Abstract. Avaya Solution & Interoperability Test Lab

Transcription

1 Avaya Solution & Interoperability Test Lab Front-Ending Avaya Communication Server 1000E 6.0 with an AudioCodes Mediant 1000 Modular Media Gateway to Support SIP Trunks to Avaya Aura Session Manager 6.0 Issue 1.0 Abstract These Application Notes present a sample configuration that uses an AudioCodes Mediant 1000 Modular Media Gateway as a PRI-QSIG/SIP gateway to connect Avaya Communication Server 1000E 6.0 with Avaya Aura Session Manager 6.0, which in turn provides call routing support to other Avaya SIP products such as Avaya Aura Communication Manager 6.0 and Avaya Modular Messaging. For the sample configuration, Session Manager runs on an Avaya S8800 Server, Communication Manager runs on duplex S8800 Servers with a G650 Media Gateway, and Avaya Communication Server 1000E runs on a CP+PM Co-resident server blade. The results in these Application Notes should be applicable to other Avaya servers and media gateways. 1 of 88

2 (This page intentionally left blank) 2 of 88

3 1 Introduction Equipment and Software Validated High level List of Tasks: Configuring Avaya Aura Communication Manager Evolution Server Verify System Capabilities and Licensing Add Node Name of Avaya Aura Session Manager Configure IP Network Regions Add SIP Signaling Group Add SIP Trunk Group Administer Private Numbering Administer Call Routing Configure Stations Verify Off Pbx Telephone Station Mapping Configure Voice Messaging Hunt Group Configure Voice Messaging Coverage Path Configure Coverage Path for Telephone Users Save Translations Configure Avaya Aura Session Manager Administer SIP Domains Define Locations Add SIP Entities Create Entity Links Add Routing Policies Add Dial Patterns Define Communication Manager Evolution Server as an Administrable Entity Configure Avaya Modular Messaging Administering Modular Messaging Verify MultiSite Configuration Verify Port Settings Administer a PBX Administer Sites of 88

4 6.6 Administer Subscribers Configure Communication Server Launch Element Manager Verify Equipped Feature Packages Administer TMDI Card Administer D Channel Administer Routes and Trunks Administer Route List Block Administer Distant Steering Code Enable TMDI Card Enable D channel Automatic Establishment Administer Station Users Configure AudioCodes Mediant Assign IP address Access Web Configuration Interface Configure PSTN Trunk Settings Configure SIP Protocol Parameters Configure Routing Tables Configure PSTN Trunk Group Configure Voice Mail Parameters Verification Steps Verify Communication Manager Verify Session Manager Verify Communication Server Verify AudioCodes Mediant Verification Scenarios Conclusion Additional References of 88

5 1 Introduction Previous Avaya Application Notes [11, 12] described how Release 6.0 of Avaya Communication Server 1000E (CS1000E) can be directly integrated with Session Manager using SIP trunks as well as with the CS1000E front-ended with an AudioCodes Mediant While an all-sip solution is a supported configuration, there are also many installations of the CS1000E which are not licensed for SIP. In these cases, an effective solution is to front-end the CS1000E with a PRI-QSIG/SIP gateway, which then signals on SIP trunks to Session Manager. This configuration supports basic and supplementary call features as well as RFC 2833 DTMF and message-waiting signaling for applications such as voice messaging. Figure 1 shows a sample configuration that uses an AudioCodes Mediant 1000 Modular Media Gateway to front-end the CS1000E via a T1/PRI QSIG connection. The Mediant 1000 supports SIP trunks to the SIP Entity interface of Session Manager, which in turn performs call routing to Communication Manager and Modular Messaging. Session Manager can support flexible intersystem call routing based on dialed number, calling number and system location, and can also provide protocol adaptation to allow multi-vendor systems to interoperate. It is managed by a separate Avaya Aura System Manager, which can manage multiple Session Managers by communicating with their management network interfaces. Modular Messaging expands the capabilities and features of messaging services. Centralized messaging enables the local Modular Messaging system to provide voic service to subscribers at both sites in a multisite configuration via SIP trunking. For the sample configuration, Session Manager and System Mananger each run on an Avaya S8800 Server, Communication Manager runs on duplex S8800 Servers with a G650 Media Gateway, and Communication Server 1000E runs on a CP+PM Co-resident server blade These Application Notes should apply to other Avaya servers and Media Gateways. As shown in Figure 1, the Avaya 9600 Series IP Telephones (SIP & H.323) and 2420 Digital telephone are supported by Communication Manager. The CS1000E supports 3903/4 Digital telephones and 1100 Series IP telephones (UNIStim). A seven digit Uniform Dial Plan (UDP) is used for dialing between systems. Unique extension ranges are associated with Communication Manager (666xxxx) and CS1000E (777xxxx). Session Manager routes calls based on this seven digit plan. Modular Messaging (MM) utilizes an 11 digit normalized dial plan with xxxx to represent subscribers on Communication Manager and xxxx to represent subscribers on the CS1000E. Furthermore, MM contains the necessary administration to convert between the 7-digit extensions and their 11-digit mailboxes. These Application Notes will focus on configuration of the QSIG trunks, SIP trunks, dial plan support, call routing, and call coverage for voice messaging. Detailed administration of the endpoint telephones will not be described (see the appropriate documentation listed in Section 11). 5 of 88

6 Figure 1 Sample Configuration 6 of 88

7 2 Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Hardware Component Software Version Avaya S8800 Server (2) Avaya Aura Session Manager 6.0 Avaya Aura System Manager 6.0 Duplex Avaya S8800 Servers with G650 Media Gateway Avaya Aura Communication Manager Evolution Server 6.0 SP1 Avaya 9630 IP Telephone (H.323) 3.0 Avaya 9630 IP Telephone (SIP) Avaya 2420 Digital (DCP) Telephone - Modular Messaging 5.2 Single Avaya S8510 Server Server w/multisite MAS:5.2 SP4 MSS: 5.2 SP4 AudioCodes Mediant 1000 Modular Media Gateway 6.00A Avaya Communication Server 1000E CP+PM Call Server Signaling Server CS1000E 3903 & 3904 Digital Telephone NA CS1000E 1140E UNIStim (IP) telephone C7J 3 High-level List of Tasks: 1) Configure Avaya Aura Communication Manager Evolution Server to communicate with Session Manager using the SIP protocol. 2) Configure Communication Manager, Mediant 1000 and Modular Messaging-MAS as SIP Entities in Avaya Aura Session Manager via Avaya Aura System Manager. 3) Configure SIP routing in Avaya Aura Session Manager. 4) Configure Communication Manager in Avaya Aura System Manager as an administrable element. 5) Configure a QSIG PRI trunk on the CS1000E to communicate with the Mediant ) Configure the Mediant 1000 to communicate with the CS1000E over the QSIG-PRI trunk. 7) Configure the Mediant 1000 to communicate with Session Manager using SIP. 8) Configure Modular Messaging as a centralized Voice Mail Server for the CS1000E and Communication Manager. 7 of 88

8 4 Configuring Avaya Aura Communication Manager Evolution Server This section describes the administration of Communication Manager using a System Access Terminal (SAT). Alternatively, some of the station administration could be performed using the Communication System Management application on System Manager. These instructions assume the G650 Media Gateway is already configured on Communication Manager. Some administration screens have been abbreviated for clarity. Verify System Capabilities and Communication Manager Licensing Administer network regions Administer IP node names Administer IP interface Administer SIP trunk group and signaling group Administer route patterns Administer numbering plan Add a Voice Messaging Hunt Group Administer Voice Messaging Coverage-path After completing these steps, the save translation command should be performed. 4.1 Verify System Capabilities and Licensing This section describes the procedures to verify the correct system capabilities and licensing have been configured. If there is insufficient capacity or a required feature is not available, contact an authorized Avaya sales representative to make the appropriate changes. 8 of 88

9 4.1.1 SIP Trunk Capacity Check Issue the display system-parameters customer-options command to verify that an adequate number of SIP trunk members are licensed for the system as shown below: 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: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP econs: 0 0 Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable Stations: 0 0 Maximum Video Capable IP Softphones: 0 0 Maximum Administered SIP Trunks: AAR/ARS Routing Check Verify that the ARS and ARS/AAR Dialing without FAC options are enabled (on page 3 of system-parameters customer-options). display system-parameters customer-options Page 3 of 11 OPTIONAL FEATURES A/D Grp/Sys List Dialing Start at 01? n CAS Main? n Answer Supervision by Call Classifier? n Change COR by FAC? n ARS? y Computer Telephony Adjunct Links? y ARS/AAR Partitioning? y Cvg Of Calls Redirected Off-net? y ARS/AAR Dialing without FAC? y DCS (Basic)? y ASAI Link Core Capabilities? y DCS Call Coverage? Enable Private Networking and Uniform Dialing Plan Use the display system-parameters customer-options command to verify that Private Networking and Uniform Dialing Plan are enabled as shown below: 9 of 88

10 display system-parameters customer-options Page 5 of 11 OPTIONAL FEATURES Multinational Locations? y Multiple Level Precedence & Preemption? n Multiple Locations? y Personal Station Access (PSA)? y PNC Duplication? n Port Network Support? n Posted Messages? n Private Networking? y Processor and System MSP? y Processor Ethernet? y Wireless? y Station and Trunk MSP? y Station as Virtual Extension? y System Management Data Transfer? n Tenant Partitioning? n Terminal Trans. Init. (TTI)? y Time of Day Routing? n TN2501 VAL Maximum Capacity? y Uniform Dialing Plan? y Usage Allocation Enhancements? y Wideband Switching? n Configure Trunk-to-Trunk Transfers Use the change system-parameters features command to enable trunk-to-trunk transfers. This feature is needed to be able to transfer an incoming/outgoing call from/to the remote switch back out to the same or another switch For simplicity, the Trunk-to-Trunk Transfer field was set to all to enable all trunk-to-trunk transfers on a system wide basis. NOTE: This feature can pose a significant security risk by increasing the risk of toll fraud, and must be used with caution. To minimize the risk, a COS can be defined to allow trunk-to-trunk transfers for a specific trunk group(s). For more information regarding how to configure a Communication Manager to minimize toll fraud, see Reference [8 in Section 11. change system-parameters features Page 1 of 18 FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? n Trunk-to-Trunk Transfer: all Automatic Callback with Called Party Queuing? n Automatic Callback - No Answer Timeout Interval (rings): 3 10 of 88

11 4.2 Add Node Name of Avaya Aura Session Manager Using the change node-names ip command, add the node-name and IP address for the Session Manager SIP Entity interface, if not previously added during the initial install of the solution. This same screen shows the node-name for the procr interface which will be used in administering a SIP signaling-group in Section 4.4. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address ASM1-SM ASM2-SM VAL01a clan-1a default gateway procr xfire-1a Configure IP Network Regions In the sample configuration shown in Figure 1, calls to/from Session Manager will be viewed by Communication Manager as calls to/from ip-network-region 2. Communication Manager and its endpoints are in ip-network-region 1. To enable communication between the two network regions requires additional administration of the ip-network-region and signaling-group forms as shown in the next few sections Configure IP Network Region 1 Using the change ip-network-region 1 command, set the Authoritative Domain to the correct SIP domain for the configuration. Verify the Intra-region IP-IP Direct Audio, and Interregion IP-IP Direct Audio fields are set to yes. change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: avaya.com Name: 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: of 88

12 Navigate to page 4 and connect ip-network-region 1 to ip-network-region 2 by typing a y under the direct WAN column for dst rgn 2. Select an ip-codec-set to be used for negotiating audio between the two regions as well. In this case ip-codec-set 2 was used. See Section for additional information on the ip-codec-set form. change ip-network-region 1 Page 4 of 20 Source Region: 1 Inter Network Region Connection Management I M G A t dst codec direct WAN-BW-limits Video Intervening Dyn A G c rgn set WAN Units Total Norm Prio Shr Regions CAC R L e 1 1 all 2 2 y NoLimit n t Configure IP Network Region 2 Using the change ip-network-region 2 command, set the Authoritative Domain to the correct SIP domain for the configuration. Verify the Intra-region IP-IP Direct Audio, and Interregion IP-IP Direct Audio fields are set to yes. change ip-network-region 2 Page 1 of 19 IP NETWORK REGION Region: 2 Location: 1 Authoritative Domain: avaya.com Name: 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: Navigate to page 4 and verify that ip-network-region 1 and 2 are directly connected and use ipcodec-set 2 as shown below. change ip-network-region 2 Page 4 of 20 Source Region: 2 Inter Network Region Connection Management I M G A t dst codec direct WAN-BW-limits Video Intervening Dyn A G c rgn set WAN Units Total Norm Prio Shr Regions CAC R L e 1 2 y NoLimit n t 2 2 all Administer IP Codec Set In Section ip-codec-set 2 was chosen as the inter-region codec to be used for calls between ip-network-region s 1 and 2. In the sample configuration G.711MuLaw was the preferred codec to be used for RTP audio between Communication Manager and the M1000, therefore ip-codecset 2 will be configured to prefer this codec as well. 12 of 88

13 Use the command change ip-codec-set 2 to administer this codec-set. Audio Codec G.711MU is entered as the first choice. Optionally enter in a secondary Codec like G.729A to help ensure there will be two way audio in most cases. Leave all other fields at their defaults. change ip-codec-set 2 Page 1 of 2 Codec Set: 2 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : G.729A n : 4.4 Add SIP Signaling Group Issue the add signaling-group n command, where n is an available signaling group number, for one of the SIP trunks to the Session Manager, and fill in the indicated fields. In the sample configuration, trunk group 10 and signaling group 10 were used to connect to Avaya Aura Session Manager. Default values can be used for the remaining fields. Group Type: sip Transport Method: tcp (1) IMS Enabled?: n Peer Detection Enabled? y Near-end Node Name: procr node-name from Section 4.2 Far-end Node Name: Session Manager SIP Entity interface node name from Section 4.2 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 2 Far-end Domain: Authoritative Domain from Section Session Estab. Timer: 3 (2) ( 1 ) TCP was used for the sample configuration. However, TLS would typically be used in production environments. ( 2 ) If any call originating from the SIP phone is not expected to be answered within 3 minutes which would happen if the call is made to a VDN and agents are not available within 3 minutes, this value may need to be increased. 13 of 88

14 add signaling-group 10 SIGNALING GROUP Group Number: 10 Group Type: sip IMS Enabled? n Transport Method: tcp Q-SIP? n IP Video? n Peer Detection Enabled? y Peer Server: SM SIP Enabled LSP? n Enforce SIPS URI for SRTP? n Near-end Node Name: procr Far-end Node Name: ASM1-SM100 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 2 Far-end Domain: avaya.com 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? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): Add SIP Trunk Group Add the corresponding trunk group controlled by this signaling group via the add trunk-group n command, where n is an available trunk group number, and fill in the indicated fields. Group Type: sip Group Name: A descriptive name. TAC: An available trunk access code Direction two-way for both incoming and outgoing calls Service Type: tie Signaling Group: The number of the signaling group added in Section 4.4 Number of Members: The number of SIP trunks to be allocated to calls routed to Session Manager (must be within the limits of the total number of trunks configured in Section 4.1.1) add trunk-group 10 Page 1 of 21 TRUNK GROUP Group Number: 10 Group Type: sip CDR Reports: y Group Name: to ASM1 COR: 1 TN: 1 TAC: #10 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Signaling Group: 10 Number of Members: of 88

15 Once the add command is completed, trunk members will be automatically generated based on the value in the Number of Members field. On Page 2, set the Preferred Minimum Session Refresh Interval to Note: to avoid extra SIP messages, all SIP trunks connected to Session Manager should be configured with a minimum value of add trunk-group 10 Page 2 of 21 Group Type: sip TRUNK PARAMETERS Unicode Name: auto Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 1200 On Page 3, set Numbering Format to be private. Use default values for all other fields. add trunk-group 10 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 4.6 Administer Private Numbering SIP users registered to Session Manager need to be added to either the private or public numbering table on the Communication Manager Evolution Server. For the sample configuration, private numbering was used and all extension numbers were unique within the private network. However, in many customer networks, it may not be possible to define unique extension numbers for all users within the private network. For these types of networks, additional administration may be required as described in References [5] and [6]. In Section 4.5 trunk-group 10 was configured as private. Use the command change privatenumbering 7 to define the caller ID number which will be sent out with calls over this trunk. For the sample configuration, extension numbers in the range of 666xxxx are used on the Evolution Server. Ext Len: Ext Code: Trunk Grp: Enter the extension length allowed by the dial plan Enter leading digit (s) from extension number Enter the SIP Trunk Group number for the SIP trunk 15 of 88

16 Private Prefix: between the Evolution Server and Session Manager Leave blank unless an enterprise canonical numbering scheme is defined in Session Manager. If so, enter the appropriate prefix change private-numbering 7 Page 1 of 2 NUMBERING - PRIVATE FORMAT Ext Ext Trk Private Total Len Code Grp(s) Prefix Len Total Administered: 1 Maximum Entries: Administer Call Routing There are several administration screens one must edit in order to enable 7-digit dialing from Communication Manager to Session Manager (and ultimately to the CS1000E) without the need to dial a Feature Access Code (FAC) like 9 or *9. These steps are shown in the next few sections Administer Dialplan Analysis In the screenshot below, the following entries were added using the change dialplan analysis command: Dialed String 666 was added for extensions local to Communication Manager 777 was added for extensions on the CS1000E Total Length 7. Call Type Ext change dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Location: all Percent Full: 2 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 0 1 attd 1 2 dac 2 2 fac ext ext 8 1 fac 9 1 fac * 3 dac # 3 dac 16 of 88

17 4.7.2 Administer Uniform Dialplan Using the command change uniform-dialplan 6 make the following changes: Matching Pattern 777 & 666 Len 7 Length of digit string Del 0 Number of digits to delete Net aar change uniform-dialplan 6 Page 1 of 2 UNIFORM DIAL PLAN TABLE Percent Full: 0 Matching Insert Node Pattern Len Del Digits Net Conv Num aar n aar n Administer AAR table In the sample configuration the AAR table is used for three purposes: To route calls to 777xxx (calls to the CS1000E) from Communication Manager to Session Manager, to route calls to 9600 Series SIP telephones which are registered to Session Manager and are in the 666xxxx extension range and to route calls to Modular Messaging. Using the command change aar analysis 6, make the following changes which instruct Communication Manager to use the appropriate route-patern for a given digit string. Dialed String 777 & 666 Total Min 7 Minimum number of dialed digits Total Max 7 Maximum number of dialed digits Route Pattern 10 (route-pattern admin shown in next section) Call type unku Unknown numbering plan change aar analysis 6 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 unku n unku n 17 of 88

18 4.7.4 Administer Route Pattern The final step for enabling 7-digit dialing to Session Manager is to add the trunk group created in Section 4.5 to a route pattern. Use the command change route-pattern 10 to add trunk-group 10 to the route pattern. Pattern Name Use a descriptive name for the route pattern Group No 10 Trunk-group number created in Section 4.5 FRL 0 Restriction Level with 0 being the least restrictive change route-pattern 10 Page 1 of 3 Pattern Number: 10 Pattern Name: ASM1-6.0 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: 10 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 M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest none 4.8 Configure Stations For each SIP user to be defined in Session Manager, add a corresponding station on the Communication Manager Evolution Server. Note: Instead of manually defining each station using the Communication Manager SAT interface, an alternative option is to automatically generate the SIP station when adding a new SIP user in System Manager. See Section 5.9 for more information on adding SIP users. The phone number defined for the station will be the number the SIP user enters to register to Session Manager. Use the add station x command where x is a valid extension number defined in the system. On Page 1 of the add station form: Phone Type: Name: Security Code: Set to 9630SIP Display name for user Numeric password used when user logs into station. 18 of 88

19 add station Page 1 of 6 STATION Extension: Lock Messages? n BCC: 0 Type: 9630SIP Security Code: TN: 1 Port: S00006 Coverage Path 1: 1 COR: 1 Name: John Smith Coverage Path 2: COS: 1 Hunt-to Station: STATION OPTIONS Time of Day Lock Table: Loss Group: 19 Message Lamp Ext: Display Language: english Button Modules: 0 Survivable COR: internal Survivable Trunk Dest? y IP SoftPhone? n IP Video? n On Page 6, set the following: -SIP Trunk : AAR which corresponds to the 666 entry from Section change station Page 6 of 6 STATION SIP FEATURE OPTIONS Type of 3PCC Enabled: None SIP Trunk: AAR 4.9 Verify Off-Pbx Telephone Station-Mapping Use the display off-pbx-telephone station-mapping command for each extension associated with SIP users defined in Session Manager to verify settings: display off-pbx-telephone station-mapping Page 1 of 3 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Application Dial CC Phone Number Trunk Config Dual Extension Prefix Selection Set Mode OPS AAR On Page 2, verify the following values: 19 of 88

20 Mapping Mode: both Calls Allowed: all display off-pbx-telephone station-mapping Page 2 of 3 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Appl Call Mapping Calls Bridged Location Extension Name Limit Mode Allowed Calls OPS 3 both all none Configure Voice Messaging Hunt Group Use the add hunt group n command to add a hunt group to be used by the voice messaging coverage path to be defined in the next section. n is an unused hunt group number. Enter the following values for the specified fields, and retain the default values for the remaining fields. Group Number: Group Name: Group Extension: Group Type: ISDN/SIP Caller Display: An unassigned hunt group number. A meaningful name. An unassigned extension number. ucd-mia mbr-name add hunt-group 1 Page 1 of 60 HUNT GROUP Group Number: 1 ACD? n Group Name: Coverage to MM5.2 Queue? n Group Extension: Vector? n Group Type: ucd-mia Coverage Path: TN: 1 Night Service Destination: COR: 1 MM Early Answer? n Security Code: Local Agent Preference? n ISDN/SIP Caller Display: mbr-name 20 of 88

21 On page 2, assign the following field values: Message Center: sip-adjunct Voice Mail Number: The Group Extension from Page 1. Voice Mail Handle: The Group Extension from Page 1. Routing Digits: The AAR feature access code add hunt-group 1 Page 2 of 60 HUNT GROUP Message Center: sip-adjunct Voice Mail Number Voice Mail Handle Routing Digits (e.g., AAR/ARS Access Code) Configure Voice Messaging Coverage Path Use the add coverage path n command to specify a coverage path to be used for telephone users. This will specify use of the voice messaging hunt group. n is an unused coverage path number. Enter the hunt group number defined in the previous section in Point 1. Default values can be used for the remaining fields. It may be desirable to adjust the Number of Rings before a no-answer call goes to coverage. add coverage path 1 Page 1 of 1 COVERAGE PATH Coverage Path Number: 32 Cvg Enabled for VDN Route-To Party? n Next Path Number: Hunt after Coverage? n Linkage COVERAGE CRITERIA Station/Group Status Inside Call Outside Call Active? n n Busy? y y Don't Answer? y y Number of Rings: 2 All? n n DND/SAC/Goto Cover? y y Holiday Coverage? n n COVERAGE POINTS Terminate to Coverage Pts. with Bridged Appearances? n Point1: h1 Rng: Point2: Point3: Point4: Point5: Point6: 21 of 88

22 4.12 Configure Coverage Path for Telephone Users The following sample station form illustrates how to configure voice mail coverage for a given station user. Set Coverage Path 1 to the value of the coverage path defined in the previous section. change station Page 1 of 5 STATION Extension: Lock Messages? n BCC: 0 Type: 9630SIP Security Code: TN: 1 Port: S00001 Coverage Path 1: 1 COR: 1 Name: AvayaSIP Coverage Path 2: COS: 1 Hunt-to Station: 4.13 Save Translations Configuration of Communication Manager is complete. Use the save translation command to save these changes. Note: After a change on Communication which alters the dial plan, synchronization between Communication Manager and Session Manager need to be completed and any 9600 Series SIP phones must be re-registered to Session Manager. To request an on demand synchronization, log into the System Manager console, navigate to Elements Inventory Synchronization Communication System and initiate an incremental syncronization of Communication Manager as shown below: 22 of 88

23 5 Configure Avaya Aura Session Manager This section provides the procedures for configuring Session Manager and includes the following items: Administer SIP domain Define Logical/physical Locations that can be occupied by SIP Entities Add Session Manager to System Manager For each SIP Entity in the sample configuration: o Define SIP Entity o Define Entity Links, which define the SIP trunk parameters used by Avaya Aura Session Manager when routing calls to/from SIP Entities o Define Routing Policies, which control call routing between the SIP Entities o Define Dial Patterns, which govern to which SIP Entity a call is routed Administer CM-ES as a Sequenced Application. Define the Communication Manager Evolution Server as an administrable entity Adding SIP Endpoints/SIP users in System Manager Configuration is accomplished by accessing the browser-based GUI of Avaya Aura System Manager, using the URL where <ip-address> is the IP address of Avaya Aura System Manager. Expand the Routing link on the left side of Navigation Menu. Select a specific item such as Entity Links. When the specific item is selected, the color of the item will change to blue as shown below: 23 of 88

24 5.1 Administer SIP Domains Expand Routing as described above and select Domains. Click New In the General Section, under Name add a domain name. Under Notes add a brief description Click Commit to save. The screen below shows the domain avaya.com which was used for the sample configuration. 5.2 Define Locations Expand Routing and select Locations. Locations are used to identify logical and/or physical locations where SIP Entities reside, for purposes of bandwidth management or location-based routing. Click New to create a new location. Name Add a descriptive name for the location Notes Add a brief description IP Address Pattern Enter pattern used to logically identify the location 24 of 88

25 The screen below shows the information for the CS1000E, Mediant 1000 and other locations in the sample configuration. The following screen shows the location information as entered for the Evolution Server. 5.3 Add SIP Entities Define Avaya Aura Session Manager as a SIP Entity One of the first steps in properly setting up Session Manager and System Manager is to add Session Manager as SIP Entity. Generally this is done during the initial installation of Session Manager and System manager. To do this, log in to System Manager and from the left-side navigation pane, expand the Routing link by selecting it, and then select SIP Entities. Fill in the fields as described and shown below. Click Commit to complete: 25 of 88

26 Name A descriptive name FQDN or IP Addr Hostname or IP address of the SM-100 interface in Session Mgr. Type Session Manager Notes Free-form text Location Appropriate location created in Section 5.2 Oubound Proxy Leave blank Time Zone Time zone value appropriate for the physical location Sip Link Mon Usually set to Use Session Mgr Config though it can be customized on a per-element basis Define Ports for Use by Avaya Aura Session Manager Session Manager has the ability to translate communication between two SIP entities that talk using different ports and protocols. However to do so, it is necessary to define the ports and protocols that Session Manager will need to communicate with. The screen shot shown below is the lower half of the same screen used to add/edit Session Manager as a SIP Entity and discussed in the previous section. As shown below, two ports (5060 & 5070) and 2 protocols (UDP & TCP) are defined for one instance of Session Manager. TLS can also be configured (5070 was not used in the sample configuration): 26 of 88

27 5.3.3 Add Avaya Aura Session Manager to System Manager To complete the linkage between System Manager and Session Manager it s necessary to identify the SIP Entity created in the previous section as an instance of Session Manager to System Manager. Generally this is done during the initial installation of System Manager and Session Manager. As shown below, expand the Elements menu on the left pane then select Session Manager then Session Manager Administration. Then click New (not shown), and fill in the fields as described below and shown in the following screen: Under General: SIP Entity Name: Description: Management Access Point Host Name/IP Under Security Module: Network Mask: Default Gateway: Select the SIP Entity added for Session Manager Descriptive comment (optional) Enter the IP address of the Session Manager management interface (not the SIP Entity interface address). Enter the network mask corresponding to the IP address of the SIP Entity Interface Enter the IP address of the default gateway for the SIP Entity Interface Use default values for the remaining fields. Click Commit to add this Session Manager. The screen below shows the resulting Session Manager definition. 27 of 88

28 5.3.4 Define a SIP Entity for Avaya Aura Communication Manager Evolution Server The following screen shows the addition of Communication Manager Evolution Server as a SIP Entity. The IP address shown below is that of the near-end node ( procr ) used on the signaling-group form from Section of 88

29 5.3.5 Define a SIP Entity for Avaya Modular Messaging The following screen shows the addition of the Modular Messaging Server as a SIP Entity. The IP address shown below is that of the MAS virtual machine. 29 of 88

30 5.3.6 Define a SIP Entity for the Mediant 1000 The following screen shows the addition of the Mediant 1000 as a SIP Entity. The IP address is Type is other. 5.4 Create Entity Links A SIP trunk between Session Manager and a telephony system is described by an Entity Link. To add an Entity Link, expand Routing from the left-pane then select Entity Links. Click on the New button on the right-pane to create a new entry. Alternately, an Entity Link can be created on the SIP Entity form as well though the SIP Entity must be added before an Entity Link can be created. 30 of 88

31 Fill in the following fields in the new row that is displayed: Name: SIP Entity 1 Protocol Port SIP Entity 2 Port Trusted A descriptive name. Select the Session Manager. Select TCP, TLS or UDP from the dropdown Port number to which the other system sends SIP requests Select the name of the other system. Port number on which the other system receives SIP requests Check this box. Note: If this box is not checked, calls from the associated SIP Entity specified in Section 5.3 will be denied. Click Commit to save each Entity Link definition. The following screens illustrate the Entity Links for Communication Manager Evolution Server, Mediant 1000 and Modular Messaging, all of which use port 5060 and TCP to communicate with Session Manager. Communication Manager: 31 of 88

32 CS1000E: Mediant 1000: 5.5 Add Routing Policies Routing policies describe the conditions under which calls will be routed to the SIP Entities specified in Section 5.3. Routing policies for Communication Manager and the CS1000E need to be added. For 96x00 Series SIP telephones registered to Session Manager, the necessary SIP communication between Session Manager and Evolution Server happens as a result of administering a Sequenced Application shown in Section 5.7. To add a routing policy, select Routing Policies on the left and click on the New button (not shown) on the right. The following screen is displayed. Fill in the following: General SIP Entity as Destination Time of Day Enter a descriptive name in Name. Click Select, and then select the appropriate SIP Entity to which this routing policy applies. Click Add, and select the default 24/7 time range. Defaults can be used for the remaining fields. Click Commit to save each Routing Policy definition. 32 of 88

33 The following screen shows the Routing Policy to send calls to the Communication Server A similar entry was created for Communication Manager (not shown): 33 of 88

34 5.6 Add Dial Patterns Define dial patterns to direct calls to the appropriate SIP Entity. Calls to 7-digit extensions beginning with 777xxxx should be routed to the Communication Server Calls to to 7- digit extensions beginning with 666xxxx should be routed to Communication Manager. To add a dial pattern, select Dial Patterns on the left and click on the New button (not shown) on the right. Fill in the following, as shown in the screens below: Under General: Pattern: Dialed number or prefix Min: Minimum length of dialed number. Max: Maximum length of dialed number. SIP Domain: SIP domain specified in Section 4.1 Notes: Comment on purpose of dial pattern. Under Originating Locations and Routing Policies: Click Add, and then select the appropriate location (or ALL ) and routing policy from the list. Default values can be used for the remaining fields. Click Commit to save the dial pattern. The following screenshot shows the dial pattern for routing calls to the Communication Server The following screenshot shows the dial pattern for routing calls to the Communcation Manager Evolution Server. 34 of 88

35 35 of 88

36 5.7 Define Communication Manager Evolution Server as an Administrable Entity Before adding SIP users, the Avaya Aura TM Communication Manager Evolution Server must be added to System Manager as an administrable entity. This action allows System Manager to access Communication Manager over its administration interface similar to how other administration tools such as Avaya Site Administration access Communication Manager. Using this administration interface, System Manager will notify the Communication Manager Evolution Server when new SIP users are added Add Communication Manager as a Manageable Element To define the Communication Manager Evolution Server as an administrable entity go to Elements Inventory Manage Elements and select New (not shown). In the section titled Application enter in the following information: Type Select CM from the drop-down Name Enter an identifier for the Communication Manager Evolution Server. Node Enter the IP address of the administration interface for the Evolution Server 36 of 88

37 Scroll down to the section titled Attributes and enter the following login information for Communication Manager: Login Enter a login ID that System Manager will use to login to a SAT session on Communication Manager. NOTE: This login ID should be dedicated for System Manager s use only. Password/Confirm Password for the login used in the above field Is SSH Connection Check this box if SSH access has been enabled for SAT access to Communication Manager. SSH is enabled by default on Communication Manager. Port 5022 if SSH is enabled (default) if Telnet is enabled Synchronize Communication Manager with System Manager Select Elements Inventory Manage Elements Synchronization Communication System on the left. Check the appropriate Element Name, click Initialize data for selected devices and click Now. This may take some time to complete while System Manager examines the entire configuration on Communication Manager. 37 of 88

38 38 of 88

39 6 Configure Avaya Modular Messaging In the sample configuration, Communication Manager and CS1000E telephone systems were added as sites to Avaya Modular Messaging Multi-Site. The initial installation and configuration steps for Modular Messaging are not covered in this document. Please see Section 11 References 9 & 10 for more detail on this topic. The steps covered in this section are the following. Administering Modular Messaging Verify Multi-Site configuration Administer PBXs Administer Sites Administer Subscribers 6.1 Administering Modular Messaging Unless otherwise indicated, most of the administration steps described below are done on the Message Application Server (MAS) via the VMSC console. The MAS is a Windows 2003 server, so outside of a direct console connection, the easiest way to access this server is via Microsoft Remote Desktop Protocol (RDP). Once the RDP or console session is established, launch the VMSC application. There should be an icon on the desktop for this application: 39 of 88

40 Subscriber Management in Modular Messaging MultiSite is accomplished via a web page to the Messaging Storage Server (MSS). The MSS runs the Linux OS. 6.2 Verify MultiSite Configuration The first step in configuring MM is to verify Multisite is enabled for the MM installation. Launch the VMSC application, double-click Sites and verify Enable MultiSite is checked: 40 of 88

41 6.3 Verify Port Settings In Section 5 Modular Messaging was configured as a SIP Entity communicating with Session Manager using TCP and port By default MM ships with only TLS/5061 enabled. Therefore to use TCP and port 5060 it is necessary to enable this on the MAS. To do so, launch the VMSC and double-click the PBX Integration icon. Be sure that TCP Port Number is set to 5060 and the Enable check-box is selected. Select OK when done. 41 of 88

42 6.4 Administer a PBX From the Modular Messaging perspective all SIP messages will be coming from Session Manager. The Communication Manager and CS1000E systems are defined to Modular Messaging as sites whereas Session Manager will be defined as a PBX. In the MAS, open the VMSC app, right-click on the PBXs icon and select Add New PBX, as shown below. On the General tab of the resulting displayed window, enter an appropriate PBX Name. Defaults can be used for the remaining fields. 42 of 88

43 On the Transfer/Outcall tab, select Full for Transfer Mode. Default values can be used for the Tone Detection tab. On the SIP tab navigate to the Gateways section, click on the icon and add the Session Manager s IP address under Address/FQDN, TCP for Protocol, and click the MWI box so message waiting notifications will be sent. Fill in SIP Domain with the domain from Section 5.1. Click on Configure to specify number translation rules for translating between the local dial plans of Communication Manager and CS1000E telephone systems and the normalized 11 digit form used by Modular Messaging. 43 of 88

44 After selecting Configure the Translation Rules screen appears. For more detailed information on how to administer this screen it is important to read Section 11 Reference 9 which has an indepth explanation of translation rules. As shown below, the right pane data contains the appropriate rules to translate between extensions as they exist on the PBX in 7-digit form (666xxxx and 777xxxx), and the canonical form of the 11-digit mailboxes as administered on the MSS. These rules are created using regular expression syntax. For the sample configuration, the first 2 rules are required and were added by selecting Add (not shown). Proper operation of the rules can be verified by adding Test inputs in the left pane and viewing the resulting output in the corresponding rule in the right pane. In the screen-shot below, test input of on an incoming call matches the first rule and is converted to canonical number Click on OK when finished, then again on OK in the original Add new PBX window (see previous screen). 44 of 88

45 6.5 Administer Sites The Communication Manager and CS1000E telephone systems must be added as sites in Modular Messaging. Adding a site is done by double-clicking Sites in the Voice Mail System Configuration tool, as shown below. In the Sites window that is displayed, click on Configure. 45 of 88

46 The Site Configuration window is displayed. By default there is a group called <ROOT> and all sites can be added under this group. Optionally, you can create multiple groups with each group having multiple sites underneath. For the sample configuration all sites were created under <ROOT> though the steps to create an additional group are also discussed. 1) Create a new site Group (optional). To create a site group which will be referenced when adding new sites click on the Add button and select Group and enter the following in the New Site Group window: Parent site group Parent site name (default is Root) Site group name Site group name (e.g. Ireland) Identifier A unique number identifying the site group Group Container Only Check this box if there is no way to uniquely identify this group via a digit string. Click on Add when finished. 46 of 88

47 1) Create new Sites. In the sample configuration two sites were created: one for Communication Manager and the 666xxxx extensions and one for the CS1000 and the 777xxxx extensions. However to simulate a more realistic environment, these extensions were considered to be in different physical locations and therefore would use different area codes was used for Communication Manager extensions and 1905 was used for the CS1000 extensions. Click on Add and select Site from the dropdown to add to the ROOT site (or select another appropriate site if one was created), and enter the following in the New Site window: Parent site group Parent site name (e.g., ROOT) Site name Site name ( CS1000E-R6 ) Identifier The unique initial digits of the 11-digit mailbox number, identifying the site. In this case the first 7 digits were chosen ( ) Full mailbox length Enter 11 for the full mailbox number length Short mailbox length Enter 7 for the extension length PBX Enter name of the PBX added in the previous Section 6.4. Click on Add when finished and repeat for the Communication Manager site. The following screen shows adding of the CS1000e as a site. When all sites are added, click OK in the Site Configuration window, and then click on OK in the original Sites window. 47 of 88

48 After adding the 2 nd site for Communication Manager, the resulting Site Configuration screen appeared as follows. The two entries for the sample configuration are highlighted: 6.6 Administer Subscribers Subscriber management is performed on the MSS. To log in to the MSS point a web browser like Internet Explorer at the IP address or FQDN of the MSS. Log in with the appropriate credentials. Select Messaging Administration Subscriber Management from the left pane, to display the Manage Subscribers screen. For the Local Subscriber Mailbox Number field toward the top of the screen, enter a mailbox number of the subscriber. Click Add or Edit box to define more information for the mailbox subscriber. 48 of 88

49 To see a list of all local subscribers simply select the MANAGE button at the right-end of the Local Subscribers row as shown on the previous screen. From either screen it is possible to add a new subscriber. Once the Add Local Subscriber screen is displayed enter in the following information: Last Name: Last name of the person s whose extension this is. First Name: First name of the person s whose extension this is. Password: Numeric password the user will need to use to access their mailbox for the first time. The system will generally make them change it. Mailbox Number: The complete mailbox number. As discussed in previous sections, all mailboxes for the sample configuration are 11-digits in length. PBX Extension: Select either Canonical or Switch Native and Enter in one of the following: - Canonical: Full mailbox number preceded by + ( ). - Switch Native: 7-digit extension as it is configured on the PBX ( ) Class of Service: From the drop-down, the appropriate COS. All other fields can be left at their defaults. Scroll down to the bottom of the screen and click Save (not shown). Repeat this section to add all subscribers. 49 of 88

50 7 Configure Communication Server 1000 This section focuses on configuring the T1 QSIG trunks on Communication Server 1000E to reach Avaya Communication Manager. In addition, this section highlights specific feature parameters that need to be configured on the CS1000E station users in order to use the Avaya Modular Messaging for voice messaging service. These Application Notes assume that ISDN PRI is not being configured for the first time, so error detection thresholds and clock synchronization control are assumed to be in place. If not, refer to the ISDN Primary Rate Interface document in Section 11 for detailed descriptions. Furthermore, these Application Notes used the Coordinated Dial Plan (CDP) feature to route calls from the Communication Server 1000, over the T1 QSIG trunks to Avaya Communication Manager. The CDP feature is assumed to be already enabled on the Communication Server 1000, and therefore will not be described in detail. The procedures below describe the details of configuring a Communication Server 1000: Launch Element Manager Verify equipped feature packages Administer TMDI card Administer D-Channel Administer routes and trunks Administer route list block Administer distant steering code Enable TMDI card Enable D-channel automatic establishment Administer station users 50 of 88

51 7.1 Launch Element Manager Access the Communication Server 1000 web based interface Element Manager by using the URL in an Internet browser window, where <ip-address> is the IP address of the Signaling Server. Note that the IP address for the Signaling Server may vary, and in this case is used. The CS 1000 ELEMENT MANAGER screen is displayed. Enter the appropriate credentials, retain the automatically populated value in the Call Server IP Address field, and click Login. 7.2 Verify Equipped Feature Packages After login to UCM, a list of manageable elements is shown. Select the CS1000 element type as shown below to complete the next few sections. 51 of 88

52 The CS1000 Element Manager: System Overview screen is displayed. From the left pane, select Tools Logs and Reports Equipped Feature Packages The Equipped Feature Packages List screen is displayed next, and shows a listing of the licensed feature packages in sequential order by package number. Scroll down the right pane as necessary to verify that the feature packages listed following the screen below are equipped (the corresponding package numbers are the Package Number values on right side of the list): 52 of 88

53 Digit Display (DDSP) Coordinated Dialing Plan (CDP) Calling Party Name Display (CPND) Integrated Services Digital Network (ISDN) Primary Rate Access (CO) (PRA) 2.0 Mb/s Primary Rate Interface (PRI2) Overlap Signaling (M1 to M1 and M1 to 1TR6 CO) (OVLP) International Primary Rate Access (CO) (IPRA) QSIG Interface (QSIG) QSIG Generic Functional protocol (QSIGGF) QSIG Supplementary service (QSIG-SS) 7.3 Administer TMDI Card Select System Core Equipment Loops from the left pane to display the Loops (Common Equipment) screen. In the Digital Trunk Interface Loop Number (DLOP) field, click Add New DLOP to add a digital trunk interface to the TMDI card. 53 of 88

54 The Add a Digital Trunk Interface screen is displayed next. The following values were used in the sample configuration: Digital Trunk Interface Loop Number Choose a loop value from Media Gateway Card Enter in the appropriate Superloop, Shelf and Card number that corresponds to card s physical location Number of voice or data calls (DATA_CALLS_LIMIT) Select 23 from the drop-down list. Frame Format Select Extended Super Frame (ESF) Mode of operation (MODE) Primary Rate Interface mode (PRI) from the drop-down list. TMDI Card Select checkbox Line Coding Method Select B8ZS Line Coding Method (B8S) Threshold (TRSH) Select 0 from the drop-down list. Retain the default values for all remaining fields, and click the Return button at the bottom of the screen. 54 of 88

55 The Loops (Common Equipment) screen is displayed again, and updated with values in the Digital Trunk Interface Loop Number field. Click the Save button. 7.4 Administer D-Channel Select Routes and Trunks > D-Channels from the left pane to display the D-Channels screen. In the Choose a D-Channel Number field, select an available D-channel from the drop-down list (in this case 20 ). Click to Add. For the sample configuration Channel: 20 was used. 55 of 88

56 The D-Channels 20 Property Configuration screen is displayed. Enter the following values for the specified fields, and retain the default values for the remaining fields. D channel Card Type (CTYP): TMDI MediaGateway Card(MG_Card): Enter the physical TMDI card location, in this case was used Port number (PORT): 1 Designator (DES): A descriptive text. User (USR): Primary Rate Interface (PRI) Interface type for D-channel: ISIG interface with GF platform (ISGF) D-Channel PRI loop number: The digital trunk interface loop number from Section 7.3. Release ID of the switch at the far end (RLS Select 4 from the drop-down list Select Basic options (BSCOPT) toward the bottom of the screen to expand it. 56 of 88

57 The screen is updated with additional parameters populated below Basic options (BSCOPT). For the PINX customer number (PINX_CUST) field, select the appropriate customer number. The system can support more than one customer with different network settings and options. For the interoperability testing, only one customer was configured on the system. For the D-channel transmission Rate (DRAT) field, select 64 kb/s clear (64KC) from the drop-down list. Retain the default values in the remaining fields, and click Edit next to the Remote Capabilities (RCAP) field. 57 of 88

58 The Remote Capabilities Configuration screen is displayed next. Scroll down the screen as necessary to check the following capabilities: Call completion on busy using integer value (CCBI) Call completion on no response using integer value (CCNI) Connected line identification presentation (COLP) Call transfer integer (CTI) Diversion info. sent. rerouting requests processed (DV3I) Name display integer ID coding (NDI) Path replacement uses integer values (PRI) Message waiting indication using integer values (QMWI) Click Return Remote Capabilities at the bottom of the screen. The D-Channels 20 Property Configuration screen is displayed again (not shown below). Click Submit. 7.5 Administer Routes and Trunks Select Routes and Trunks Routes and Trunks from the left pane to display the Routes and Trunks screen. Next to the applicable Customer row, click Add route. 58 of 88

59 The Customer 0, New Route Configuration screen is displayed next. Enter the following values for the specified fields, and retain the default values for the remaining fields. Route Number (ROUT): Select an available route number. For the sample configuration, Route 20 was added. Designator field for trunk (DES): A descriptive text. Trunk Type (TKTP): TIE trunk data block (TIE) Incoming and Outgoing trunk (ICOG): Incoming and Outgoing (IAO) Access Code for the trunk route (ACOD): An available access code of 88

60 Scroll down the screen, and check the Digital Trunk Route (DTRK) checkbox, to enable two additional fields to appear. For the Digital Trunk Type (DGTP) field, select ISDN 23B + D (PRI) from the drop-down list. Scroll down the screen, check the Integrated Services Digital Network option (ISDN) checkbox to enable additional fields to appear. For the Mode of operation (MODE) field, select ISDN/PRA route, DTRK must be YES (PRA) from the drop-down list. For the Interface type for route (IFC) field, select ISIG interface with GF platform. (ISGF) from the drop-down list. For the Call Type for outgoing direct dialed TIE route (CTYP) field, select Unknown Call type (UKWN) from the drop-down list. Scroll down to the bottom of the screen, and click Save. The Routes and Trunks screen is displayed again, and updated with the newly added route. Click the Add trunk button next to the newly added route. 60 of 88

61 The Customer 0, Route 20, New Trunk Configuration screen is displayed. Enter the following values for the specified fields, and retain the default values for the remaining fields. Scroll down to the bottom of the screen, and click Save. The Multiple trunk input number (MTINPUT) field may be used to add multiple trunks in a single operation, or repeat the operation for each trunk. Multiple trunk input number (MTINPUT): 23 Terminal Number (TN): The TMDI slot number and port 1. Designator field for trunk (DES): A descriptive text. Route number, Member number (RTMB): Current route number and starting member. Trunk Group Access Restriction (TGAR): Desired trunk group access restriction level. 61 of 88

62 7.6 Administer Route List Block Select Dialing and Numbering Plans Electronic Switched Network from the left pane to display the Electronic Switched Network (ESN) screen. Select Route List Block (RLB). The Route List Blocks screen is displayed. In the Please enter a route list index field, enter an available route list block number (in this case 20 ). Click to Add. The Route List Block screen is updated with a listing of parameters. For the Route Number (ROUT) field, select the route number from Section 7.5. Retain the default values for the remaining fields, and scroll down to the bottom of the screen and click Save (not shown). 62 of 88

63 63 of 88

64 7.7 Administer Distant Steering Code The Electronic Switched Network (ESN) screen is displayed again. Select Distant Steering Code (DSC) to add an entry to route 666xxxx calls to the PRI via the Route List created in the previous section. The Distant Steering Code List screen is displayed next. From the drop-down that reads Display select Add. In the Please enter a distant steering code field, enter the dialed prefix digits to match on (in this case 666 ). Click to Add. The Distant Steering Code screen is displayed. For Flexible Length number of digits (FLEN) enter in the maximum length of dialed digits (in this case 7 was used). For the Route List to be accessed for trunk steering code (RLI) field, select the route list index in Section 64 of 88

65 7.6 ( 20 ) from the drop-down list. Retain the default values in all remaining fields, and scroll down to the bottom of the screen to click Submit. 7.8 Enable TMDI Card Even though the TMDI card can be enabled via the web based interface Element Manager, the D-channel cannot come into service unless the TMDI card is enabled via the command line interface. The CS1000E s command line interface (CLI) is most commonly accessed by using a terminal emulation program such as PuTTY to SSH into the Linux shell on the CP+PM and then by typing cslogin to access the CLI. 65 of 88

66 The Communication Server 1000 command line interface is a character-based interface to the operating system and overlay programs on each system component. The program issues a prompt for input, and the system administrator enters the appropriate response through the keyboard followed by the Return key. The output from the Communication Server 1000 command line interface has been trimmed down in the subsequent sections in order to focus on the key settings for the configuration. Values highlighted in bold represent values entered by the system administrator. > login USERID? xxxxx PASS? yyyyy Command TTY #00 LOGGED IN xxxxx 16:52 24/4/2007 > ld 96. enl tmdi all Comment Issue the login command. Enter a valid user ID. Enter a valid user password. A sample response indicating successful log in. Use load 96 to enable the TMDI card. Enable the TMDI card with the physical slot number of the TMDI card and the option all. 7.9 Enable D-channel Automatic Establishment Use the command line interface to enable automatic establishment for the administered D- channel. To bring up the T1 it is typically easier to disable then enable the loop associated with the TMDI card. Command >ld 96. enl auto 20 > ld 60. disl 20. enll 20 Comment Use load 96 to enable D-channel Automatic Establishment If the D-Channel does not come up on its own, it may be necessary to disable and enable the Loop associated with the physical TMDI T1 card. Use load 60 to disable and re-enable the loop Administer Station Users For each CS1000E station user that will be using Avaya Modular Messaging for voic , use the CS1000E command line interface to administer feature parameters below (administration can also be completed using Element Manager or Telephony Manager): 66 of 88

67 Command > ld 11 > REQ: chg > TYPE: 3904 > TN > ECHG yes > ITEM cls fna > ITEM fdn > ITEM cls hta > ITEM hunt > ITEM cls cfxa > ITEM cls mwa > ITEM cls cnda Comment Use load 11 to change station user parameters. Enter chg for change. Enter the specific station type, in this case Enter the applicable TN number for the station. Enter yes for easy change. Activate FNA to allow call forwarding upon no answer. Set the flexible call forwarding no answer destination FDN to the Avaya Modular Messaging pilot number, in this case Activate HTA to allow for coverage upon busy. Set the coverage upon busy destination HUNT to the Avaya Modular Messaging pilot number, in this case Activate CFXA to allow call forward of all calls to external numbers. Activate MWA to allow for message waiting indicator. Activate CNDA to allow for display of calling party name. 67 of 88

68 8 Configure AudioCodes Mediant 1000 The following sections describe the configuration steps required to implement T1/PRI QSIG and SIP trunks and inter-trunk routing on the AudioCodes Mediant 1000, using the web interface. It is assumed that basic hardware and software installation has been performed as described in [10]. This section focuses on the following configuration areas: IP address assignment PSTN trunk settings SIP protocol parameters Routing tables PSTN trunk group Voice mail parameters 8.1 Assign IP address Connect the provided serial cable to the console port, execute a serial port communications program such as HyperTerminal, and set the following communications parameters: Baud Rate: 115,200 bps Data bits: 8 Parity: None Stop bits: 1 Flow control: None Log in to the Command Line Interface (CLI) using the appropriate credentials. Ethernet Port 1 Console Port 68 of 88

69 The following CLI commands set the IP address and default gateway of Ethernet Port 1 to correspond to the sample configuration. User input is shown in bold. The sc ip command has the arguments IP_address, Network_mask, and Default_gateway. After the last command (sar), the Mediant 1000 will reboot with the new settings. Connect Ethernet Port 1 to the network. SIP/ SECurity/ MGmt/ PStn/ DebugRecording/ ControlProtocol/ CONFiguration/ IPNetworking/ TPApp/ BSP/ PING SHow />conf AutoUPDate SaveAndReset RestoreFactorySettings SetConfigParam GetParameterDescription GetConfigParam CHangeUserName CHangePassWord ConfigFile /CONFiguration>sc ip scp ip Network parameters successfully changed: IP address , netmask , gateway AutoUPDate SaveAndReset RestoreFactorySettings SetConfigParam GetParameterDescri ption GetConfigParam CHangeUserName CHangePassWord ConfigFile /CONFiguration>sar Resetting the board... MAC address = F-16-BC-0A Local IP address = Subnet mask = Default gateway IP address = TFTP server IP address = Boot file name = M1000_SIP_F6.00A cmp INI file name = Call agent IP address = Log server IP address = BootLoad Version 2.06 Boot from Flash - Program area 0 SIP/ SECurity/ MGmt/ PStn/ DebugRecording/ ControlProtocol/ CONFiguration/ IPNetworking/ TPApp/ BSP/ PING SHow /> 69 of 88

70 8.2 Access Web Configuration Interface Subsequent configuration can be accomplished by accessing the web interface with an Internet browser, using the URL with the assigned IP Address. The following home/status page will be displayed. Click on Configuration and set the mode to Full. The menus on the left can be expanded as necessary to configure the appropriate features, as described in the following sections. 70 of 88

71 8.3 Configure PSTN Trunk Settings Expand the PSTN Settings menu and click on Trunk Settings. The following web page is displayed. Click on Stop Trunk, which will enable editing of the parameters. Set the following parameters, leaving the remaining parameters at their default values. Under General Settings: Protocol Type: T1 QSIG Under Trunk Configuration: Clock Master Line Code Framing Method Recovered B8ZS T1 FRAMING ESF CRC6 71 of 88

72 Under ISDN Configuration: ISDN Termination Side Q931 Layer Response Behavior Outgoing Calls Behavior Incoming Calls Behavior Network side 0x x400 0x60 Under Miscellaneous: Play Ringback Tone to Trunk Don t Play Click on Apply Trunk Settings to save all of the above changes and put the trunk into service. Successful trunk configuration will be indicated by the green status indications for the trunk board, as shown in the first figure in Sections 8.2 and Configure SIP Protocol Parameters To configure the SIP parameters used when signaling with Session Manager, Communication Manager, and Modular Messaging, expand the Protocol Configuration menu followed by the Protocol Definition menu General Parameters Click on SIP General Parameters. Set the following parameters, leaving the remaining parameters at their default values. 72 of 88

73 Under SIP General: PRACK Mode Asserted Identity Mode Fax Signaling Method Supported Adding PAsserted Identity T.38 Relay SIP Transport Type TCP SIP Destination Port 5060 Enable Remote Party ID Enable History-Info Header Disable Enable 73 of 88

74 Click on Submit to save these changes Proxy Parameters Click on Proxy & Registration on the left. Set the following parameters, leaving the remaining parameters at their default values. Use Default Proxy Proxy Name Always Use Proxy Yes The SIP domain entered in Section 5.1 ( avaya.com ) Enable Gateway Name: The SIP domain entered in Section 5.1 ( avaya.com ) Click on Submit to save these changes. 74 of 88

75 In the left pane expand Protocol Configuration expand Proxies, Registration, IP Groups and select Proxy Sets Table. In Row 1 of the table that is displayed, enter the IP address of the SIP Entity interface of Session Manager in the Proxy Address column, and TCP in the Transport column. Click on Submit to save these changes Audio Coders Click on Coders And Profile Definitions Coders on the left. In the rows of the table that is displayed, enter the desired codecs in order of preference. In the sample configuration, G.711 mu-law and G.729 audio codecs were tested. The remaining parameters are set automatically, although if G.729B is desired, then Silence Suppression must be set to Enabled. Click on Submit to save these changes. 75 of 88

76 8.4.4 DTMF Signaling Click on Protocol Definition DTMF & Dialing on the left. Set the following parameters, leaving the remaining parameters at their default values. Click on Submit to save these changes. Declare RFC 2833 in SDP Yes 1 st Tx DTMF Option RFC 2833 RFC 2833 Payload Type 101 Default Destination Number serveduser 8.5 Configure Routing Tables To configure the tables used for routing calls between the PSTN and SIP interfaces, expand the Routing Tables menu under Protocol Configuration on the left. Since use of a SIP proxy was specified in Section 8.4.2, the Tel to IP Routing does not need to be configured - all calls from the PSTN are routed to the specified SIP proxy (Avaya Aura Session Manager). To configure routing from SIP to PSTN, click on IP to Trunk Group Routing on the left. Set the following parameters in Row 1, leaving the remaining parameters at their default values. These values specify that all SIP calls are to be routed to the T1 PRI interface (Trunk Group 1) configured in Sections 7.3 and 7.6. Click on Submit to save these changes. Dest. Phone Prefix * Source Phone Prefix * Source IP Address * Trunk Group ID 1 (see Section 8.6) IP Profile ID 0 76 of 88

77 8.6 Configure PSTN Trunk Group To configure the trunk group associated with the T1 PR1 trunk configured in Section 8.3, expand the Trunk Group menu under Protocol Configuration on the left. Click on Trunk Group and set the following parameters for Group Index 1, leaving the remaining parameters at their default values. Click on Submit to save these changes. Module Select Module 1 PRI (see first screen in Section 8.3) From Trunk Starting physical trunk - Select 1 (see first screen in Section 8.3) To Trunk Ending physical trunk - Select 1 (see first screen in Section 8.3) Channels 1-24 Phone Number Enter a logical phone number that will be used if a call from the PSTN does not contain a calling number (optional) Trunk Group ID 1 (should corresponding to the Trunk Group ID in the IP to Trunk Group routing entry in Section 8.5) IP Profile ID 1 (note default values for this profile were used in the sample configuration) 77 of 88

78 8.7 Configure Voice Mail Parameters Configure QSIG MWI The AudioCodes Mediant 1000 can translate message waiting indication (MWI) signaling between the QSIG and SIP protocols. To configure this, expand Advanced Applications on the left. Click on Voice Mail Settings and set the following parameter, leaving the remaining parameters at their default values. Click on Submit to save these changes. Under General: Voice Mail Interface Select QSIG Configure SIP MWI To configure the Mediant 1000 to properly handle SIP NOTIFY messages which are used to convey the status of a subscribers MWI, it is necessary to manually add the necessary parameters to the the Mediant 1000 s configuration file. This file is easily edited in a Windows program such as Notepad. 1) First upload the Mediant 1000 s config file from the web interface. In the GUI select the Management button expand Software Update select Configuration File. The following screen appears: 78 of 88

79 2) Select the Save INI File button to upload the file to a PC. The filename will be BOARD.ini. 3) Edit the file with a program like Notepad and add the following parameters under the section titled [SIP Params]: DEFAULTNUMBER = 'serveduser' ensures correct extension is notified of MWI status MWISERVERIP = ' ' IP address of Session Manager ENABLEMWI = 1 enables SIP NOTIFY messages for MWI SUBSCRIPTIONMODE = 1 allows Mediant 1000 to periodically query for MWI status Note: Do not add the arrows or comments to the BOARD.ini file Add the following parameters in the section titled [PSTN Params]: ISDNIBehavior= Add the following parameters in the section titled [Voice Engine Params]: ECNLPMode = 1 4) Save the file and load it back on to the Mediant From the same screen as shown above in Step 1, use the Send INI File button to load the modified Board.ini file back on to the Mediant Note: This will cause the Mediant 1000 to reset and therefore may require a reset of the TMDI board on the CS1000E. See Section 7.9 on how to bring the T1 back into service if it does not come back on its own. 79 of 88

80 9 Verification Steps 9.1 Verify Communication Manager Verify the status of the SIP trunk group by using the status trunk n command, where n is the trunk group number administered in Section 3.6. Verify that all trunks are in the inservice/idle state as shown below. status trunk 10 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy 0010/001 T00226 in-service/idle no 0010/002 T00227 in-service/idle no 0010/003 T00228 in-service/idle no 0010/004 T00229 in-service/idle no 0010/005 T00230 in-service/idle no 0010/006 T00231 in-service/idle no 0010/007 T00232 in-service/idle no 0010/008 T00233 in-service/idle no 0010/009 T00234 in-service/idle no 0010/010 T00235 in-service/idle no Verify the status of the SIP signaling groups by using the status signaling-group n command, where n is the signaling group number administered in Section 3.6. Verify the signaling group is in-service as indicated in the Group State field shown below. status signaling-group 10 STATUS SIGNALING GROUP Group ID: 10 Active NCA-TSC Count: 0 Group Type: sip Active CA-TSC Count: 0 Signaling Type: facility associated signaling Group State: in-service 80 of 88

81 Make a call between the Avaya 9600 Series IP Telephone and an 1140E UNIStim Telephone. Verify the status of connected SIP trunks by using the status trunk x/y, where x is the number of the SIP trunk group from Section 5.2 to reach Session Manager, and y is the member number of a connected trunk. Verify on Page 1 that the Service State is inservice/active. On Page 2, verify that the IP addresses of the Procr and Avaya Session Manager are shown in the Signaling section. In addition, the Audio section shows the correct Codec Type and the IP addresses of the Avaya telephone and the AudioCodes Mediant The Audio Connection Type displays ip-direct, indicating direct media between the two endpoints. status trunk 10/1 Page 1 of 3 TRUNK STATUS Trunk Group/Member: 0010/001 Port: T00226 Signaling Group ID: 10 Service State: in-service/active Maintenance Busy? no IGAR Connection? no Connected Ports: S00504 status trunk 10/1 Page 2 of 3 CALL CONTROL SIGNALING Near-end Signaling Loc: PROCR Signaling IP Address Port Near-end: : 5060 Far-end: : 5060 H.245 Near: H.245 Far: H.245 Signaling Loc: H.245 Tunneled in Q.931? no Audio Connection Type: ip-direct Authentication Type: None Near-end Audio Loc: Codec Type: G.711MU Audio IP Address Port Near-end: : 5200 Far-end: : Verify Session Manager In System Manager, expand the Elements menu on the left then select Session Manager System Status SIP Entity Monitoring. Verify that none of the links to the defined SIP entities are down, indicating that they are all reachable for call routing. In the sample screen below, SM1 shows one SIP Entity link down. 81 of 88

82 Select the corresponding Avaya Session Manager (SM1 in this example) to view the Entity Link that is down and the Reason Code. The Reason Code reflects the result of Session Manager sending a SIP OPTIONS message to that SIP Entity. 82 of 88

83 9.3 Verify Communication Server 1000 In Element Manager, under the Tools menu on the left, select Services->Logs and Reports->IP Telephony Nodes. Click Status for the SS_Node to verify that the signaling server is enabled and operational. See Section 11 for documentation on verification of successful PRI QSIG trunk configuration. 9.4 Verify AudioCodes Mediant 1000 Use the web interface to verify that the QSIG trunk to the Communication Server 1000 is up. Select the Home icon and verify that the connector icon of the T1 trunk, shown below in Slot 1, is green in color. If not, use the colored legend on the page to determine what the error condition is and check the cabling and signaling parameters (e.g., framing, line code, clock master, network/user, etc.) of the AudioCodes M1000 and Communication Server of 88