Abstract. Testing was conducted via the Internal Interoperability Program at the Avaya Solution and Interoperability Test Lab.

Transcription

1 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. 1 of 117

2 Table of Contents Table of Contents 1. Introduction Interoperability Testing Test Description and Coverage Test Results and Observations Dial Out functionality Unattended Transfer / Cross PBX Unattended Transfer / Attended Transfer by Flare Devices Known Issues and Limitations Interconnectivity between Conferencing and Siemens HiPath Reference Configuration Equipment and Software Validated Configure Avaya Aura Communication Manager Verify Avaya Aura Communication Manager License Administer System Parameter Features Administer IP Node Names Administer IP Network Region and Codec Set Create SIP Signaling Group and Trunk Group SIP Signaling Group SIP Trunk Group Administer Route Pattern Administer Private Numbering Administer Locations Administer Dial Plan and AAR Analysis Save Changes Configure Avaya Aura Session Manager 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 Pattern of 117

3 6.10. Administer Avaya Aura Session Manager Add Avaya Aura Communication Manager as an Evolution Server Create a Avaya Aura Communication Manager Instance Create an Evolution Server Application Synchronize Avaya Aura Communication Manager Data Configure Avaya Extensions for Conferencing Configure Siemens HiPath 4000 V Configuring HiPath Assistant HiPath 4000 System Configuration HiPath Network Domain Configuration HiPath System Configuration Configuring HiPath Expert Access (ComWin) Configuring SIP Trunking Configure LEGK Configure the HG3500 Board Configuring the Gate Keeper HG 3500 SIP Gateway Configuration via Web Interface SIP Trunk Gateway SIP Registrar Gateway H.323 Gateway Least Cost Routing Configure Dial Codes Configure Digit Pattern Configure LCR Route Configure LCR Route Element Configure Outdial Rule Configure LCR Authorizations Siemens Station End-Points Configuration of H.323 and SIP Endpoints via Web Interface Enable Direct Media Connection and BCHANL on Siemens Endpoints Class of Parameter Class of Trunk Class of Service Altering Codec Settings Other System Settings Configure Avaya Aura Conferencing 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 of 117

4 8.5. Configuring a Web Conferencing Server Configuring a Web Conferencing Host Additional Configuration of Conferencing for Siemens HiPath 4000 Interconnectivity Administer Avaya Extensions for Conferencing Configure SIP phone with Conference Profile Configure H.323 phone with Conference Profile Verification Steps Verification of Siemens HiPath 4000 V Verification of Avaya Endpoints Verification of Avaya Conferencing Conclusion Additional References of 117

5 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 ondemand 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. 5 of 117

6 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 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 and Section 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. 6 of 117

7 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 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 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. 7 of 117

8 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 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. 8 of 117

9 4. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Equipment Avaya S8800 Media Server Avaya S8800 Media Server Software Avaya Aura Conferencing 7.0 (SP2) MCP Core Linux Element Manager Console: Management Console Version: MCP_ _ Element Manager OMI Version: Avaya Aura System Manager R6.3 GA Software Update Revision: Build VSP: Avaya S8800 Media Server Avaya Aura Session Manager R Avaya S8800 Media Server Avaya Aura Communication Manager R6.2 (SP4) Patch VSP: Version: Avaya Handset 9621G SIP S96x1_SALBR6_2_1r26_v4r70.tar Avaya Handset 9650 SIP - SIP96xx_2_6_8_4.bin Avaya Handset 9650 H ha96xxua3_1_05_s.bin Avaya Handset ADVD175 Flare SIP_A715_1_1_1_ tar Avaya Flare Experience for Windows Release Siemens HiPath 4000 Siemens HiPath 4000 V5 R Communications Server Siemens SIP Gateway HG3500 L0-T2R / pzksti Siemens SIP Registrar HG3500 L0-T2R / pzksti Siemens H.323 Gateway HG3500 L0-T2R / pzksti Siemens OpenStage 15T Firmware:V1 R [TDM] Siemens OpenStage 40 S Firmware:V2 R [SIP] Siemens OptiPoint 420 Std Firmware:V7 R6.2.0 [SIP] Siemens OpenStage 20 S Firmware :V2 R [SIP] Siemens OptiPoint 420 Std S Firmware:V5 R6.3.0 [IP] 9 of 117

10 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. 10 of 117

11 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: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: Maximum Concurrently Registered Remote Office Stations: Maximum Concurrently Registered IP econs: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: 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 11 of 117

12 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 Dest #2 IP address: Port #: 514 Dest #3 IP address: Port #: 514 Override ip-codec-set for SIP direct-media connections? y 12 of 117

13 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 are entered as Name and IP Address for the Communication Manager. In addition, smvl109 and are entered for Session Manager. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address clan default gateway medpro procr procr6 :: smvl 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: P/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 13 of 117

14 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 Codec Set: 1 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : G.711A n : G K : G.729 n : G.729A n : 7: Media Encryption 1: none 2: 3: 14 of 117

15 5.5. Create SIP Signaling Group and Trunk Group 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 Peer Detection Enabled? y Peer Server: SM Enforce SIPS URI for SRTP? y 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 15 of 117

16 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 16 of 117

17 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 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: 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 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 3: y y y y y n n rest none 4: y y y y y n n rest none 5: y y y y y n n rest none 6: y y y y y n n rest none 17 of 117

18 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 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 Total Administered: Maximum Entries: 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 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: of 117

19 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 ext aar ext ext ext ext ext 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 aar n unku n unku n unku n unku n aar n unku n unku n unku n unku n unku n unku n unku n unku n aar n 19 of 117

20 5.10. Save Changes Use the save translation command to save all changes. save translation SAVE TRANSLATION Command Completion Status Error Code Success 0 20 of 117

21 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 where <ip-address> is the IP address of System Manager. Login with appropriate credentials. Once connected successfully, the home screen will be displayed. 21 of 117

22 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. 22 of 117

23 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. 23 of 117

24 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 Name Add a descriptive name 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) 24 of 117

25 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. 25 of 117

26 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 V 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. 26 of 117

27 To create a SIP Entity for the Siemens HiPath 4000, browse to Home Elements Routing SIP Entities and click New. In the General section, Name FQDN or IP Address Type Notes Adaptation Location Time Zone SIP Link Monitoring Add a descriptive name 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) select Other Add a brief description Leave blank Click on the drop down arrow select Location created in Section 6.3 Select the appropriate Time Zone 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. 27 of 117

28 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. 28 of 117

29 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 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 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 Click Commit to save. 29 of 117

30 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. 30 of 117

31 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 Notes Add a brief description 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 Port enter 5060 Trusted Tick the field Notes Add a brief description 31 of 117

32 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 Notes Add a brief description 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. 32 of 117

33 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 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. 33 of 117

34 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. Click Commit to save. 34 of 117

35 A Routing Policy is also required for Communication Manager. 35 of 117

36 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). 36 of 117

37 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. 37 of 117

38 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. 38 of 117

39 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. 39 of 117

40 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 Description Select the names of the SIP entity added for Session Manager 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. 40 of 117

41 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 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 of 117

42 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. 42 of 117

43 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. 43 of 117

44 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. 44 of 117

45 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 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. 45 of 117

46 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 / Equipment HiPath Assistant ComWin / Expert Access Deployment Licensing Server* (Note: This is an optional product for H4K systems) Access of HiPath Communications Server> Accessed via HiPath Assistant Siemens proprietary Software (requires installation and additional software) HG3550 Boards HiPath Assistant / Direct Web GUI / ComWin SIP Endpoints IP Endpoints TDM Endpoints Via HiPath Assistant or ComWin or DLS or Direct Web GUI or via handset Via HiPath Assistant or ComWin or DLS or Direct Web GUI or via handset Via HiPath Assistant or ComWin only Description Web GUI program most areas of the PBX within this tool i.e. HG3500 Boards/Least Cost Routing / trunks/trunk groups / extensions / dial patterns Command Line Interface which offers the end user more functionality than that available via the web GUI HiPath Assistant, however is quite complex to use. Mostly used by Siemens trained engineers Tool that allows end user to upgrade/downgrade phone firmware or change SIP to IP or vice versa using FTP Server. Provides a degree of programming on SIP devices that is not available via HiPath Assistant. HG3550 boards within the HiPath 4000 can be initially configured onto the system via 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 Use HiPath Assistant or ComWin to create the extension number. Limited further configuration available via these tools. 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. Use HiPath Assistant or ComWin to create and manage the extension number. Use either Web GUI direct to IP address of handset or register the handset with Siemens DLS (Deployment Licensing Server) if available (limited functionality for handset configuration). Use HiPath Assistant or ComWin to create and manage the extension number. 46 of 117

47 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 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: i.e. 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). 47 of 117

48 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. 48 of 117

49 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. 49 of 117

50 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 plugin which is required to run HiPath 4000 Assistant web interface. Click on Run. 50 of 117

51 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. 51 of 117

52 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. 52 of 117

53 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. 53 of 117

54 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. 54 of 117

55 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) 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. 55 of 117

56 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. 56 of 117

57 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. 57 of 117

58 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. 58 of 117

59 Once the command is complete, it can be submitted to the console screen by pressing Enter on the keyboard. 59 of 117

60 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 Communications Server SIP Gateway Phone Registrar [Siemens OptiPoint / OpenStage SIP Endpoints ] N/A TDM Interface [Siemens Digital Phones] 1-2 N/A LTU2 ISDN Admin Link Links Cabinet 2 to HiPath 4000 Communications Server H.323 GW [Siemens OptiPoint IP Phones] SIP GW SIP Trunks 60 of 117

61 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 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 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 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 Board Type Set to 1 if not already in use, else a max. of 2 digits. Set to Traditional CGW Board 61 of 117

62 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. 62 of 117

63 CGW Function Block Select the Function Block created in Section 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. 63 of 117

64 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 64 of 117

65 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 65 of 117

66 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. 66 of 117

67 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 Name Max. No, of Lines Click Save Enter a trunk group number Enter a name for the trunk group Enter how many lines for the trunk group 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 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 of 117

68 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 Paging Mode Set to ECMAV2 for CorNet-NQ Issue 2.1 Dec. 96 basis ETSI Set to DSC (Linear- Descending) 68 of 117

69 Digital ISDN Tab Ensure B-Channels are enabled 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 69 of 117

70 Reference Access Code Enter the Dial Code created in the previous step Position Enter 1 Trunk Group Enter the trunk group created in Section 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 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 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; 70 of 117

71 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 GWDIRNO DIPLNUM 0 DPLN 0 LAUTH 1 GATEWAY REGISTERED: NO IP GATEWAY IS CONFIGURED BY GKREG INFO:LOCAL-GW SECLEVEL: TRADITIO AMO-GKREG-111 DISPLAY COMPLETED; GATEKEEPER REGISTRY 71 of 117

72 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 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. 72 of 117

73 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) 73 of 117

74 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 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. 74 of 117

75 To view registered SIP clients select Explorers Voice Gateway Clients SIP 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. 75 of 117

76 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 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 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 76 of 117

77 Route Auth.Code (1-64) in the Route and LCR Auth fields. The value "0-15" is visible in the DPLN field if an assignment of this type applies "for all DPLN groups". Contains the number of the LCR ROUTE. (See next Section) 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 of 117

78 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 of 117

79 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 Tr.Grp.No Trunk Group Number. See Section of 117

80 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. 80 of 117

81 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. 81 of 117

82 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 of 117

83 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. 83 of 117

84 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. ( 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 84 of 117

85 The image below illustrates the configuration of a SIP Station Endpoint. Station No. Selected from drop down list PEN. Must match the Card location described in previous section (1-1-1 is the SIP Registrar Interface Card) Device Combination. Select from Drop down list Device Family. Selected from Drop down list 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 85 of 117

86 The image below illustrates the configuration of a H.323 Station Endpoint. Station No. Selected from drop down list PEN. Must match the Card location described in previous section. (1-2-1 is the H.323 GW Interface Card) Device Combination. Select from Drop down list Device Family. Selected from Drop down list 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 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 To access the Web GUI for SIP handset browse to You will be prompted for a password to enter user or admin menus this is usually Handsets may be factory reset from the web interface. 86 of 117

87 Screenshot from H.323 web interface Screen shot from SIP web interface 87 of 117

88 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 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 Common menus are 01-Network and 02-System 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 = COS1 = 32 DPLN = 0 SSTNO = NO PEN : 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 = HOTIDX = STNOOOS = MVHFAIP = NO STNOAPE = AMOALTRT = NO GWIPADR : CLUSTID = SVCDOM = PRECLEV = BWLIDX = CLASSSEC = SECURE ATTRIBUTES KN VC DMCALLWD MBCHL of 117

89 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 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 CO TRUNK ACCESS: TRUNK ACCESS TOLL ACCESS: TRUNK ACCESS L3AR TA TA 89 of 117

90 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 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 RECALL IF USER HANGS UP IN CONSULTATION CALL TRUNK CALL TRANSFER TRUNK SIGNALING ANSWER KNOCKING OVERRIDE POSSIBLE CALL EXTEND FOR BUSY, RING OR CALL STATE NETWORKWIDE AUTOMATIC CALLBACK ON BUSY NETWORKWIDE AUTOMATIC CALLBACK ON FREE NETWORKWIDE CALL FORWARDING PERMITTED NETWORKWIDE FORWARDING NO-ANSWER DON'T RELEASE CALL TO BUSY HUNT GROUP END-OF-DIAL FOR BLOCK IS SET SEND NO NODE NUMBER TO PARTNER ACTIVATE TRANSIT COUNTER ADMINISTRATION FOR S0/S2 LINE INCOMING CIRCUIT FROM SYSTEM WITHOUT LCR TSC-SIGNALING FOR NETWORKWIDE FEATURES (MANDATORY) TRUNK SENDS CALL CHARGES TO ORIGINATING NODE NUMBER USE DEFAULT NODE NUMBER OF LINE INCOMING CIRCUIT FROM SYSTEM WITHOUT LCR (DATA) SEND NO BILLINGELEMENTS IGNORE INCOMING BILLING B-CHANNEL NEGOTIATION (PREV. PREFERRED-PREFERRED COLLISION) CORNET-NQ PICKUP-INFO NOT SUPPORTED NO FLAG TRACE CORNET-NQ NETWORKWIDE PICK-UP NOT SUPPORTED PRI RCL XFER ANS KNOR CEBC CBBN CBFN FWDN FNAN BSHT BLOC LWNC ATRS NLCR TSCS TRSC DFNN NLRD SNBE IICB BCNE NQPI NOFT NQNP 90 of 117

91 NETWORK CALL TRANSFER, EXPLICIT CALL TRANSFER NO SIMPLE DIALOG AVAILABLE DON'T SEND CINT LEG2,IF CENTR. ATND IN HETERO. NETWORK PARTNER NODE DOES NOT KNOW HIPATH GEP SIGNALLING NO TONE NCT NOSD NIN2 PGEP NTON AMO-COT -111 DISPLAY COMPLETED; < CLASS OF TRUNK FOR CALL PROCESSING 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. 91 of 117

92 DISPLAY-COSSU:TYPE=COS,COS=32,FORMAT=L; H500: AMO COSSU STARTED COS VOICE FAX DTE >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 DISPLAY COMPLETED; CLASSES OF SERVICE 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. 92 of 117

93 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 HG LTU = 2 SLOT = 10 SMODE = NORMAL POOLNO: ASC DATA - CONFIGURABLE VALUES: TOSPL = 184 (184) TOSSIGNL = 104 (104) UDPPRTLO = (29100) UDPPRTHI = (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 DISPLAY COMPLETED; CONFIGURATION OF HG3500 BOARD 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 of 117

94 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; H500: AMO CTIME STARTED 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; H500: AMO CTIME STARTED 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 94 of 117

95 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 = , DISPMODE = MODE1, PNODECD = , 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 ; 95 of 117

96 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 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. 96 of 117

97 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. 97 of 117

98 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 of 117

99 99 of 117

100 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. 100 of 117

101 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. 101 of 117

102 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 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. 102 of 117

103 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 As the Siemens is not configurable with a FQDN, use the IP address of the Siemens SIP Gateway HG35XX board. 103 of 117

104 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. 104 of 117

105 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. 105 of 117

106 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 106 of 117

107 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. 107 of 117

108 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 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). 108 of 117

109 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. 109 of 117

110 Once on the conferencing tab, the Participant and Moderator codes generated by the system will be displayed. Click Save. 110 of 117

111 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: 111 of 117

112 10. Verification Steps Verification of Siemens HiPath 4000 V5 The HiPath 4000 Communications Server is the central controlling unit in the Siemens PBX setup. 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. 112 of 117

113 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. 113 of 117