Abstract. Avaya Solution & Interoperability Test Lab

Transcription

1 Avaya Solution & Interoperability Test Lab Configuring SIP Trunks in Avaya Aura Session Manager R6.1, Avaya Communication Server 1000E R7.5, and Avaya Aura Conferencing Standard Edition R6.0 to support Audio Conferencing Issue 1.0 Abstract These Application Notes present a sample configuration for a network that uses Avaya Aura Session Manager R6.1 to route calls between Avaya Communication Server 1000E R7.5 conference participants and Avaya Aura Conferencing Standard Edition R6.0 to support audio conferencing. For the sample configuration, Avaya Communication Server 1000E Call Server and Signaling Server applications run co-resident on a single Common Processor Pentium Mobile (CPPM) card, and Avaya Aura Conferencing Standard Edition audio bridging and conference reservation applications run co-resident on a single Avaya S8800 server. The validation test of the sample configuration was conducted at the Avaya Solution and Interoperability Test Lab at the request of Avaya Communication Server 1000 Product Management. 1 of 56

2 1. Introduction These Application Notes present a sample configuration for a network that uses Avaya Aura Session Manager R6.1 to route calls between Avaya Communication Server 1000E R7.5 (hereafter referred to as CS1000E) conference participants and Avaya Aura Conferencing Standard Edition R6.0 to support audio conferencing. As shown in the Figure 1, all telephones are supported by the Communication Server 1000E, which was configured as a co-resident single server system. SIP trunks are used to connect the CS1000E Signaling Server and the Avaya Aura Conferencing bridge port to the signaling port of Avaya Aura Session Manager over the LAN. Signaling messages are carried over these TCP-based SIP trunks while DTMF tones are transmitted within the RTP stream using RFC 2833 compliant signaling. The telephones are configured in the 20xxx extension range, while the conference access numbers on Avaya Aura Conferencing are in the range 390xx. Figure 1: Sample Network Configuration These Application Notes will focus on configuration of the SIP trunks and associated call routing. Detailed installation and basic administration of the products will not be described (see the appropriate documents listed in Section 9). 2 of 56

3 2. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Equipment Avaya S8800 Server Avaya S8800 Server Avaya S8800 Server Software Version Avaya Aura Conferencing Standard Edition R6.0 Load conferencing-patch-aif bundle s6x00-management-patch crs-patch awc-patch webapps-config-patch Avaya Aura System Manager R6.1.0 (Build No ) Avaya Aura Session Manager R6.1 Load Release 7.5, Version Avaya Communication Server nortel-cs1000-linuxbase E Service_Pack_Linux_ Deplists_CPL_X21_07_50Q Avaya 1120e IP Deskphone UNIStim: 0624C8A Avaya 1165e IP Deskphone UNIStim: 0624C8A Avaya 1230 IP Deskphone SIP: Avaya 2007 IP Deskphone UNIStim: 0621C7D Avaya one-x Communicator CS GA Avaya 3903 Digital Phone N/A Table 1: Version Numbers of Equipment and Software 3 of 56

4 3. Configure Avaya Communication Server 1000E The CS1000E uses the Signaling Server to provide SIP and H.323 signaling interfaces to IP networks. The Signaling Server communicates with a Call Server over a private Ethernet interface. There can be one or more Signaling Servers supported per CS1000E system. The applications that can run on the Signaling Server include the following: SIP Gateway Element Manager Provides SIP signaling for IP networks. Provides web interface for system administrative tasks. In the sample configuration, the CS1000E contained one Signaling Server and Call Server coresident on the same CS1000E CPPM server. Avaya provides a single web GUI which runs on the System Manager, called Unified Communications Management (UCM) Services, for the provisioning of the various telephony software components. Within this application, the Element Manager was used to configure system resources such as SIP virtual routes and trunks. See Reference [1] for additional information on this application. These Application Notes assume that basic configuration for the Signaling Server, Call Server, and the Coordinated Dial Plan (CDP) feature has been completed, and therefore this configuration will not be described. Formally the Network Routing Service (NRS) was used to route calls from CS1000E to Aura Conferencing. From CS onwards we will use Session Manager for this functionality. The procedures below describe the details of configuring SIP trunks and call routing on the CS1000E: Launch Unified Communications Manager Obtain Node and IP addresses Administer ISDN Administer D-Channel Administer Zones Administer Virtual SIP Routes and Trunks Administer Route List Block and Distant Steering Code Administer Node SIP parameters 4 of 56

5 3.1. Launch Communication Server 100E Unified Communications Manager Access the Unified Communications Manager (UCM) web based interface by using the URL in an Internet browser window, where <ip-address> is the IP address of the Linux Server. Log in with the appropriate user name and password. If System Manager (SMGR) 6.1 with UCM functionality is being used click on UCM Services as shown in the following screen. The following Unified Communications Management screen will be displayed. Click on the Element Name corresponding to the Element Type of CS of 56

6 3.2. Obtain Node and IP Addresses The Element Manager System Overview screen is displayed. Expand the IP Network menu under System on the left pane and select Nodes: Servers, Media Cards. The IP Telephony Nodes screen is displayed. Click on the appropriate Node ID to expand it. Note: The node number and IP address may vary. 6 of 56

7 The Node Details screen is updated with additional details as shown below. Make a note of the Node IPv4 address and Signaling Server TLAN IPv4 address. These values are used to configure other sections. 7 of 56

8 3.3. Administer ISDN Select Customers in the left pane. The Customers screen is displayed. Click the link associated with the appropriate customer, in this case 00. The system can support more than one customer with different network settings and options. In the sample configuration, only one customer was configured on the system. 8 of 56

9 The Customer Details screen is displayed next. Select Feature Packages. 9 of 56

10 The Feature Packages screen is displayed, containing a listing of feature packages (not all features shown below). Select Integrated Services Digital Network to edit its parameters. The screen is updated with parameters populated below Integrated Services Digital Network. Check the Integrated Services Digital Network checkbox, and retain the default values for all remaining fields. Scroll down to the bottom of the screen, and click Save (not shown). 10 of 56

11 3.4. 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 example; 1 ). Click to Add. 11 of 56

12 The D-Channels 1 Property Configuration screen is displayed next. Enter the following values for the specified fields, and retain the default values for the remaining fields. The screen below displays the parameters after the D-channel was added. D channel Card Type: D-Channel is over IP (DCIP) Designator: A descriptive name Click on Basic options (BSCOPT) to expand its parameters, and then click the EDIT button next to Remote Capabilities. Check the checkbox next to Network name display method 3 (ND3). Click on Return Remote Capabilities. Then Click on Submit (not shown). 12 of 56

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14 3.5. Administer Zones Select System IP Network Zones from the left pane to display the Zones screen, and then select Bandwidth Zones, followed by Add (not shown). The Zone Basic Property and Bandwidth Management screen is displayed next. Enter the following values for the specified fields, retain the default values for the remaining fields, and click Submit. Zone Number: Enter an unused zone number Zone Intent (ZBRN): select VTRK (VTRK) from the drop-down list, indicating that the zone is intended to be used for the virtual SIP trunk Description (ZDES): A descriptive name 14 of 56

15 3.6. Administer Virtual SIP Routes and Trunks Add Superloop A virtual superloop must be added to support the virtual SIP trunk (VTRK) created later in Section Select System Core Equipment Superloops from the left pane and select Add from the Superloops screen that is displayed (not shown). Select an unused Superloop number and select Virtual for the Superloop type. Click Save (not shown) Add Route 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. 15 of 56

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17 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 Designator field for trunk (DES): 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 This route is for a virtual trunk route (VTRK): Check the checkbox Zone for codec selection and bandwidth Management (ZONE): Select the zone defined in Section 3.5 Node ID of signaling server of this route (NODE): Select the node from Section 3.2 Protocol ID for the route (PCID): SIP (SIP) 17 of 56

18 Scroll down the screen, check the Integrated services digital network option (ISDN) checkbox to enable additional fields to appear. 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 (not shown). Mode of operation (MODE): Route uses ISDN Signaling Link (ISLD) D channel number (DCH): D-Channel number from Section 3.4. Network calling name allowed (NCNA): Check the checkbox. Network call redirection (NCRD): Check the checkbox Add Trunk After the route is created, 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. 18 of 56

19 The Customer 0, Route 1, New Trunk Type TIE trunk data block configuration screen is displayed. Enter the following values for the specified fields, and retain the default values for the remaining fields. The Multiple trunk input number field may be used to add multiple trunks in a single operation, or one can repeat the operation for each trunk. In the sample configuration, ten trunks were created, and the Auto increment member number checkbox is checked by default. Trunk data block: IP Trunk (IPTI) Terminal number: An available terminal number using the super loop created in Section Designator field for trunk: Descriptive text Member number: Starting member number Card Density Select Octal Density (8D) Start arrangement Incoming: Wink or Fast Flash (WNK) Start arrangement Outgoing: Wink or Fast Flash (WNK) Channel ID for this trunk: An available starting channel ID Click on the Edit button next to Class of Service, and on the screen that follows, set Media Security to Media Security Never (MSNV). This is required since SRTP media encryption was not configured on Avaya Conferencing. Click Return Class of Service followed by Save (not shown). 19 of 56

20 3.7. Administer Route List Block and Distant Steering Code 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 as shown below. In the Please enter a route list index field; enter an available route list block number (in this case 1 ). Click to Add. 20 of 56

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22 The Route List Block screen is displayed with a listing of parameters. For the Route Number field, select the route number from Section Retain the default values for the remaining fields, scroll down to the bottom of the screen and click Submit (not shown). 22 of 56

23 Select Dialing and Numbering Plans Electronic Switched Network again from the left pane to display the Electronic Switched Network (ESN) screen. Select Distant Steering Code (DSC) to add an entry to route 390xx calls to Avaya Conferencing. 23 of 56

24 The Distant Steering Code List screen is displayed next. In the Please enter a distant steering code field; enter the dialed prefix digits to match on (in this case 390 ). Click to Add. The Distant Steering Code screen is displayed. For the Route List to be accessed for trunk steering code field, select the route list index (configured in section 3.7) from the drop-down list. Retain the default values in all remaining fields and click on Submit. 24 of 56

25 3.8. Administer Node SIP Parameters Select System IP Network Nodes: Servers, Media Cards Configuration from the left pane, and in the IP Telephony Nodes screen displayed (not shown), select the node ID of this CS1000E system (see Section 3.2). The Node Details screen is displayed. It is assumed that the TLAN and ELAN IP addresses have already been configured as a result of basic configuration of the Signaling Server. Scroll down the top pane and click on Voice Gateway (VGW) and Codecs. 25 of 56

26 In the following screen, verify the default options shown under General. Scroll down the parameters box and check the desired codecs under Voice Codecs. Note that G.711 A- and mu-law were verified for the sample configuration and are the only codecs supported by Avaya Aura Conferencing. G.711 is checked by default and will cause the CS1000E to offer both companding algorithms. Click on Save. 26 of 56

27 When the Node Details screen is displayed again, scroll down the top pane and click on Gateway (SIPGw & H323Gw). Under General on the Virtual Trunk Gateway Configuration Details screen, enter the following values for the specified fields, and retain the default values for the remaining fields. Vtrk gateway application: Select SIP Gateway (SIPGw) SIP domain name Domain name also used in Section 5.2 Local SIP port 5060 Gateway endpoint name An appropriate name Gateway password Enter a password 27 of 56

28 Click on SIP Gateway Settings, and under Proxy or Redirect Server and Proxy Server Route 1, enter the following values for the specified fields, and retain the default values for the remaining fields. Primary TLAN IP address: The IP address of the Session Manager Port: 5060 Transport protocol: TCP 28 of 56

29 Scroll down the parameters box to the SIP URI Map section. Under Public E.164 domain names, enter the appropriate National and Subscriber values for the network configuration. In the test configuration, 353 is the country code and 91 is the area code. The remaining fields can be left at their default values. Click on Save. Return to the Node Details screen and click Save, as shown below. 29 of 56

30 The Node Saved screen is displayed. Click Transfer Now. The Synchronize Configuration Files screen is displayed. Select the Signaling Server and click on Start Sync. When the Synchronization Status column shows Synchronized, select the Signaling Server and click on Restart Applications. 30 of 56

31 4. Configure Avaya Aura Conferencing Standard Edition This section describes the steps for configuring Avaya Conferencing to interoperate with Session Manager and the CS1000E to support audio conferencing for CS1000E station users. The focus is primarily on basic interoperability and call routing and not configuration of the various audio conferencing features supported by Avaya Conferencing. A list of the functionality that was verified is provided in Section 7. It is assumed that Avaya Conferencing is installed, configured, and licensed as per Reference [2]. The administrative procedures include the following areas: Log in to Avaya Aura System Manager Administer SIP signaling options Administer incoming SIP URI conversion rules Administer access numbers Administer Dial-Out Apply changes 4.1. Log in to Avaya Aura System Manager The Avaya Aura Conferencing server contains an instance of System Manager that has been tailored for audio conferencing, but has the same look and feel as the System Manager used to configure Avaya Aura Session Manager. To access the Avaya Aura Conferencing System Manager, enter into a web browser, where <ip-address> is the IP address of the Conferencing System Manager virtual machine ( in the sample configuration, see Figure 1). Log in using appropriate credentials Administer SIP Signalling Options There are several SIP signaling options used by Avaya Conferencing when handling calls. These options govern SIP session length and contact/display information used when calls are made or answered. Select Elements Conferencing Media Configuration on the left panel menu. Check the box next to Bridge and then click the Configure button which will become enabled. 31 of 56

32 Expand the SIP section in the subsequent screen and complete the following fields as necessary: SIP Listener URI Response Contact This URI be used in the From header when Avaya Conferencing places calls out to conferees, and the user portion (e.g ) will be displayed as the calling party on conferees telephones. The IP address and transport of the Bridge should be consistent with that specified in Session Manager when adding the SIP Entity and Entity Links for Avaya Conferencing (see Sections 5.3 and 5.4). In the sample configuration, this field was set to <sip:39000@ :5060;transport=tcp>, so that the general access number of Avaya Conferencing is displayed. This URI will be used in the Contact header when Avaya Conferencing answers calls from conferees, and the user portion (e.g., ) will be displayed as the answering party on conferees telephones. Again, the IP address of and transport of the Bridge should be consistent with Sections 5.3 and 5.4. Session Refresh Timer: 1800 (a default value widely used by other SIP signaling systems, and which can eliminate unnecessary SIP messages to re-negotiate session refresh intervals). Min Session Refresh Timer Allowed: 1800 Click Save to save these settings. Click Add More changes on the subsequent screen (not shown) if proceeding to the following step(s). Click Apply Changes if not. 32 of 56

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34 4.3. Administer Access Numbers Call Branding refers to the definition of access numbers with which are associated a specific combination of conferencing features and user experience. Avaya Conferencing administrators and operators can then schedule conferences that use these conferencing attributes. To administer an access number, navigate to Elements Conferencing Audio Conferencing Call Routing. Click on the Edit button next to Call Branding and then click on Add on the following screen (not shown). Fill in the following fields and click Save: DDI: Name: On entry: The access telephone number for this conference type. Note that this should be consistent with the Dial Plan and Routing Policy configured in Session Manager (see Sections 5.5 and 5.6). A meaningful name. The type of treatment conferees receive on entry to the audio conference. In the example below ( Scan call flow ), callers are prompted for their conference pass code. Defaults can be used for the remaining fields, which can be used to modify prompt messages and behaviour when failures occur. See Reference [2] for more information on Call Branding and conference options. 34 of 56

35 4.4. Administer Dial-Out Dial-out options govern how Avaya Conferencing makes calls to conferees for conference setup. This is used when a conference operator wishes to request a telephone user to join an existing conference, or a conference moderator initiates an outbound call (blast dial) to a preconfigured list of internal extensions and/or PSTN users. To administer dial-out options, navigate to Elements Conferencing Audio Conferencing Call Routing. Click on the Edit button next to Telnum to URI. Then click on Add, and fill in the following fields, and click Save: Telnum: URI: Comment: A match specification for the telephone number that will be dialed. The same wildcard asterisk notation as in Section 4.3 can be used. The URI to be used in the outgoing INVITE to Session Manager. Note that the IP address and port number should be consistent with the SIP Entity Link configured in Session Manager for Avaya Conferencing (see Section 5.4) Optional comment The screen below shows the single rule defined for all outbound calls. Since all SIP signaling is directed to Session manager, a single wildcard representing any dialed number is used as the match specification, and the URI specified corresponds to: sip:{telnum}@{ip address:port of Session Manager SM-100 interface};transport=tcp. Click Done. Note that although not tested in the sample configuration, UDP and TLS transport are also supported by Avaya Conferencing. 35 of 56

36 To enable dialling out from a conference, navigate to Elements Conferencing Audio Conferencing Conference Features. Expand the Conference Settings section, and use the pull-down menu for Originator Dial Out to select All. Click Save. 36 of 56

37 4.5. Apply Changes At various times throughout the steps outlined in Section 4, the Apply Changes screen may appear, giving the user an opportunity to save the configuration done thus far. A typical screen is shown below. In general, one can click Add more changes if it is desired to continue on with the configuration steps. When configuration is complete, clicking Apply Changes will cause all changes to be saved, and may also require the Bridge (in this case Host ) to be restarted. 5. Configure Avaya Aura Session Manager This section provides the procedures for configuring Session Manager. It is assumed that the basic Session Manager installation and configuration has been completed. For further information on Session Manager, see reference [3]. The procedures include the following areas: Login to Avaya Aura System Manager Administer SIP domain Administer Adaptations Administer SIP Entities Administer Entity Links Administer Routing Policies Administer Dial Patterns 37 of 56

38 5.1. Log in to Avaya Aura System Manager Access the Avaya Aura System Manager used to manage Session Manager. Using a Web Browser, enter where <ip-address> is the IP address of System Manager ( in the sample configuration). Log in using appropriate credentials. Select Routing 38 of 56

39 5.2. Administer SIP Domain Add the SIP domain for which the communications infrastructure will be authoritative by selecting Domains on the left panel menu and clicking the New button (not shown) to create a new SIP domain entry. This entry should be consistent with that configured in the CS1000E (see Sections 3.8) and Avaya Aura Conferencing (see Sections 4.2 and 4.5). Complete the following options: Name: Type: Notes: The authoritative domain name (e.g., avaya.com). Use the default type of sip. Description for the domain (optional). Click Commit to save changes. Note: Since the sample network does not deal with any foreign domains, no additional SIP Domain entries are needed Administer Adaptations To enable calls between phones on Avaya Communication Server 1000E and SIP endpoints registered to Session Manager, Session Manager should be configured to use an Adaptation Module designed for Avaya Communication Server 1000E to convert SIP headers in messages sent by Avaya Communication Server to the format used by Avaya Aura Conferencing. Expand Elements Routing and select Adaptations from the left navigational menu. Click New (not shown). In the General section, enter the following values and use default values for remaining fields. Adaptation Name: Enter an identifier for the Adaptation Module 39 of 56

40 Module Name: Module Parameter: Select CS1000Adapter from drop-down menu. If CS1000 is not shown, select <click to add module> from the Module Name drop-down menu and then configure a module name called CS1000Adapter fromto=true In the Digit Conversion for Incoming Calls to SM section, click Add and enter the following values. Matching Pattern Enter dialed prefix for calls to Avaya Aura(r) Conferencing from Session Manager. In sample configuration, 3900 was used Min Enter minimum number of digits that must be dialed Max Enter maximum number of digits that may be dialled In the sample configuration, 5 was used Phone Context Enter value of Private CDP domain name defined in Section 3.8 e.g. cdp.udp Delete Digits Enter 0, unless digits should be removed from dialled number before call is routed by Session Manager Address to modify Select both Click Commit. The Adaptation Module defined for sample configuration is shown below. 40 of 56

41 Administer SIP Entities A SIP Entity must be added for Session Manager and for each SIP-based telephony system supported by a SIP Trunk. Select SIP Entities on the left panel menu and then click on the New button (not shown). Enter the following for each SIP Entity: Under General: Name: An informative name (e.g., SessionManager1 ). FQDN or IP Address: IP address of the SIP signaling interface. Time Zone: Time zone for this location. Note that although a Location was specified, it was not required and not pertinent to the sample configuration. For SIP Entities of Type Session Manager, under Port, click Add, and then edit the fields in the resulting new row: Port: Protocol: Default Domain: Port number on which the Session Manager listens for SIP requests. Transport protocol to be used to receive SIP requests. The domain (e.g., avaya.com). For the sample configuration, TCP was used for communicating with both the CS1000E and Avaya Conferencing. Defaults can be used for the remaining fields. Click Commit to save each SIP Entity definition. 41 of 56

42 The following screen shows the SIP Entity for Session Manager. The following screen shows the SIP Entity for the CS1000E. The IP address is that of the Signaling Server. Note the Adaptation is set to CS1000 as defined in Section of 56

43 The following screen shows the SIP Entity for Avaya Conferencing. The IP address is that of the Bridge. 43 of 56

44 5.4. Administer Entity Links A SIP trunk between Session Manager and a telephony system is described by an Entity Link. To add an Entity Link, select Entity Links on the left panel menu and click on the New button (not shown). Fill in the following fields in the new row that is displayed: Name: A descriptive name, e.g., CS1000E_R7.5. SIP Entity 1: Select the Session Manager. Protocol: Select TCP. Port: Type The Session Manager will listen for SIP requests on TCP port SIP Entity 2: Select one of the SIP Entities created in the previous section. Port: Port number to which the other system receives SIP requests. Trusted: Verify that this box is checked; otherwise SIP calls will be denied. Notes (Optional): A description for the Entity Link. Click Commit to save changes. The following screens show the Entity Links used in the sample configuration for the CS1000E and Avaya Aura Conferencing. 44 of 56

45 5.5. Administer Routing Policies Create routing policies to direct how calls will be routed to a system. Two routing policies must be added, one for CS1000E users and one for access numbers for Avaya Aura Conferencing. To add a routing policy, select Routing Policies on the left panel menu and then click on the New button (not shown). Under General: Name: Descriptive name, e.g., To CS1000E_R7.5. Notes (Optional): Brief description. Under SIP Entity as Destination: Click Select, and then select the appropriate SIP Entity created in Section 5.3 to which this routing policy applies. Under Time of Day: Click Add, and select the default 24/7 time range. Defaults can be used for the remaining fields. Click Commit to save changes. The following screen shows the Routing Policy Details for the CS1000E. 45 of 56

46 The following screen shows the Routing Policy Details for Avaya Aura Conferencing. 46 of 56

47 5.6. Administer Dial Patterns A dial pattern must be defined that will direct calls to the appropriate telephony system. In the sample configuration, 5-digit extensions beginning with 20 reside on the CS1000E. 5-digit extensions in the range 390xx are used for access to Avaya Aura Conferencing. To add a dial pattern, select Dial Patterns on the left panel menu and click on the New button (not shown) on the right. Fill in the following, as shown in the screens below: Under General fill in the following fields: Pattern: Dialed number or prefix to be matched. Min: Minimum length of dialled number. Max: Maximum length of dialled number. SIP Domain: SIP domain specified in Section 5.2, or ALL if calls from other domains are allowed to that SIP entity. Under Originating Locations and Routing Policies: Click Add. Select the following entries: Originating Location Name: Select ALL Routing Policy Name: Select a Routing Policy created in Section 5.5. Default values can be used for the remaining fields. Click Commit to save the dial pattern. The following screen shows the dial pattern for the CS1000E. 47 of 56

48 The following screen shows the dial pattern for Avaya Aura Conferencing.. 48 of 56

49 6. Verification Steps This section provides tests that can be performed to verify proper configuration of the CS1000E, Avaya Aura Conferencing, and Avaya Aura Session Manager Verify Avaya Communication Server 1000E In the Element Manager web interface, select System IP Network Nodes: Servers, Media Cards Maintenance and Reports on the left pane. Click GEN CMD. The General Commands page is displayed. From the Group drop-down menu select Sip, from the Command drop-down menu select SIPGwShow select Sip from the next dropdown menu to the right, and click on RUN. 49 of 56

50 Confirm that Primary Proxy IP Address is that of the Session Manager and that TCP port 5060 is used for SIP message transport. Next, the status of the D-Channel for the SIP trunk to Session Manager can be verified by navigating to System Maintenance, selecting Select by Overlay, LD 96 D-Channel, and D-Channel Diagnostics. 50 of 56

51 The screen below shows the APPL_STATUS of the D-Channel as OPER and the LINK_STATUS as ACTV Verify Avaya Aura Conferencing Standard Edition The operational status of Avaya Conferencing can be verified at three levels: 1. Use the Avaya System Platform GUI to verify that all virtual machines are running and to reboot any of them if necessary. The primary virtual machine for audio conferencing is bridge. Click on bridge to obtain a screen that will allow rebooting. 2. The Conferencing System Manager can be used to verify the status of the software application running on each virtual machine. Navigate to Elements Conferencing Services. Each service can be stopped and restarted if necessary by checking it and then selecting Stop Service(s) or Stop Service(s) as appropriate. The primary service for audio conferencing is the Bridge. 51 of 56

52 3. Configure a mirror port on the bridge port (eth2) of the Avaya Aura Conferencing Server, and monitor the SIP signaling messages resulting when calls are made into or out from Avaya Aura Conferencing. If a mirror port is not available, then use ssh (e.g., via Putty) to access the command line interface on the eth2 server interface (IP address in the sample configuration). Run the command: tcpdump n s 0 vvv w /tmp/{filename} ip host {SM IP address} where filename is the name of a trace file that will be created and SM IP Address is the IP address of the Session Manager SM-100 interface ( in the sample configuration). This file can be opened in Wireshark to view the SIP messages. 1 1 Note that the tcpdump command is only available to the root user - contact Avaya Global Services if configuring a mirror port on the bridge port is not feasible. 52 of 56

53 6.3. Verify Avaya Aura Session Manager Log in to Avaya Aura System Manager. Navigate to Elements Session Manager System Status SIP Entity Monitoring on the left pane to verify that none of the links to the defined SIP entities is down. The SIP Link Entity Link Monitoring Status Summary screen is displayed. Click on the individual SIP Entities to check their status. The following screens display the SIP Entity, Entity Link Connection Status of the Avaya Aura Conferencing and CS1000E Entities. Check the Conn. Status column and make sure it displays Up. 53 of 56

54 Troubleshooting SIP signaling may best be accomplished by port mirroring of the Session Manager SM-100 interface, and capturing the signaling using a protocol sniffer. Below is an example of SIP message flow for a call from an Avaya 1140E IP Deskphone on the CS1000E into Avaya Aura Conferencing via Avaya Aura Session Manager. Note that the URI of the INVITE from the CS1000E in frame 34 that is routed by Session Manager to Avaya Aura Conferencing in frame 40 contains the phone context parameters mentioned in Section 4.3, which the URI to Telnum rules added to Avaya Aura Conferencing will remove. Note also that MCDN parameters not supported by Avaya Aura Conferencing have been included in the SDP, to which Avaya Aura Conferencing replies with 415 Unsupported Media Type (frame 42). The CS1000E acknowledges this and responds with an INVITE without those parameters in frame 54, and the call is successful. 54 of 56

55 7. Verification Scenarios Verification scenarios for the configuration described in these Application Notes included: Conference calls including various telephone types (see Figure 1) on the Avaya Communication Server 1000E can be made using G.711mu- and A-law. Automatic gain, silence suppression, and comfort noise. Scan, Flex, and Direct To Conference modes. Name Record/Play (NRP). RFC 2833 DTMF support for all moderator and conferee commands. Manual and automatic blast dial-out to conference participants. Network outage failure and recovery. Bridge process ( softms ) failure and recovery. 8. Conclusion As illustrated in these Application Notes, Avaya Communication Server 1000E and Avaya Aura Conferencing can be used together in an integrated solution with Avaya Aura Session Manager. 9. Additional References Avaya Communication Server 1000E ( [1] Nortel Communication Server Element Manager System Reference Administration, Release: 7.5, Document Revision: 05.04, Document #NN Avaya Aura Conferencing Standard Edition ( [2] Implementing Avaya Aura Conferencing, Document # , June, Avaya Aura Session Manager ( [3] Administering Avaya Aura Session Manager, Doc # , Issue 3, June, of 56

56 Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by and are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya Solution & Interoperability Test Lab at 56 of 56