
Block Sending Calling Party Location in INVITE? n
Enable Q-SIP? n
5.6. 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 to be used. Configure this route
pattern to route calls to trunk group 150, as configured in Section 5.5.2. Assign the lowest FRL
(facility restriction level) to allow all callers to use this route pattern, Assign 0 to No. Del Digits.
change route-pattern 150 Page 1 of 3

Pattern Number: 151 Pattern Name: To SM
SCCAN? n Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC
No Mrk Lmt List Del Digits QSIG
Dgts Intw
1: 150 0 0 n user
2: n user
3: n user
4: n user
5: n user
6: n user
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 M 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest next
2: y y y y y n n rest none
5.7. Administer Private Numbering
Use the change private-numbering command to define the calling party number(s) to be sent
out through the SIP trunk. The trunk group has been set to use private numbering. See Section
5.5.2. In the sample network configuration, all calls originating from a 5 digit extension
beginning with 4 will result in a 5-digit calling number. The calling party number will be in the
SIP “From” header.
[38XXX – Conferencing, 421XX – Avaya H.323, 422XX –Avaya SIP]
change private-numbering 0 Page 1 of
2
NUMBERING - PRIVATE FORMAT
Ext Ext Trk Private Total

Len Code Grp(s) Prefix Len
5 38 150 5 Total Administered: 5
5 39 150 5 Maximum Entries: 540
5 421 150 5
5 422 150 5
5.8. Administer Locations

Use the change locations command to define the proxy route to use for outgoing calls. In the
sample network, the proxy route will be the trunk group defined in Section 5.5.2.
change locations Page 1 of 1
LOCATIONS
ARS Prefix 1 Required For 10-Digit NANP Calls? y
Loc Name Timezone DST City/ Proxy Sel

No Offset Area Rte Pat
1: Main + 00:00 0 150
5.9. Administer Dial Plan and AAR Analysis
Configure the dial plan for dialing 6-digit extensions beginning with 81 to stations registered
with the Siemens. Additionally configure a dial string starting 38 to access the Conferencing
Service. Also configure dial strings to reach other Avaya Extensions (i.e. 421XX – H.323 and
422 – SIP). Use the change dialplan analysis command to define Dialed String 81 and 38 as an
aar Call Type. Configure dial string 421 and 422 as ext.
change dialplan analysis Page 1 of 12
DIAL PLAN ANALYSIS TABLE
Location: all Percent Full: 3
Dialed Total Call Dialed Total Call Dialed Total Call

String Length Type String Length Type String Length Type
1 3 dac 7 1 fac
2 5 udp 81 6 aar
24 5 ext 9 1 fac
35 5 aar * 3 fac
36 5 aar # 3 fac
38 5 aar
39990 5 ext
39995 5 aar
420 5 ext
421 5 ext
422 5 ext
423 5 ext
433 5 ext
500 5 aar
56 5 aar
Use change aar analysis 0 command to configure an aar entry for Dialed String 81XXXX to
use Route Pattern 150. Add entries for the non SIP phone extensions which begin with 422 and
use unku for call type. Add an entry for 38 to reach Conferencing.
change aar analysis 3 Page 1 of 2

AAR DIGIT ANALYSIS TABLE
Location: all Percent Full: 1
Dialed Total Route Call Node ANI

String Min Max Pattern Type Num Reqd
3 7 7 999 aar n
35 5 5 150 unku n
38 5 5 150 unku n
39990 5 5 150 unku n
39995 5 5 150 unku n
4 7 7 999 aar n
420 5 5 150 unku n
421 5 5 150 unku n
422 5 5 150 unku n
423 5 5 150 unku n
433 5 5 150 unku n
55 5 5 150 unku n
56 5 5 150 unku n
81 6 6 150 unku n
9 7 7 999 aar n
5.10. Save Changes
Use the save translation command to save all changes.
save translation
SAVE TRANSLATION
Command Completion Status Error Code
Success 0
6. Configure Avaya Aura® Session Manager
This section provides the procedure for configuring Session Manager. For further reference
documents, refer to Section 12 References [7] and [8] of this document. The procedures include
the following areas:
Log in to Avaya Aura® System Manager
Administer SIP Domains
Administer Locations
Administer Adaptations
Administer SIP Entities
Administer SIP Entity Links
Administer Time Ranges
Administer Routing Policy
Administer Dial Patterns
Administer Avaya Aura® Session Manager
Add Avaya Aura® Communications Manager as an Evolution Server
Synchronize Avaya Aura ® Communication Manager Data
Configure Avaya Extensions for Conferencing
6.1. Log in to Avaya Aura® System Manager

Configuration is accomplished by accessing the browser-based GUI of System Manager, using
the URL http://<ip-address>/SMGR where <ip-address> is the IP address of System Manager.
Login with appropriate credentials.
Once connected successfully, the home screen will be displayed.
The Home screen is divided into three sections with hyperlinked categories below.
Configuring SIP Connectivity is carried in the Routing section in under Elements.
In the main panel, a short procedure for configuring the network Routing Policies is shown.
6.2. Administer SIP Domains
SIP domains are created as part of the Avaya Aura ® System Manager basic configuration.
There will be at least one which the Session Manager is the authoritative SIP controller. In these
sample notes it is mmsil.local. The Session Manager can also deal with traffic from other
domains, so multiple SIP domain entries may be listed. Note: This Domain will also be used in
the configuration of Conferencing. (See Section 8.4).
The location of where you are currently in the system is listed at the top of the screen.
Underneath will be listed the domain(s) available in the system.
To create a new SIP Domain, from the Home (first screen available upon successful logon)
select the following; Home  Elements  Routing  Domains  Domain Management
and click New (not shown).
Name Add a descriptive name
Type Set to SIP
Notes Add a brief description in the Notes field
Click Commit to save.
6.3. Administer Locations
Session Manager uses the origination location to determine which dial patterns to look at when
routing the call if there are any dial patterns administered for specific locations. Locations are
also used to limit the number of calls coming out of or going into physical locations. This is
useful for those locations where the network bandwidth is limited. For this sample configuration,
one Location has been created which will reference both the Session Manager location and the
Siemens location. Navigate to Home  Elements  Routing  Locations. To create a new
Location, click New.
In the General section
Notes add a brief description
Leave the settings for Overall Managed Bandwidth and Pre-Call Bandwidth Parameters, as
default unless advised to do otherwise.
Note: This Location will also be used in the configuration of Conferencing. See Section 8.2)
Continue scrolling down the screen until Location Pattern is displayed as shown below. In the
Location Pattern section, under IP Address Pattern enter IP addresses used to logically
identify the location(s). Under Notes add a brief description. Click Commit to save.
In the example above, IP addresses have been entered with a (*) wildcard to indicate a range.
Click Commit to save the changes.
6.4. Administer Adaptations
Adaptations are used to manipulate digits in the SIP URI strings of incoming and outgoing calls.
For this sample configuration, no adaption is required to reformat the uri string to or from
Siemens HiPath 4000 V5.
6.5. Administer SIP Entities

Each SIP device (other than Avaya SIP Phones) that communicates with the Session Manager
requires a SIP Entity configuration. This section details the steps to create four SIP Entities for
the Siemens HiPath 4000, Session Manager, Communication Manager and Conferencing.
To create a SIP Entity for the Siemens HiPath 4000, browse to Home  Elements  Routing
 SIP Entities and click New.
In the General section,
FQDN or IP Address Add the IP Address of the target entity (Siemens HiPath
4000 V5 – use the HG35XX board IP address registered as
the SIP Gateway (See Section 7.3.2)
Type select Other
Notes Add a brief description
Adaptation Leave blank
Location Click on the drop down arrow select Location created in
Section 6.3
Time Zone Select the appropriate Time Zone
SIP Link Monitoring Set to Use Session Manager Configuration
Click Commit to save. A message will appear advising that “Entity Links can be added to the
record once the Entity has been saved”. Section 6.6 advises how to create Entity Links. To create
a SIP Entity for the Session Manager, Communication Manager and Conferencing, repeat
the above process. Screenshots are on the next page showing sample data for creating SIP
Entities for Session Manager, Communication Manager and Conferencing.
Screen shot for Session Manager SIP Entity. Change the Type to Session Manager when
programming the SIP Entity for Session Manager.
Screenshot for Communication Manager SIP Entity. Change the Type to CM when
programming the SIP Entity for Communication Manager.
Screenshot for Conferencing SIP Entity. This SIP entity requires more configuration than
standard. In the General section,
Name Add a descriptive name i.e. Conferencing 7.0
FQDN or IP Address Add the IP Address of the target entity (Conferencing
server)
Type select Conferencing
Adaptation Leave blank
Location Click on the drop down arrow select Location created in
Section 6.3
Time Zone Select the appropriate Time Zone
In SIP Link Monitoring section

SIP Link Monitoring Set to Use Session Manager Configuration
Supports Call Admission Control Set to enabled
Shared Bandwidth Manager Set to enabled
Primary Session Manager bandwidth Association
Select the relevant Session Manager SIP Entity
6.6. Administer SIP Entity Links
A SIP Trunk between a Session Manager and another telephony system is described by an Entity
Link. The next step is to create SIP Entity Links, which included the transport parameters to be
used for communications between the Session Manager and external SIP devices. SIP Entity
Links are required between Session Manager and Siemens, Session Manager and
Communications Manager and between Session Manager and Conferencing.
Create a SIP Entity Link between Session Manager and Siemens. Browse to Home 
Elements  Routing  Entity Links, click New.
Name Enter a suitable identifier e.g. SM to Siemens H4K
SIP Entity 1 Drop-down and select the appropriate Session Manager
Protocol Drop down and select UDP
Port Enter 5060
SIP Entity 2 Drop-down select the SIP Entity added previously, i.e. Siemens H4K
Port Enter 5060
Trusted Set the field as Trusted
Note: Siemens will only connect via UDP.
Create a SIP Entity Link between Session Manager and Communication Manager. Browse to
Home  Elements  Routing Entity Links. Click New.
Name Enter a suitable identifier e.g. SM to CM
SIP Entity 1 Drop-down select the appropriate Session Manager
Protocol Dropdown select TCP
Port Enter 5060
SIP Entity 2 Drop-down and select the SIP Entity added previously, i.e. CMES-
09.127
Port enter 5060
Trusted Tick the field
Create a SIP Entity Link between Session Manager and Conferencing. Browse to Home 
Elements  Routing Entity Links. Click New.
Name Enter a suitable identifier e.g. SM to Conferencing
SIP Entity 1 Drop-down select the appropriate Session Manager
Protocol Dropdown select TCP
Port Enter 5060
SIP Entity 2 Drop-down and select the SIP Entity added previously, i.e.
Conferencing
Port enter 5060
Trusted Tick the field
Once the Entity Links have been created, return to the Session Manager SIP Entity and check
to see if the Entity Links have been assigned to the SIP Entities.
6.7. Administer Time Ranges
Create a Time Range for LCR routing which defines policies will be active. To create a Time
Range, browse to Home  Elements  Routing  Time Ranges. Click New. Under Name
enter a suitable identifier. Select which Days are to be included in the Range. Set a suitable Start
Time and End Time. This will be used in configuring the Dial Plan. In Session Manager, a
default policy (24/7) is available that would allow routing to occur anytime. This was used in the
example network.
6.8. Administer Routing Policy

To complete the routing configuration, a Routing Policy is created. Routing policies direct how
calls will be routed to a system. Routing policies must be created, one for the Communications
Manager, one for Conferencing and finally one for Siemens. These are to be associated with the
Dial Patterns which will be created in the next step.
To create a Routing Policy to route traffic to Siemens, browse to Home  Elements  Routing
 Routing Polices. Click New. Under Name enter a suitable identifier. Under Notes enter
suitable description. Under SIP Entity as Destination click on Select.
From the new window that opens select the relevant SIP entity that this policy relates to and
click on the Select button to return to the Routing Policy screen.
After returning to the Routing Policy screen under Time of Day, assign a suitable time range.
A Routing Policy is also required for Communication Manager.
A Routing Policy is also required for Conferencing.
At this stage the records are missing the Dial Pattern which will be created next (Section 6.9).
6.9. Administer Dial Pattern
As one of its main functions, Session Manager routes SIP traffic between connected devices.
Dial Patterns are created as part of the configuration to mange SIP traffic routing, which will
direct calls based on the number dialed to the appropriate system. In the sample network, 5 digit
extensions beginning 421 are designated as Avaya handsets (Digital and H.323), whilst 6 digit
extension starting 81 are Siemens handsets. The 5 digit pattern starting 38 is required to reach the
Conferencing system. To create a Dial Pattern for calls to the Siemens, browse to Home 
Elements  Routing  Dial Patterns. Click New. Under Pattern enter a dial string pattern
e.g. 81 (all calls with 6 digit ext beginning with 81 will be routed to Siemens). Under SIP
Domains drop-down and select All. Under Notes enter a suitable description. Next, click on
Add in the section Originating Locations and Routing Policies.
In the Originating Locations and Routing Polices section which opens in anew window select
Apply the Selected Routing Policies to All Originating Locations. In the Routing Policies
(created in Section 6.8), select the Routing Policy to be applied. Click Select to save these
choices.
Once the screen has returned to the Dial Pattern Details screen, click Commit to save the
changes.
Dial Patterns should also be created for Avaya extensions beginning 421 (Digital and H323) and
set the Originating Location and Routing Polices, choosing the relevant Routing Policy for the
Communication Manager.
Dial Patterns should also be created for Conferencing Access Numbers beginning 38 (MeetMe
and AdHoc) and set the Originating Location and Routing Polices, choosing the relevant
Routing Policy for Conferencing.
6.10. Administer Avaya Aura® Session Manager
To complete the configuration, adding the Session Manager will provide the linkage between the
System Manager and Session Manager. On the System Manager Home screen, browse to Home
 Elements  Session Manager Session Manager Administration. On the right hand side,
under Session Manager Instances, click on New.(Not shown).
Under General:
SIP Entity name Select the names of the SIP entity added for Session Manager
Description Descriptive Comment
Management Access Point Host Name/IP
Enter the IP address of the Session Manager management
interface
Direct Routing to Endpoints
Set to Enable
Under Security Module
Network Mask Enter the network mask corresponding to the IP address of the
Session Manager
Default Gateway Enter the IP address of the default gateway for Session Manager.
Use default values for the remaining fields.
6.11. Add Avaya Aura® Communication Manager as an Evolution
Server
In order for Communication Manager to provide configuration and support to SIP Phones when
they register to Session Manager, Communication Manager must be added as an application.
6.11.1. Create a Avaya Aura® Communication Manager Instance

On the System Manager Managements screen browse to Home  Elements  Inventory 
Manage Elements. Click New. From the drop down list select Communication Manager. This
will then automatically change over to the General Screen.
On the General Screen

Name Enter a Descriptive Name for the Communication Manager
Node Enter the IP Address of the CM i.e. the procr address as shown in Section
5.3
Click on the Attributes Tab and enter detail in the following fields.
Login Login used for SAT access
Password Password used for SAT access
Confirm Password Password used for SAT access
Port Set to 5022
All other fields may be left with default settings. Click Commit to save the changes.
6.11.2. Create an Evolution Server Application
For Evolution Server support, further configuration of the Session Manager is required. Once
complete the Session Manager will support Avaya SIP phone registration. Endpoint Users are
created through the Session Manager User Management screens. Session Manager creates
corresponding stations on the Communication Manager. Configuration of the Application via
Session Manager is a two stage sequence, with the Application being created first, followed by
the Application Sequence. To configure browse to: Home  Elements  Session Manager 
Application Configuration  Applications. Click New. Under Name enter a suitable
identifier. Under SIP Entity drop-down select the SIP Entity of the Feature Server. Under
Description enter a suitable description. Click Commit to save.
To configure the Application Sequences Configuration. Browse to: Home  Elements 
Session Manager  Application Configuration  Applications Sequences. Click New.
Under Name enter a suitable identifier. Under Description enter a suitable description. From the
Available Applications section, select the + sign beside the Application that is to be added to
this sequence. Verify that the Application in this Sequence is updated correctly Click Commit
to save.
6.12. Synchronize Avaya Aura® Communication Manager Data
On the System Manager Management screen browse to Home  ElementsInventory
Synchronization  Communication System. Select the appropriate Element Name and the
select Initialize data for selected devices. Then click on Now. The Sync Status column will
show the current area being synchronized if the refresh is clicked. (Fields shown in blue boxes
below).
Note: This Process can take some time.
6.13. Configure Avaya Extensions for Conferencing
Please see Section 9 for configuring Avaya extension to utilize Conferencing as extensions may
not be configured with a Conferencing profile until after the Conferencing Server configuration
has been completed.
7. Configure Siemens HiPath 4000 V5
There are a number of tools available to the end user of the Siemens HiPath 4000 for
programming the various components of the system. These consist of either Web GUI or
Command Line Interface tools dependant on the device being programmed. Below is a short
table describing the tools and what they are used for.
Tool / Access Description

Equipment
HiPath Assistant https://<IPAddress of Web GUI – program most areas of the PBX
HiPath within this tool i.e. HG3500 Boards/Least
Communications Cost Routing / trunks/trunk groups /
Server> extensions / dial patterns
ComWin / Expert Accessed via HiPath Command Line Interface which offers the end
Access Assistant user more functionality than that available via
the web GUI HiPath Assistant, however is
quite complex to use. Mostly used by Siemens
trained engineers
Deployment Siemens proprietary Tool that allows end user to
Licensing Server* Software (requires upgrade/downgrade phone firmware or
(Note: This is an installation and change SIP to IP or vice versa using FTP
optional product additional software) Server. Provides a degree of programming on
for H4K systems) SIP devices that is not available via HiPath
Assistant.
HG3550 Boards HiPath Assistant / HG3550 boards within the HiPath 4000 can
Direct Web GUI / be initially configured onto the system via
ComWin ComWin Access. Further changes / alteration
of settings can be done either via HiPath
Assistant, ComWin access or via Direct Web
Access onto to the board itself
SIP Endpoints Via HiPath Assistant Use HiPath Assistant or ComWin to create the
or ComWin or DLS extension number. Limited further
or Direct Web GUI or configuration available via these tools.
via handset Use either Web GUI direct to IP address of
handset or register the handset with Siemens
DLS (Deployment Licensing Server) if
available for handset feature configuration.
IP Endpoints Via HiPath Assistant Use HiPath Assistant or ComWin to create
or ComWin or DLS and manage the extension number.
or Direct Web GUI or Use either Web GUI direct to IP address of
via handset handset or register the handset with Siemens
DLS (Deployment Licensing Server) if
available (limited functionality for handset
configuration).
TDM Endpoints Via HiPath Assistant Use HiPath Assistant or ComWin to create
or ComWin only and manage the extension number.
For the remaining steps in configuring the Siemens HiPath 4000, the majority of information will
be displayed using ComWin with references to the corresponding HiPath Assistant screen or
other web GUI’s for more user friendly screens.
7.1. Configuring HiPath Assistant

Before using Siemens HiPath Assistant, the client PC must be prepared. To access the web
interface for HiPath Assistant use Internet Explorer:
http://<IPAddressHiPathCommunicationServer> i.e. http://192.168.81.2. Once the screen has
opened, select Client Preparation.
The next few screens will lead the end user through the process of checking the current version
of browser, installing a suitable version of Java and relevant certificates and installing the Applet
Cache Manger. A number of restarts of the browser may be required during the process. As a
recommendation where possible, install the HiPath Assistant on a separate, older PC as the
product is quite strict about the level of Java and Internet Explorer that should be used.
The next screen will display the process to be followed to configure the PC for HiPath Assistant.
Click the Next button to continue. (Button not shown).
Verify the Client PC is running a supported internet browser. If an unsupported version is
running, please ensure a suitable version is installed, before progressing any further. Otherwise
click on Next.
Verify the Client PC has a suitable version of Java Runtime Environment Plug-In. In the image
below a warning is displayed. Listed at the bottom of the screen are Java versions which are
suitable for use with Siemens HiPath. Select ones suitable and install if required. A browser
restart will be necessary.
Return to the client preparation screens and click through until the Java screen and ensure that
suitable version has now installed. Click Next.
The next screen will verify the Siemens I&C Security CA Certificate. If an error message is
displayed, click on the Installation button to install the Security Certificate.
Follow the instructions to install the certificate and restart the browser once the process is
completed. Return to the Client Preparation screens and click Next to move through the screens
already processed.
The next three screens are informational screens and allow the user to confirm the configuration
so far through the use of diagnostics. (Screens not shown). Move through to the final step.
Installation of the Applet Cache Manager. If the Applet Cache Manager is not installed, select
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 4000 Assistant web interface. Click on Run.
The browser may need to be restarted to the next step. Verify that the Applet Cache Manager is
installed. Click on Next.
Client Preparation finished. Click Logon.
7.2. HiPath 4000 System Configuration
Once the client PC is prepared, access to the Administration applications is available. Return to
the main login page and click on the HiPath admin link, listed as the IP address of the HiPath
Communications Server.
Login to the Assistant using the engineering logon level. Please refer to your Siemens
Representative for this information.
7.2.1. HiPath Network Domain Configuration
The HiPath Assistant portal will be displayed. To check the Network Domain Configuration
expand Configuration Management  Network  Domain. Wait for the Domain screen to
load.
To search for any existing information, click on the Search button.
A new domain name is entered in the Domain field. The associated system name is listed under
the Systems tab.
Please Note: The Domain field in the HiPath has no relationship with SIP Domains configured
in Session Manager in Section 6.2 or FQDN.
Shown below are the values used in the sample configuration.
7.2.2. HiPath System Configuration
To edit the System Configuration expand Configuration Management  Network 
System. Click on Search to retrieve the current configuration. Confirm the System Type
currently in use and the version. Ensure the AMO language field is set to English. The
Dimensioning tab will show the features available to the PBX and the quantity of licenses
available and how many have been used. (Not shown below).
7.2.3. Configuring HiPath Expert Access (ComWin)

HiPath Expert Access or ComWin is a command line editor available to configure the Siemens
HiPath 4000 using AMO commands. Whilst in many instances the changes can be made via
HiPath Assistant, some changes can only be made via the Expert Access Mode. However the
AMO commands are more complex in nature, so caution is advised when using.
To configure the client PC for Expert Access select Expert Mode  HiPath 4000 HiPath
4000 Expert Access. Wait for the screen to load.
To configure the Expert Access client it is necessary to install the client software first. Click on
the link Install HiPath 4000 Expert Access Client.
Click on the link to download the Installation-File and execute comwin.exe. (Not shown)
To accept the install select Run.
Once the program has completed installation return to the Expert Access page. To launch click
on Open.
The Expert Access console will be launched. It will open a number of screens, which should not
be closed. The last screen to be opened will be the console screen itself.
Note: the configuration details shown below illustrate the running set-up used for these
Application Notes. AMO command DISPLAY was used to list the current configuration. AMO
commands CHANGE and ADD would be used to edit or add these settings on the system.
Additionally a command tool is available to build the commands. To use this tool, from the
console window select Edit  MML Editor. The MML Editor will open.
The AMO command is built in the following way:

Choose the COMMAND – this relates to the area of the HiPath 4000 to be configured
i.e. Station, Trunk, Trunk Group, Board etc., What is selected in the first field will affect
the choices in the second field.
Choose the ACTION – i.e. ADD, DELETE, EXECUTE, REGEN
The third field may vary, dependant on the COMMAND chosen. Where possible, hover
the mouse key over the blank field to show a brief tool tip explaining the field. If the field
offers a select button (as shown in the example below), click on this for more detail
before making a selection. Fields in bold are mandatory as part of the command creation.
Once the command is complete, it can be submitted to the console screen by pressing
Enter on the keyboard.
Press Enter again to run the command. The console will return the results of the
command.
The boards list for this sample configuration are:
Board Location IP Address Function

/ PEN
1-1 N/A LTU1 ISDN Admin Link – Links Cabinet 1 to HiPath 4000
1-1-1 192.168.81.3 SIP Gateway Phone Registrar
[Siemens OptiPoint / OpenStage SIP Endpoints ]
1-1-2 N/A TDM Interface [Siemens Digital Phones]
1-2 N/A LTU2 ISDN Admin Link– Links Cabinet 2 to HiPath 4000
1-2-1 192.168.81.4 H.323 GW [Siemens OptiPoint IP Phones]
1-2-10 192.168.81.10 SIP GW SIP Trunks
7.3. Configuring SIP Trunking
Unless the Siemens is interconnecting to another Siemens system, it is recommended that SIP be
configured as “native SIP trunks” rather than using SIP-Q.
7.3.1. Configure LEGK

Configure the Siemens as a Large Enterprise Gate Keeper. In HiPath Assistant go to
Configuration Management Network System. Click on Search. On the Base Data tab
check the field Large Enterprise Gate Keeper.
7.3.2. Configure the HG3500 Board

In this example the Common Gateway Board Q2324-X500 (STMI4) is configured as the
Gateway to connect with the Avaya Aura ® Session Manager. This board will be configured for
IP trunking. There are a number of smaller steps to be carried out in adding a board.
7.3.2.1 Define Function Block and Add the Board

The first step is to define a FUNCTION BLOCK which determines the board function, number
of B-channels and then to add the board to the system. In HiPath Assistant got to Configuration
Management System Data Board CGW Function Block. Click New and configure the
following:
Function block Set to 1 if not already in use, else a max. of 2 digits.
Board Type Set to Traditional CGW Board
Dedicates the block for boards with
o Enable 60 b-channels
o Enable 120 b-channels
CGW Functionalities
o Enable HG3550
Click on Save at this stage to commit the changes made so far. A message should appear stating
“Block Created”. This will then make further fields available for configuration.
o Number of Lines Set to 1
o Number of Predefined Blocks Set to 3
Click on Save at this stage to commit the changes made so far. A message should appear stating
“Block Data Changed”. This will then make further fields available for configuration. Finally
tick the field Finish configuration of this function block and click Save again.
Next Stage is to add the board to the system. In HiPath Assistant go to Configuration
Management System Data Board Board. Click on New. Fill out the following fields:
LTU Set to n where n is the relevant cabinet in the system

SLOT Set to n to indicate the first empty slot in the cabinet
Part Number Click drop-down and select the part number that describes the
board.
Function ID Click the drop-down and select the function ID – usually 1
Category Set to IPGW
Board Name Should fill in automatically, based on previous field settings.
CGW Function Block Select the Function Block created in Section 7.3.2.1
Description Enter a description to describe the function of the board.
Click on the tab CGW Functionalities
IP trunking (HG3550) Enter number of trunks i.e. 30
Click on the tab STMI2-IGWBoard Data
Customer LAN Address Enter the IP Address to be assigned to the card
Subnet Mask Enter the subnet mask
Trunk Protocol: SIP Enter the number of trunks i.e. 30
Click Save at this stage.
After clicking Save, a warning message may appear, advising to restart the board. For the
moment continue with the configuration.
Click Search Criteria and locate the board just added to continue the configuration. This will
add some more tabs:
Click on the tab STMI Feature Access

Codec Settings
o Adjust codecs to desired settings and enable VAD on G711A/U
The HG3500 boards are administered using one gateway number. This number must be unique.
Click on the tab STMI2-IGW Board Data
Large Enterprise Gatekeeper Data
Gateway Number Set to a unique 1 or 2 digit number i.e
Default Gateway IP Address Enter the default gateway for the network
To configure without authentication and registration check the following tabs and fields:
On the tab STMI2-IGW Board Data
GW Registration at External Gate Keeper Clear the tick
(Field shown in previous screen shot)
On the tab STMI2-IGW Management Data
SIP Trunking for ERH
Digest Authentication is required Clear the tick
SIP Trunking for SSA
Register Gateway at SIP Registrar Clear the tick
On the tab PKI/SPE
DNS Server IP Address Enter a suitable IP Address for DNS Server
When saving changes, the following message may appear:
The system will highlight the tabs where there are issues and highlight the fields where settings
need attending to before the save can be completed.
In this example, clear the field TCP Port of its entry. (This will be re-added automatically after
the save.). On the STMI2-IGW Board Data tab, clear the 0 value from the field SIPQ (not
shown) and click the Save button again.
The following message may appear;
At this stage the board should be restarted. This should be done using the AMO Command
RESTART-BSSU:ADDRTYPE=PEN,LTG=1,LTU=1,SLOT=4;
The screen shot below shows a sample restart of a board and the results of the restart.
7.3.2.2 Configure Trunk Group and Trunks
The next stage is to create a trunk group and assign trunks to the trunk group. In HiPath Assistant
got to Configuration Management System Data Trunk Trunk Group. Click on New.
Fill out the following fields:
Trunk Group Enter a trunk group number
Name Enter a name for the trunk group
Max. No, of Lines Enter how many lines for the trunk group
Click Save
Next add a Digital Trunk to the trunk group. In HiPath Assistant got to Configuration
Management System Data Trunk Trunk. Click on New. Fill out the following fields:
PEN Enter the PEN number from the board created in Section 7.3.2
Trunk Type Set to Digital
B-Channel Grp Set to 1
Device Set to HG3550IP
Base Data Tab
Trunk Group Set to the trunk group previous created i.e. 400
Trunk Name Set a suitable trunk name
COT Number Set the Class of Trunk number. See Section 7.10
COP Number Set the Class of Parameter number. See Section 7.9
COS Number Set the Class of Service number. See Section 7.11
Node Number Set a Fictitious Node Number
DPLN Group Set the Dial Plan Group to 0 (default)
ITR Group Set the Internal Traffic Restriction Group to 0 (default)
Digital Extension Tab

Protocol Set to ECMAV2 for CorNet-NQ Issue 2.1 Dec. 96 basis ETSI
Paging Mode Set to DSC (Linear- Descending)
Digital ISDN Tab
Ensure B-Channels are enabled
7.3.2.3 Configure Trunk Access Code

Before you can configure trunk access codes, you must create the corresponding Dial Codes.
Then, when you configure the Trunk Access Codes window:
The check boxes in the section Dial Plan Group Number must be configured
with the same value entered into the DPLN field in the Dial Codes window.
The CPS check boxes in the CPS - Overwrite section must be
checked/unchecked in the same way they were checked/unchecked in the
CPS/DPLN tab of the Dial Codes window.
First configure the Trunk Access Code number in Dial Codes. Dial Codes are used to program
Feature Access Codes as well as the Digit pattern to reach the Avaya. In HiPath Assistant,
browse to Configuration Management  Tables  Dial Plan Dial Codes. Configure the
following fields:
Dial Code Set to a suitable number
Code Type Set to NETRTE (Auxiliary Dial Access Route) for networking
Next create the Trunk Access Code. In HiPath Assistant, browse to Configuration
Management  System Data  Trunk Trunk Access Codes. Click New and enter the
following.
Access Code Enter the Dial Code created in the previous step
Reference Access Code Enter the Dial Code created in the previous step
Position Enter 1
Trunk Group Enter the trunk group created in Section 7.3.2.2
Voice Enable field
Data Terminal Enable field
Facsimile Enable field
Dial Plan Group Number Should be left blank and should correspond to the blank
field DPLN in the Dial Codes screen in the previous step
CPS-Overwrite Enable 0-Auxiliary CPS for DAR NETRTE and should
correspond to the field CPS 0 in the Dial Codes screen in
the previous step.
Save the changes.
7.3.3. Configuring the Gate Keeper

Further configuration must be made to the Gate Keeper to indicate the board capabilities. These
settings are NOT available via the GUI screen, so should enabled using an AMO command.
Open a ComWin session (See Section 7.2.3). In the Console window:
First check the status of the Gate Keeper with the command
DISP-GKREG:GWNO=<N>; where N is the Gate Keeper number programmed on the board.

To check what number is being used go to Configuration Management System Data
Board Board – search for the board created in Section 7.3.2.1 and click on the STMI2-
IGW Board Data tab.
To change the attributes of the Gate Keeper to recognize SIP and IP calls use the following
command:
ADD-GKREG:GWNO=40,GWATTR=INTGW&HG3550V2&SIP,DIPLNUM=0,DPLN=0,LAUTH=1,
INFO="LOCAL GW",SECLEVEL=TRADITIO;
Re run the command DISP-GKREG:GWNO=40; The output should be similar to this:
<DISP-GKREG:GWNO=40;
DISP-GKREG:GWNO=40;
H500: AMO GKREG STARTED
+-----------------------------------------------------------------------------+
| GWNO 40 GWATTR INTGW HG3550V2 SIP |
| GWIPADDR 192.168.81 .10 GWDIRNO |
| DIPLNUM 0 DPLN 0 |
| LAUTH 1 |
| GATEWAY REGISTERED: NO |
| IP GATEWAY IS CONFIGURED BY GKREG |
| INFO:LOCAL-GW |
| SECLEVEL: TRADITIO |
+-----------------------------------------------------------------------------+
AMO-GKREG-111 GATEKEEPER REGISTRY

DISPLAY COMPLETED;
7.4. HG 3500 SIP Gateway Configuration via Web Interface
As a final step to configuring the SIP Gateway, the board must also be configured via a web GUI
to confirm SIP settings and to add a SIP Trunk Profile to link to the Avaya. To access the
configuration of the HG3500 Gateway’s, use the Web Console interface https://IPAddress-Of-
HGGW/ and Login screen will appear.
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.
7.4.1. 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. Changes may be saved initially by scrolling to the bottom of the screen and
clicking the Apply button. After applying changes, check the SAVE symbol at the bottom of the
screen. If it shows red, click to save the changes. If the green activity dot shows red, then a reset
of the board is required. Click the RESET icon. Only the following fields are configurable via
these screens.
Section Field Name

SIP Server (Registrar / Redirect) Period of registration (sec)
RFC 3261 Timer Values Transaction Timeout (msec)
SIP Transport Protocol SIP via UDP
SIP Session Timer RFC 4028 support
SIP Session Timer Session Expires (sec)
SIP Session Timer Minimal SE (sec)
DNS-SRV Records Blocking time for unreachable destination(sec)
Outgoing Call Supervision MakeCallReq Timeout (sec)
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 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.
7.4.2. SIP Registrar Gateway

This is the card used by the handsets when they register. The board is added in a similar way to
the Gateway. The image below shows 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.
7.4.3. 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.
7.5. Least Cost Routing
A diallable route from the Siemens HiPath 4000 to the Avaya Aura ® Session Manager must be
inserted into the Siemens LCR tables before a Siemens 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 of dialing 42XXX to reach extensions on the Avaya.
7.5.1. 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 (0-15)
CPS (0-22) Call Progress State. Various states to determine how the call is handled.
7.5.2. 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 Management 
Least Cost Routing  Digit Pattern
Digit Pattern Digit Pattern of the number being dialed. The X indicates
the number of digits to follow
Dial Plan Number Default is 0 meaning the number is assigned to all dial plans
Digit Counter The values of the digit counter assigned to the digit pattern
displayed
DPLN 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".
Route Contains the number of the LCR ROUTE. (See next Section)
Auth.Code (1-64) 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 Authorization, refer to Section 7.5.6
7.5.3. Configure LCR Route
The LCR Route determines the trunk group used. Configuration Management  Least Cost
Routing  LCR Route.
Dest. Node Fictitious Number representing the far end PBX
Route Assign a Route number
Trunk Group Trunk Group number from Section 7.3.2.2
7.5.4. Configure LCR Route Element
The LCR Route Element carries further settings regarding the LCR Route used. Configuration
Management Least Cost Routing LCR Route Element.
Route Route Number
Name Route Name
Outdial Rule Rule for how the number is dialed See Section 7.5.5
Tr.Grp.No Trunk Group Number. See Section 7.3.2.2
7.5.5. Configure Outdial Rule
The Outdial Rule details how the number is dialed. Configuration Management  Least Cost
Routing  Outdial Rule.
Outdial Rule Identifying Number for the rule
Outdial Rule Elements Sequential series of steps to determine 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 out-pulse all the digits of a specified field.
The Parameter indicates which field to apply this to. END
is mandatory on all Outdial rules.
7.5.6. Configure LCR Authorizations
When entering the Digit Pattern (Section 7.5.2), authorizations were assigned to the Digit
Pattern. Configuration Management  Least Cost Routing  Digit Pattern. Each digit
pattern created can belong to between 1-64 Authorization 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, where by a different COS/LCRCOS may be used.
Each LCRCOS is assigned LCR Authorization Code(s), controlling the numbers a user can dial.
The LCR COS is accessed via Configuration Management  Least Cost Routing  LCR
Authorizations  LCR Class of Service – Voice.
LCOS Voice Identifying number for LCOSvoice
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 42-XXX was created, it was assigned Authority Code 1. Any subscriber that uses a
LCR COS Voice, with Authority Code 1 enabled is now permitted to dial number starting 42.
7.6. Siemens Station End-Points
The Siemens Stations can be managed from the HiPath Assistant. To check/edit the Stations,
expand Configuration Management  Station  Station. Wait for the config screen to
display. Click on Search to access the current configuration.
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
The image below illustrates the configuration of a SIP Station Endpoint.
PEN. Must match the Card location described in previous section (1-
1-1 is the SIP Registrar Interface Card)
The image below illustrates the configuration of a H.323 Station Endpoint.
PEN. Must match the Card location described in previous section. (1-
2-1 is the H.323 GW Interface Card)
7.6.1. Configuration of H.323 and SIP Endpoints via Web Interface

H.323 endpoints may be configured via Hi Path Assistant, Com Win or via the handsets own
web interface. SIP endpoints have limited programming functionality from HiPath Assistant or
ComWin, but can be programmed from DLS or via the phone web interface. Both H.323 and IP
can also be programmed from the handset interface itself as well.
To access the Web GUI for H.323 handsets browse to http://<ip_address_of_phone>:8085

To access the Web GUI for SIP handset browse to https://<ip_address_of_phone>
You will be prompted for a password to enter user or admin menus – this is usually 123456.
Handsets may be factory reset from the web interface.
Screenshot from H.323 web interface
Screen shot from SIP web interface
To access the menus on the handsets directly:
OpenStage – press the Menu key (indicated with 3 lines). Select Admin and enter the password.
Press OK. The screen offers a “tree” structure to its menu system. Common areas are Network
for administering network IP addresses / DHCP and System to control SIP server related settings
and extension number.
OptiPoint (configured as SIP) – leave the handset at rest and press  key until screen displays
05-SETUP?. Press . Press 6 (a hidden admin config menu) and enter the password 123456 and
then . Use the arrows keys to scroll between menus. Common menus are 01-Network and 02-
System.
OptiPoint (configured as H.323) – leave the handset at rest and press 1,0,3 keys at the same
time. If successful, the screen will display Administration/01=Configuration. Press to enter
and enter the password 123456. Common menus are 01-Network and 02-System.
7.7. Enable Direct Media Connection and BCHANL on Siemens

Endpoints
It may be necessary when configuring Siemens endpoints to configure Direct Media Connect and
BCHANL to ensure the handsets function correctly. The easiest way to do this is via the
ComWin console. Use the following command:
CHANGE-SDAT:STNO=810012,TYPE=ATTRIBUT,AATTR=DMCALLWD&MBCHL;
Use the command DISPLAY-SDAT:STNO=810012; to show the details of the station.
DISPLAY-SDAT:STNO=810012;
H500: AMO SDAT STARTED
---------------------------- SUBSCRIBERDATA ----------------------------
STNO = 810012 COS1 = 32 DPLN = 0 SSTNO = NO
PEN : 1- 1- 1- 1 COS2 = 32 ITR = 0 TRACE = NO
DVCFIG : S0PP LCOSV1 = 7 COSX = 0 ALARMNO = 0
AMO : SBCSU LCOSV2 = 7 SPDI = 0 RCBKB = NO
LCOSD1 = 7 SPDC1 = RCBKNA = NO
KEYSYS : LCOSD2 = 7 SPDC2 =
------------------------------------------------------------------------
CDRACC =
SRCGRP = 1 TCLASS =
CLASSMRK = EC G711 G729AOPT
PUBNUM = TON = NPI =
NNO = 1 -69 -1 HOTIDX =
STNOOOS = MVHFAIP = NO
STNOAPE = AMOALTRT = NO
GWIPADR : CLUSTID =
SVCDOM =
PRECLEV =
BWLIDX =
CLASSSEC = SECURE
------------------------------- ATTRIBUTES -----------------------------
KN VC DMCALLWD MBCHL
------------------------------------------------------------------------
7.8. Class of Parameter
This window assigns specific line parameters to a number. The line parameters are used by the
device handler in the HiPath 4000. There are two tabs: the functionality of which is described as
follows:
● Parameters - This tab defines the needed trunk parameters.
● Access Authorization - This tab defines the trunk access rights and the toll access rights.
In HiPath Assistant browse to Configuration Manager  System Data  Trunk Class of
Parameter.
DISPLAY-COP:COPNO=202,FORMAT=L;
H500: AMO COP STARTED
COP: 202 INFO: COP FUER SIP ANSCHALTNG AVAYA

DEVICE: S2CONN SOURCE: DB
PARAMETER:
REGISTRATION OF LAYER 3 ADVISORIES L3AR
CO TRUNK ACCESS:
TRUNK ACCESS TA
TOLL ACCESS:
TRUNK ACCESS TA
7.9. Class of Trunk
This window assigns specific trunk parameters to a number, and are used for the call
processing line tables in HiPath 4000. In HiPath Assistant browse to Configuration Manager
 System Data  Trunk Class of Trunk.
DISPLAY-COT:COTNO=203,FORMAT=L;
H500: AMO COT STARTED
COT: 203 INFO:

DEVICE: INDEP SOURCE: DB
PARAMETER:
PRIORITY FOR AC WILL BE DETERMINED FROM MESSAGE PRI
RECALL IF USER HANGS UP IN CONSULTATION CALL RCL
TRUNK CALL TRANSFER XFER
TRUNK SIGNALING ANSWER ANS
KNOCKING OVERRIDE POSSIBLE KNOR
CALL EXTEND FOR BUSY, RING OR CALL STATE CEBC
NETWORKWIDE AUTOMATIC CALLBACK ON BUSY CBBN
NETWORKWIDE AUTOMATIC CALLBACK ON FREE CBFN
NETWORKWIDE CALL FORWARDING PERMITTED FWDN
NETWORKWIDE FORWARDING NO-ANSWER FNAN
DON'T RELEASE CALL TO BUSY HUNT GROUP BSHT
END-OF-DIAL FOR BLOCK IS SET BLOC
SEND NO NODE NUMBER TO PARTNER LWNC
ACTIVATE TRANSIT COUNTER ADMINISTRATION FOR S0/S2 LINE ATRS
INCOMING CIRCUIT FROM SYSTEM WITHOUT LCR NLCR
TSC-SIGNALING FOR NETWORKWIDE FEATURES (MANDATORY) TSCS
TRUNK SENDS CALL CHARGES TO ORIGINATING NODE NUMBER TRSC
USE DEFAULT NODE NUMBER OF LINE DFNN
INCOMING CIRCUIT FROM SYSTEM WITHOUT LCR (DATA) NLRD
SEND NO BILLINGELEMENTS SNBE
IGNORE INCOMING BILLING IICB
B-CHANNEL NEGOTIATION (PREV. PREFERRED-PREFERRED COLLISION) BCNE
CORNET-NQ PICKUP-INFO NOT SUPPORTED NQPI
NO FLAG TRACE NOFT
CORNET-NQ NETWORKWIDE PICK-UP NOT SUPPORTED NQNP
NETWORK CALL TRANSFER, EXPLICIT CALL TRANSFER NCT
NO SIMPLE DIALOG AVAILABLE NOSD
DON'T SEND CINT LEG2,IF CENTR. ATND IN HETERO. NETWORK NIN2
PARTNER NODE DOES NOT KNOW HIPATH GEP SIGNALLING PGEP
NO TONE NTON
AMO-COT -111 CLASS OF TRUNK FOR CALL PROCESSING

DISPLAY COMPLETED;
<
7.10. Class of Service

Classes of service can be allocated to either individual stations or multiple stations. COS
is also allocated lines for inter-PBX traffic. In HiPath Assistant, browse to Configuration
Management  Tables  COS.
DISPLAY-COSSU:TYPE=COS,COS=32,FORMAT=L;
H500: AMO COSSU STARTED
+------+-----------------+---------------+-----------------+
| COS | VOICE | FAX | DTE |
+------+-----------------+---------------+-----------------+
| 32 |>32 OPTI/SET600 MIT RUFWEITERSCHALTUNG (RWS) |
| | TA | TA | TA |
| | TSUID | TNOTCR | TNOTCR |
| | TNOTCR | BASIC | BASIC |
| | CDRS | MSN | MSN |
| | CDRSTN | MULTRA | MULTRA |
| | CDRC | | |
| | CDRINT | | |
| | COSXCD | | |
| | MB | | |
| | CFNR | | |
| | VCE | | |
| | FWDNWK | | |
| | TTT | | |
| | MSN | | |
| | CFB | | |
| | CFSWF | | |
| | FWDECA | | |
| | FWDEXT | | |
| | CW | | |
| | SUTVA | | |
+------+-----------------+---------------+-----------------+
AMO-COSSU-111 CLASSES OF SERVICE

DISPLAY COMPLETED;
Additional Class of Service settings may be set on an extension only basis on the Station tab.
In HiPath Assistant, browse to Configuration Management  Station  Station.
7.11. Altering Codec Settings
It may be necessary to alter the codec settings on the SIP registrar or other HG3500 boards.
Although the codecs appear to be configurable from the HG3500 web interface and from HiPath
Assistant, the most reliable place to change these settings is via AMO using ComWin.
To display the current codec settings: DISPLAY-CGWB:LTU=2,SLOT=10,TYPE=ASC;
<DISPLAY-CGWB:LTU=2,SLOT=10,TYPE=ASC;
DISPLAY-CGWB:LTU=2,SLOT=10,TYPE=ASC;
H500: AMO CGWB STARTED
------------------------------------------------------------------------------
| CGW BOARD DATA |
------------------------------------------------------------------------------
| HG3550 |
------------------------------------------------------------------------------
| LTU = 2 SLOT = 10 SMODE = NORMAL POOLNO: 0 |
------------------------------------------------------------------------------
ASC DATA - CONFIGURABLE VALUES:
-------------------------------
TOSPL = 184 (184) TOSSIGNL = 104 (104)
UDPPRTLO = 29100 (29100) UDPPRTHI = 29580 (29219)
T38FAX = YES (YES) REDRFCTN = YES (YES)
RFCFMOIP = YES (YES) RFCDTMF = YES (YES)
PRIO1 : CODEC = G711U VAD = YES RTP-SIZE = 20

PRIO2 : CODEC = G711A VAD = YES RTP-SIZE = 20
PRIO3 : CODEC = G729 VAD = NO RTP-SIZE = 20
PRIO4 : CODEC = G729A VAD = NO RTP-SIZE = 20
PRIO5 : CODEC = G723 VAD = NO RTP-SIZE = 30
PRIO6 : CODEC = NONE VAD = NO RTP-SIZE = 20
PRIO7 : CODEC = G729AB VAD = YES RTP-SIZE = 20
AMO-CGWB -111 CONFIGURATION OF HG3500 BOARD

DISPLAY COMPLETED;
To disable a codec
CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=3,TYPE=ASC,PRIO=PRIO1,CODEC=NONE;
To enable a codec
CHANGE-
CGWB:MTYPE=CGW,LTU=1,SLOT=3,TYPE=ASC,PRIO=PRIO1,CODEC=G711U,VAD=YES,RTP=20;
After making a change to a codec, please log onto the board via the Web Browser and save the
changes. See Section 7.4.
7.12. Other System Settings
For timers related to the Siemens system, use the AMO command CTIME – filter CP1 and CP2
relate to the timers used by the endpoints and TRK relates to trunk timers.
DISPLAY-CTIME:TYPESWU=CP1;
H500: AMO CTIME STARTED
CALL PROCESSING TIMERS1:
+-----------------------+-----------------------+-----------------------+
| USERGD2 = 2 SEC. | USERGD5 = 5 SEC. | LONGPATH = 2 SEC. |
| CBK = 20 SEC. | ANSADV = 20 SEC. | PARK = 30 SEC. |
| SPDPD = 5 SEC. | RECALL = 30 SEC. | CDRDISDE = 10 SEC. |
| ECCSSUPV = 3 SEC. | FAXCD = 5 SEC. | DCIEOD = 5 SEC. |
| AUTOTONE = 200 MS. | RELREC = 90 SEC. | TSUPVCB = 2 SEC. |
| ONHOOK = 15 MIN. | TSUPVAUT = 30 SEC. | CBRLEN = 40 SEC. |
| KNLEN = 30 SEC. | RELCON = 90 SEC. | LCREOD = 10 SEC. |
| RECNIGHT = 180 SEC. | RECPARK = 120 SEC. | PINVO1 = 300 SEC. |
| PINVO2 = 300 SEC. | PINVO3 = 300 SEC. | PINVO4 = 300 SEC. |
| PINVO5 = 300 SEC. | PINVO6 = 300 SEC. | PINVO7 = 300 SEC. |
| PINNV1 = 1800 SEC. | PINNV2 = 1800 SEC. | PINNV3 = 1800 SEC. |
| PINNV4 = 1800 SEC. | PINNV5 = 1800 SEC. | PINNV6 = 1800 SEC. |
| PINNV7 = 1800 SEC. | DISPCH = 20 SEC. | OFLPM = 20 SEC. |
| ATDLYANN = 12 SEC. | DANNDIAL = 15 SEC. | ANNDIAL = 15 SEC. |
| LCRET = 5 SEC. | NOTRNG = 5 SEC. | FAXOFLPM = 30 SEC. |
| ROUTEREQ = 5 SEC. | CTHLDREC = 45 SEC. | UUS3T1 = 10 SEC. |
| UUS3T2 = 10 SEC. | UUS3T3 = 10 SEC. | QDELESTA = 1 SEC. |
+-----------------------+-----------------------+-----------------------+
AMO-CTIME-111 CUSTOMER-SPECIFIC CP TIMERS, SWITCHING UNIT
DISPLAY COMPLETED;
DISPLAY-CTIME:TYPESWU=CP2;
CALL PROCESSING TIMERS2:
+-----------------------+-----------------------+-----------------------+
| CCBS5 = 3600 SEC. | AOCT1 = 15 SEC. | AOCT2 = 15 SEC. |
| CTLS1 = 2 SEC. | CTLS2 = 2 SEC. | GPUNST = 180 SEC. |
| DCPA1 = 10 SEC. | DCPA2 = 15 SEC. | DCPA3 = 10 SEC. |
| RTAD = 0 SEC. | PINVO10 = 300 SEC. | PINNV10 = 1800 SEC. |
| T400 = 20 SEC. | PREDCALL= 20 SEC. | NIRECALL= 900 SEC. |
| TRKTOTRK= 120 SEC. | REPAUSE = 3 SEC. | SYSRCPAR= 30 SEC. |
| AUTCMPON= 60 SEC. | PINVO11 = 300 SEC. | PINNV11 = 1800 SEC. |
| PINVO12 = 300 SEC. | PINNV12 = 1800 SEC. | PINVO13 = 300 SEC. |
| PINNV13 = 1800 SEC. | PINVO14 = 300 SEC. | PINNV14 = 1800 SEC. |
| PINVO15 = 300 SEC. | PINNV15 = 1800 SEC. | VARDIALT= 20 SEC. |
| FTRQTOUT= 10 SEC. | FTWOTOUT= 180 SEC. | GWNAVAIL= 60 SEC. |
| SMPFTIME= 8 SEC. | EARLYDMC= 900 MSEC. | |
+-----------------------+-----------------------+-----------------------+
AMO-CTIME-111 CUSTOMER-SPECIFIC CP TIMERS, SWITCHING UNIT
DISPLAY COMPLETED;
DISPLAY-CTIME:TYPESWU=TRK;
CCT/SET TIMERS:
+-----------------------+-----------------------+-----------------------+
| PREEMT = 90 SEC. | NOANSCON = 30 SEC. | NOCALLTR = 30 SEC. |
| ATNDTRNS = 30 SEC. | RPTSZ = 5 SEC. | RPTSZEND = 40 SEC. |
| PRREQI = 30 SEC. | PRREQC = 15 SEC. | NEWCONN = 110 SEC. |
| RELJOIN = 900 SEC. | | |
+-----------------------+-----------------------+-----------------------+
System wide settings may be found in the ZAND and ZANDE AMO commands (DISPLAY-
ZAND; and DISPLAY-ZANDE;. There are many sub commands associated with these two
commands and it is beyond the scope of this document to cover them in detail. However from
previous tests, cross-PBX transfers were failing. Under further investigation, the setting
EXCOCO was set to YES and cross-PBX transfers could be performed successfully. This
setting can be viewed in DISPLAY-ZAND:TYPE=ALLDATA;.
DISPLAY-ZAND:TYPE=ALLDATA;
H500: AMO ZAND STARTED
GENERAL SYSTEM DATA:
====================
TRANSFER = EXTEND , ALERTN = NO ,
AUTHUP = TA ,
RNGBKTN = NO , TRANSINH = NO ,
NIGHT = TA ,
ITRFWD = NO , HOLDTN = RA , ANATESIG = TONE ,
DSSLT = 10, CODTN = NO , CONFSUB = YES ,
DATEDIS = DDMM, CNTRYCD = 0 , RCLLT = NO ,
MELODY = 1, TRCD = , CPBLOWL = 80 ,
CPBUPPL = 100, CUTHRU1A = NO , PREDIA = NO ,
SIUANN = 1, CO = NO , COEXN = 0 ,
CBKNO = 5 , SEVDIG = NO , PNNO = 1 -69 -999,
DISPMODE = MODE1, PNODECD = 900999, ROUTOPTP = NO ,
ROUTOPTD = NO , CALLOFF = NO , PARARING = YES ,
DSSDEST = NO , ONEPARTY = YES , MSGDELAY = NO ,
EXCOCO = YES , TRDGTPR = NO , COANN = NO ,
HOTDIAL = NO , TRANSTOG = NO , NOCFW = NO ,
HOLDHUNT = NO , POSTDDLY = NO , EXBUSYOV = NO ,
OVRMST = NO , OVRHUNT = NO , CONITPRO = YES ,
RECHUNT = NO , CALLACMP = NO ;
8. Configure Avaya Aura® Conferencing 7.0
This section provides a sample configuration for Conferencing 7.0 Co-resident Simplex. It is
assumed that the basic installation and configuration has already taken place using the Intelligent
Workbook provided for Conferencing 7.0. For further information on Conferencing 7.0 please
consult the references in Section 12. The procedures below include the following areas:
Add Meet Me and Adhoc URIs to Provision Client
Add a Location
Assign Media Server Clusters to Locations
Add System Manager Domains to Provisioning Client
Configuring a Web Conferencing Server
Configuring a Web Conferencing Host
Additional Configuration of Conferencing for Siemens HiPath 4000 Interconnectivity
Configuration of Conferencing is performed via System Manager after Single Sign On has been
enabled for the Conferencing server. For further information on Conferencing 7.0 please consult
the references in Section 12.
8.1. Add Meet Me and Adhoc URIs to Provision Client

Service URI is the dialing access number for participants to join an Avaya Aura ® Conferencing
MeetMe Conference. The Service URI number matches the routing dial pattern that was
previously configured on the Dial Pattern Details page (see Section 6.9). The Service URI is the
username part of the SIP URI string.
Access the System Manager using a web browser and from the Home screen select Elements 
Conferencing.
Within the Conferencing Dashboard, click on the Conferencing Provisioning Service already
created e.g., AAC-PROV.
If prompted, enter the User Name and Password that allows access to the Provisioning Client and
then click on Login.
In the Provisioning Client window, select System Management  Routing  Service URI.
In the Service URI field, enter an access number for a MeetMe conference. This number must
match the digit pattern that was previously configured in System manager (See Section 6.9).
From the Locale box, select the appropriate locale. The locale specifies the default locale of the
prompts that are used if the SIP client does not provide a locale. From the Conference Type
field select MeetMe. Click Save to commit the changes.
Repeat the same process to create a Service URI for an Adhoc Conference. This number must
also be a match for digit pattern created in Section 6.9.
8.2. Add a Location
For any Location that has been configured on System Manager it must also be added manually
to the Provisioning Client. Add a location using the following procedure. In the Provisioning
Client window, select System Management  Routing  Locations.
Click on Add Location (not shown).

In the Location Name box enter the name of the location. This is case-sensitive and must match
the location name programmed in System Manager (see Section 6.3). Click Save to commit the
changes.
If there is more than one location programmed in System Manager, repeat for each location.
8.3. Assign Media Server Clusters to Locations
Use the following procedure to assign a media service cluster to serve conference calls for a
particular location. In the Provisioning Client window, select System Management  Routing
 Media Server Resources.
Select the Media Server Serving Locations tab. From the Media Server Cluster box, select the
Media Server Cluster 1 to associate it with a location. In the Add Locations area, select the
check box of the location to associate the selected media server cluster. Press Save.
Next, select the Media Server Physical Location tab. From the Select By list, select Location.
From the Select Physical Location list, select the Location to which to assign a media server
cluster. From the Available Media Server Clusters box, select the appropriate media server
cluster, and then click Copy. Press Save.
8.4. Add System Manager Domains to Provisioning Client

Manually add all the System Manager domains into Provisioning Client. In the Provisioning
Client window, select System Management  System Manager Domains.
In the System Manager Domain box, enter the name of the domain and click Add. Repeat for
each System Manager domain created in Section 6.2 i.e. mmsil.local.
Note: Conferencing 7.0 needs to determine the location of the caller for a call to succeed. This
information will be displayed in the caller invite; location information is stored against domains
therefore caller domains must be administered on the system.
A second domain should also be added for the Siemens HiPath 4000. As the Siemens is not
configurable with a FQDN, use the IP address of the Siemens SIP Gateway HG35XX board.
8.5. Configuring a Web Conferencing Server
User may chose to utilise the Collaboration Agent web interface to host and manage their
conferences. The Web Conferencing Server requires configuring in the Provision Client. In the
Provisioning Client window, select System Management  Routing  Web Conferencing
Server Resources.
Select the Web Conferencing Server Groups tab. From the Location box select the location to
which to associate web conferencing server resources. From the Primary Web Conference
Server list select the appropriate web conferencing server administered during initial
Conferencing Server installation. Click Save.
8.6. Configuring a Web Conferencing Host
In the Provisioning Client window, select System Management  Routing  Web
Conferencing Host.
In the Host Name field, enter the Fully Qualified Domain Name (FQDN) for the Web
Conferencing Service administered during installation i.e., cavl81.mmsil.local. Press Save.
Note: Systems external to Conferencing must be able to translate the Web Conferencing Server
FQDN into an IP address for routing. Contact a network engineer to configure entries into the
corporate DNS servers.
8.7. Additional Configuration of Conferencing for Siemens HiPath
4000 Interconnectivity
Additional configuration of Avaya Aura® Conferencing is required to ensure Siemens endpoint

connectivity. This requires changes to be made to the Conferencing Element Manager. Access
the System Manager using a web browser and from the Home screen select Elements 
Conferencing.
In the Conferencing Dashboard that appears click on the link to the Element Manager.
Click on the link Launch Element Manager Console (not shown). A Java applet will initiate
and the first Element Manager Console box will appear.Click on Connect
Accept the certificate for this session only (not shown). Enter a suitable username and password.
From the menu tree on the left hand side expand Feature Server Elements Application
Servers AS1 Configuration Parameters. If the Configuration Parameter box does not
appear on the right hand side, double click Configuration Parameters to make it appear. Within
the AS1 Configuration Parameters window, change the Parm Group field to LongCall. Front
he list of fields, highlight the field Duration and click on the -/+ button to edit the field. Set the
duration to 0 (zero) and Apply the changes.
9. Administer Avaya Extensions for Conferencing.
The Avaya extensions must be administered with a Conferencing Profile to provide the
individual user with Moderator and Participant pin numbers that may be used by others joining
the relevant conference. This document assumes that user extension has already been created and
will concentrate on configuring the Conferencing Profile. The example endpoint used is a SIP
phone, but details are given later on regarding H.323 endpoint configuration for Conferencing
Profile.
9.1. Configure SIP phone with Conference Profile

On the System Manager screen select Users, and then select User Management Manage
Users. Select the extension to be modified and click Edit (not shown).
On the Communication Profile tab enter scroll down the screen until Conferencing Profile is
shown. Click the tick field next to Conferencing Profile -this will automatically expand the
section.
In the Location select the relevant location. To chose a template first click on the Get
Templates button. Then from the Template field, choose a relevant template i.e.
desktop_user_no_video. Click Commit (not shown).
The addition of the profile to the extension should automatically create a user in the Provisioning
Client. Log onto the Provisioning Client. In the Provisioning Client window, select User
Management  Search Users.
In the search window that appears, search for the extension that the profile has been assigned to.
Click on the Search button. Once the extension has been found, click on the hyperlink for the
extension.
To confirm the pin numbers for Moderator and Participant have been assigned to the extension,
click on the Actions tab and then select Conferencing.
Once on the conferencing tab, the Participant and Moderator codes generated by the system will
be displayed. Click Save.
9.2. Configure H.323 phone with Conference Profile
The process for configuring an Avaya H.323 endpoint with a Conferencing Profile is very
similar as for a SIP endpoint. The record for the H.323 device must be added to the System
Manager, in the same way as for a SIP device, however the endpoint must have an E.164 entry
(rather than a SIP entry) as a Communication Address on the Communication Profile screen:
10. Verification Steps
10.1. Verification of Siemens HiPath 4000 V5

The HiPath 4000 Communications Server is the central controlling unit in the Siemens PBX set-
up. Connection the HG3500 gateway chassis is via ISDN link. Check the ISDN link light on the
front panel of the HG3500 chassis to verify that it displays “green” link light. Connection to the
HG3500 IP gateways is via LAN. Verify that the link light on the front of the cards for the LAN1
connection is a “bright green”. (Screen shot below shows lights on for LAN1, but lights off for
LAN2 as it is not in use.) Using a PC on the same network, verify ping tests to the SIP Trunk
GW, the H.323 GW and the SIP Registrar GW. Each of the Siemens gateway’s can be accessed
via web browser. Use the Front Panel tab to view link status and line status. The image below
shows the Front Panel status for the SIP Trunk gateway. LAN link status is displayed on the
bottom left. Channel status is listed on the right, in this example a single call from Siemens to
Avaya is active, indicated by the green indicator.
10.2. Verification of Avaya Endpoints
To confirm functionality of Avaya endpoints with the Siemens HiPath 4000, a number of tools
are available for testing and confirmation of status.
Handset Registration can be confirmed via System Manager. Go to the Home screen and
select Elements  Session Manager  System Status  User Registrations
Successful SIP Entity connectivity can be viewed via the Session Manager Dashboard.
The field Entity Monitoring indicates how many links are down / total number of links. If
the field is red and/or there is indication of a link being down, troubleshoot to determine
the cause.
SIP tracing on Session Manager can be carried out via the traceSM tool.
WireShark traces may also be utilized to monitor across the network as well.
10.3. Verification of Avaya Conferencing
The MeetMe conference can be verified by dialing users in to a Conference. Dial in one
participant as a Moderator and the remaining users as Participants. Open a Collaboration Agent
window to view the conference.
11. Conclusion
The Siemens HiPath 4000 V5 functions with Avaya Aura® Conferencing 7.0 in conjunction
with Avaya Aura ® Session Manager R6.3 for basic call functionality and call features.
12. Additional References
Product documentation for Avaya products may be found at http://support.avaya.com
Avaya Aura ® Conferencing 7.0
[1] Deploying Avaya Aura® Conferencing Release 7.0 Issue 4 October 2012
[2] Administering Avaya Aura® Conferencing Release 7.0 Issue 3 October 2012
[3] Avaya Aura® Conferencing Alarms and Logs Reference Version 7.0 Issue 2 October 2012
[4] Intelligent Workbook for Avaya Aura ® Conferencing 7.0
Avaya Aura ® Communication Manager
[5] Avaya Aura® Communication Manager Screen Reference Release 6.203-602878 Issue 3.0
December 2012
[6] Administering Avaya Aura® Communication Manager Release 6.2 03-300509 Issue 7.0
December 2012
Avaya Aura ® System Manager / Avaya Aura ® Session Manager
[7] Administering Avaya Aura® System Manager Release 6.3 December 2012 Issue 1.0
[8] Administering Avaya Aura® Session Manager Release 6.3 December 2012
Product Documentation for Siemens Products may be found the following websites and service
manuals.
[9] Siemens Wiki Website http://wiki.siemens-enterprise.com/index.php/Main_Page

[10] Siemens eTAC (a support website with limited information available to customer/end user)
http://siemenstac.custhelp.com/app/home
[11] HiPath 4000 V5 IP Solutions, SIP Connectivity Service Documentation ref: A31003-
H3150-S104-2-7620 Dec 2008.
[12] Other Service Documentation may be obtained via your Service Support Provider or
Account Manager within Siemens Enterprise Communications.
©2013 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya Solution &
Interoperability Test Lab at interoplabnotes@list.avaya.com

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Avaya Solution & Interoperability Test Lab

Application Notes for configuring Siemens HiPath 4000

2.1. Test Description and Coverage

2.2. Test Results and Observations

2.2.1. Dial Out functionality

2.2.2. Unattended Transfer / Cross PBX Unattended Transfer / Attended

2.3. Known Issues and Limitations

The following assumptions have been made as part of this document:

display system-parameters customer-options Page 2 of 11

IP PORT CAPACITIES USED

5.2. Administer System Parameter Features

change system-parameters features Page 1 of 19

FEATURE-RELATED SYSTEM PARAMETERS

Music (or Silence) on Transferred Trunk Calls? no

change system-parameters features Page 19 of 19

change system-parameters ip-options Page 4 of 4

SYSLOG FROM TN BOARDS

Dest #1 IP address: Port #: 514

Override ip-codec-set for SIP direct-media connections? y

change node-names ip Page 1 of 2

5.4. Administer IP Network Region and Codec Set

display ip-network-region 1 Page 1 of

change ip-codec-set 1 Page 1 of 2

Audio Silence Frames Packet

add signaling-group 150 Page 1 of 2

Group Number: 150 Group Type: sip

Near-end Node Name: procr Far-end Node Name: smvl109

Far-end Domain: mmsil.local

display trunk-group 150 Page 1 of 21

Group Number: 150 Group Type: sip CDR Reports: y

Navigate to Page 3 and set Numbering Format to private.

Add trunk-group 150 Page 3 of 21

Numbering Format: private

Replace Restricted Numbers? n

Modify Tandem Calling Number: no

Show ANSWERED BY on Display? y

add trunk-group 150 Page 4 of 21

Mark Users as Phone? n

Convert 180 to 183 for Early Media? n

5.6. Administer Route Pattern

change route-pattern 150 Page 1 of 3

Ext Ext Trk Private Total

5.8. Administer Locations

ARS Prefix 1 Required For 10-Digit NANP Calls? y

Loc Name Timezone DST City/ Proxy Sel

Dialed Total Call Dialed Total Call Dialed Total Call

change aar analysis 3 Page 1 of 2

Dialed Total Route Call Node ANI

Command Completion Status Error Code

6.1. Log in to Avaya Aura® System Manager

Once connected successfully, the home screen will be displayed.

Configuring SIP Connectivity is carried in the Routing section in under Elements.

Click Commit to save.

6.5. Administer SIP Entities

In SIP Link Monitoring section

Note: Siemens will only connect via UDP.

6.8. Administer Routing Policy

6.11.1. Create a Avaya Aura® Communication Manager Instance

On the General Screen

Note: This Process can take some time.