Application Notes For Configuring Avaya Aura Communication Manager R7 .

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Avaya Solution & Interoperability Test LabApplication Notes for Configuring Avaya Aura Communication Manager R7.0, Avaya Aura SessionManager 7.0 and Avaya Session Border Controller forEnterprise R7.0 to support Colt SIP Trunk - Issue 0.1AbstractThese Application Notes describe the steps used to configure Session Initiation Protocol (SIP)trunking between Colt SIP Trunk and an Avaya SIP enabled Enterprise Solution. The Avayasolution consists of Avaya Session Border Controller for Enterprise, Avaya Aura SessionManager and Avaya Aura Communication Manager as an Evolution Server. Colt is amember of the DevConnect Service Provider program.Information in these Application Notes has been obtained through DevConnect compliancetesting and additional technical discussions. Testing was conducted via the DevConnectProgram at the Avaya Solution and Interoperability Test Lab.BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.1 of 59Colt CM70 SM

1. IntroductionThese Application Notes describe the steps used to configure Session Initiation Protocol (SIP)trunking between the Colt SIP Trunk and an Avaya SIP-enabled enterprise solution. The Avayasolution consists of the following: Avaya Aura Communication Manager R7.0(Communication Manager); Avaya Aura Session Manager R7.0 (Session Manager); AvayaSession Border Controller for Enterprise R7.0 (Avaya SBCE). Note that the shortened namesshown in brackets will be used throughout the remainder of the document. Customers using thisAvaya SIP-enabled enterprise solution with the Colt SIP Trunk are able to place and receivePSTN calls via a dedicated Internet connection and the SIP protocol. This converged networksolution is an alternative to traditional PSTN trunks. This approach generally results in lowercost for the enterprise customer.2. General Test Approach and Test ResultsThe general test approach was to configure a simulated enterprise site using an Avaya SIPtelephony solution consisting of Communication Manager, Session Manager and Avaya SBCE.The enterprise site was configured to connect to the Colt SIP Trunk.DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. Thejointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinentto the interoperability of the tested products and their functionalities. DevConnect ComplianceTesting is not intended to substitute full product performance or feature testing performed byDevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability orcompleteness of a DevConnect member’s solution.2.1. Interoperability Compliance TestingThe interoperability test included the following: Incoming calls to the enterprise site from PSTN phones using the Colt SIP Trunk, callsmade to SIP and H.323 telephones at the enterprise. Outgoing calls from the enterprise site completed via the Colt SIP Trunk to PSTNdestinations, calls made from SIP and H.323 telephones. Calls using the G.729A, G.711A and G.726-32 codecs. Fax calls to/from a group 3 fax machine to a PSTN connected fax machine using T.38. DTMF transmission using RFC 2833 with successful Voice Mail/Vector navigation forinbound and outbound calls. User features such as hold and resume, transfer, conference, call forwarding, etc. Caller ID Presentation and Caller ID Restriction. Direct IP-to-IP media between the Avaya SBCE and the SIP and H.323 telephones. Call coverage and call forwarding for endpoints at the enterprise site. Transmission and response of SIP OPTIONS messages sent by the Colt SIP Trunkrequiring Avaya response and sent by Avaya requiring Colt response.BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.2 of 59Colt CM70 SM

2.2. Test ResultsInteroperability testing of the sample configuration was completed with successful results for theColt SIP Trunk with the following observations: No OPTIONS were received from the network during testing. This was noted as undercertain failure conditions, call failures may not be handled as effectively as possible. When testing outbound calls with no matching codec, the network responded to theINVITE with a 180 Ringing and an alternative codec listed in the SDP. A moreappropriate response in this case is “488 Not Acceptable Here”. Communication Managersent a CANCEL and failure tone was heard on the calling phone. No inbound Toll-Free access was available to test. Routing was not in place to test Operator or Directory Enquiries calls. Emergency calls were not tested as there was no test call booked with the EmergencyServices Operator Initial testing of outbound T.38 Fax calls was unsuccessful. When the network sent a reINVITE to change to T.38 and Communication Manager responded with 200 OK, thenetwork did not send an ACK. After repeated sending of 200 OK, CommunicationManager released the call. A fix was put in place by Colt and outbound Fax was retestedsuccessfully. When testing congestion and failure of the SIP Trunk, it was approximately 15 secondsbefore a failure tone was heard on the calling phone. This was because the call was reattempted from the network a number of times before it was rejected.2.3. SupportFor technical support on Colt products please contact Colt on0800 358 3999 or visit their website at www.colt.netBG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.3 of 59Colt CM70 SM

3. Reference ConfigurationFigure 1 illustrates the test configuration. The test configuration shows an Enterprise siteconnected to the Colt SIP Trunk. Located at the Enterprise site is an Avaya SBCE, SessionManager and Communication Manager. Endpoints are Avaya 96x0 series and Avaya 96x1 seriesIP telephones (with SIP and H.323 firmware), Avaya 16xx series IP telephones (with H.323firmware), Avaya analogue telephones and an analogue fax machine. Also included in the testconfiguration was an Avaya one-X Communicator soft phone and Avaya Communicator forWindows running on laptop PCs.Soft-SwitchMedia GatewayPSTNColt LANSession Border ControllerABC1External IP: 9#TelephoneColt SIP TrunkServiceSignallingMediaPublicInternetAvaya Labs Simulating anEnterprise Customer SiteAvaya Session BorderController for Enterprise(External: 192.168.122.55Internal: 10.10.9.71)Avaya Aura Session Manager(10.10.9.31)Avaya Aura System Manager(10.10.9.17)ESTESTESTAvaya Lab Private LanAvaya Aura or WindowsAvaya one-X CommunicatorESTAvaya G430 Media OMPACT FLASHUSBMDMA UDIOA LMCPURSTA SBCARDIN US EPWR1AvayaAnalogueTelephone2CCA 10/1SERVICESLAN 10/3Series 3 FaxABC13JKL4Avaya 9600 SeriesSIP & H.323TelephonesDEF2GHIAvaya 1600Series F3MNO456PQRSTUVWXYZ78#9[0ABC9GHI**0#Figure 1: Test Setup Colt SIP Trunk to Avaya EnterpriseBG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.4 of 59Colt CM70 SM

4. Equipment and Software ValidatedThe following equipment and software were used for the sample configuration provided:Equipment/SoftwareAvayaAvaya Aura Session ManagerAvaya Aura System ManagerAvaya Aura Communication ManagerAvaya Session Border Controller forEnterpriseAvaya G430 Media GatewayAvaya 96x0 Phone (SIP)Avaya 9608 Phone (SIP)Avaya 96x0 Phone (H.323)Avaya 9608 Phone (H.323)Avaya 1616 Phone (H.323)Avaya One-X CommunicatorAvaya Communicator for WindowsAvaya 2400 Series Digital HandsetsAnalogue HandsetAnalogue FaxColtSonus GSXSonus PSXBG; Reviewed:RRR -441 Build 0.224777.0.0-21-660237.19.02 6 14 57.0.0 9.2.4V08.04.08A002Solution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.5 of 59Colt CM70 SM

5. Configure Avaya Aura Communication ManagerThis section describes the steps for configuring Communication Manager for SIP Trunking. SIPtrunks are established between Communication Manager and Session Manager. These SIP trunkswill carry SIP signalling associated with the Colt SIP Trunk. For incoming calls, the SessionManager receives SIP messages from the Avaya SBCE and directs the incoming SIP messages toCommunication Manager. Once the message arrives at Communication Manager furtherincoming call treatment, such as incoming digit translations and class of service restrictions maybe performed. All outgoing calls to the PSTN are processed within Communication Manager andmay be first subject to outbound features such as automatic route selection, digit manipulationand class of service restrictions. Once Communication Manager selects a SIP trunk, the SIPsignalling is routed to the Session Manager. The Session Manager directs the outbound SIPmessages to the Avaya SBCE at the enterprise site that then sends the SIP messages to the Coltnetwork. Communication Manager configuration was performed using the System AccessTerminal (SAT). Some screens in this section have been abridged and highlighted for brevity andclarity in presentation. The general installation of the Servers and Avaya G430 Media Gatewayis presumed to have been previously completed and is not discussed here.5.1. Confirm System FeaturesThe license file installed on the system controls the maximum values for these attributes. If arequired feature is not enabled or there is insufficient capacity, contact an authorized Avaya salesrepresentative to add additional capacity. Use the display system-parameters customer-optionscommand and on Page 2, verify that the Maximum Administered SIP Trunks supported by thesystem is sufficient for the combination of trunks to the Colt SIP Trunk, and any other SIP trunksused.display system-parameters customer-optionsOPTIONAL FEATURESPageIP PORT CAPACITIESMaximum Administered H.323 Trunks:Maximum Concurrently Registered IP Stations:Maximum Administered Remote Office Trunks:Maximum Concurrently Registered Remote Office Stations:Maximum Concurrently Registered IP eCons:Max Concur Registered Unauthenticated H.323 Stations:Maximum Video Capable Stations:Maximum Video Capable IP Softphones:Maximum Administered SIP Trunks:Maximum Administered Ad-hoc Video Conferencing Ports:Maximum Number of DS1 Boards with Echo Cancellation:BG; Reviewed:RRR m/d/y4000240040002400681002400240040004000802 of12USED030000002000Solution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.6 of 59Colt CM70 SM

On Page 5, verify that IP Trunks field is set to y.display system-parameters customer-optionsOPTIONAL FEATURESEmergency Access to Attendant?Enable 'dadmin' Login?Enhanced Conferencing?Enhanced EC500?Enterprise Survivable Server?Enterprise Wide Licensing?ESS Administration?Extended Cvg/Fwd Admin?External Device Alarm Admin?Five Port Networks Max Per MCC?Flexible Billing?Forced Entry of Account Codes?Global Call Classification?Hospitality (Basic)?Hospitality (G3V3 Enhancements)?IP Trunks?yyyynnyyynnyyyyyPage5 of12IP Stations? yISDN Feature Plus?ISDN/SIP Network Call Redirection?ISDN-BRI Trunks?ISDN-PRI?Local Survivable Processor?Malicious Call Trace?Media Encryption Over IP?Mode Code for Centralized Voice Mail?nyyynynnMultifrequency Signaling?Multimedia Call Handling (Basic)?Multimedia Call Handling (Enhanced)?Multimedia IP SIP Trunking?yyyyIP Attendant Consoles? y5.2. Administer IP Node NamesThe node names defined here will be used in other configuration screens to define a SIPsignalling group between Communication Manager and Session Manager. In the IP NodeNames form, assign the node Name and IP Address for the Session Manager. In this case,Session Manager and 10.10.9.31 are the Name and IP Address for the Session Manager SIPinterface. Also note the procr IP address as this is the processor interface that CommunicationManager will use as the SIP signalling interface to Session Manager.display node-names ipIP NODE NAMESNameSession Managerdefaultprocrprocr6BG; Reviewed:RRR m/d/yIP Address10.10.9.310.0.0.010.10.9.12::Solution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.7 of 59Colt CM70 SM

5.3. Administer IP Network RegionUse the change ip-network-region 1 command to set the following values: The Authoritative Domain field is configured to match the domain name configured onSession Manager. In this configuration, the domain name is avaya.com. By default, IP-IP Direct Audio (both Intra- and Inter-Region) is enabled (yes) to allowaudio traffic to be sent directly between endpoints without using gateway VoIP resources.When a PSTN call is shuffled, the media stream is established directly between theenterprise end-point and the internal media interface of the Avaya SBCE. The Codec Set is set to the number of the IP codec set to be used for calls within the IPnetwork region. In this case, codec set 1 is used. The rest of the fields can be left at default values.change ip-network-region 1Page1 of20IP NETWORK REGIONRegion: 1Location: 1Authoritative Domain: avaya.comName: defaultStub Network Region: nMEDIA PARAMETERSIntra-region IP-IP Direct Audio: yesCodec Set: 1Inter-region IP-IP Direct Audio: yesUDP Port Min: 2048IP Audio Hairpinning? nUDP Port Max: 3329DIFFSERV/TOS PARAMETERSCall Control PHB Value: 46Audio PHB Value: 46Video PHB Value: 26802.1P/Q PARAMETERSCall Control 802.1p Priority: 6Audio 802.1p Priority: 6Video 802.1p Priority: 5AUDIO RESOURCE RESERVATION PARAMETERSH.323 IP ENDPOINTSRSVP Enabled? nH.323 Link Bounce Recovery? yIdle Traffic Interval (sec): 20Keep-Alive Interval (sec): 5Keep-Alive Count: 5BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.8 of 59Colt CM70 SM

5.4. Administer IP Codec SetOpen the IP Codec Set form for the codec set specified in the IP Network Region form inSection 5.3 by typing change ip-codec set 1. Enter the list of audio codec’s eligible to be used inorder of preference. For the interoperability test the codecs supported by Colt were configured,namely G.729A, G.711A and G.726A-32K.change ip-codec-set 1Page1 of2IP CODEC SETCodec Set: 1AudioCodec1: G.729A2: G.711A3: G.726A-32K4:5:SilenceSuppressionnnnFramesPer Pkt222PacketSize(ms)202020The Colt SIP Trunk supports T.38 for transmission of fax. Navigate to Page 2 and define T.38fax as follows: Set the FAX - Mode to t.38-standard Leave ECM at default value of ychange ip-codec-set 1Page2 of2IP CODEC SETAllow Direct-IP Multimedia? nFAXModemTDD/TTYH.323 Clear-channelSIP 64K ize(ms)ECM: y20Note: Redundancy can be used to send multiple copies of T.38 packets which can help thesuccessful transmission of fax over networks where packets are being dropped. This was notexperienced in the test environment and Redundancy was left at the default value of 0.BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.9 of 59Colt CM70 SM

5.5. Administer SIP Signaling GroupsThis signalling group (and trunk group) will be used for inbound and outbound PSTN calls to theColt SIP Trunk. During test, this was configured to use TCP and port 5060 though it’srecommended to use TLS and port 5061 in the live environment to enhance security. Configurethe Signaling Group using the add signaling-group x command as follows: Set Group Type to sip. Set Transport Method to tcp. Set Peer Detection Enabled to y allowing Communication Manager to automaticallydetect if the peer server is a Session Manager. Set Near-end Node Name to the processor interface (node name procr as defined in theIP Node Names form shown in Section 5.2). Set Far-end Node Name to the Session Manager (node name Session Manager asdefined in the IP Node Names form shown in Section 5.2). Set Near-end Listen Port and Far-end Listen Port to 5060 (Commonly used TCP portvalue). Set Far-end Network Region to the IP Network Region configured in Section 5.3(logically establishes the far-end for calls using this signalling group as network region1). Leave Far-end Domain blank (allows Communication Manager to accept calls from anySIP domain on the associated trunk). Set Direct IP-IP Audio Connections to y. Leave DTMF over IP at default value of rtp-payload (Enables RFC2833 for DTMFtransmission from Communication Manager).The default values for the other fields may be used.add signaling-group 1Page1 of2SIGNALING GROUPGroup Number: 1Group Type: sipIMS Enabled? nTransport Method: tcpQ-SIP? nIP Video? nEnforce SIPS URI for SRTP? yPeer Detection Enabled? y Peer Server: SMPrepend ' ' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? yRemove ' ' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? nAlert Incoming SIP Crisis Calls? nNear-end Node Name: procrFar-end Node Name: Session ManagerNear-end Listen Port: 5060Far-end Listen Port: 5060Far-end Network Region: 1Far-end Domain:Incoming Dialog Loopbacks: eliminateDTMF over IP: rtp-payloadSession Establishment Timer(min): 3Enable Layer 3 Test? yH.323 Station Outgoing Direct Media? nBG; Reviewed:RRR m/d/yBypass If IP Threshold Exceeded?RFC 3389 Comfort Noise?Direct IP-IP Audio Connections?IP Audio Hairpinning?Initial IP-IP Direct Media?Alternate Route Timer(sec):Solution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.nnynn610 of 59Colt CM70 SM

5.6. Administer SIP Trunk GroupA trunk group is associated with the signaling group described in Section 5.5. Configure thetrunk group using the add trunk-group x command, where x is an available trunk group. OnPage 1 of this form: Set the Group Type field to sip. Choose a descriptive Group Name. Specify a trunk access code (TAC) consistent with the dial plan. The Direction is set to two-way to allow incoming and outgoing calls. Set the Service Type field to public-netwrk. Specify the signalling group associated with this trunk group in the Signaling Groupfield as previously configured in Section 5.5. Specify the Number of Members supported by this SIP trunk group.add trunk-group 1Page1 of21TRUNK GROUPGroup Number:Group Name:Direction:Dial Access?Queue Length:Service Type:1OUTSIDE CALLtwo-wayn0public-ntwrkGroup Type: sipCDR Reports: yCOR: 1TN: 1TAC: 101Outgoing Display? nNight Service:Auth Code? nMember Assignment Method: autoSignaling Group: 1Number of Members: 10On Page 2 of the trunk-group form, the Preferred Minimum Session Refresh Interval (sec)field should be set to a value mutually agreed with Colt to prevent unnecessary SIP messagesduring call setup. During testing, a value of 600 was used that sets Min-SE to 1200 in the SIPsignalling.add trunk-group 1Group Type: sipPage2 of21TRUNK PARAMETERSUnicode Name: autoRedirect On OPTIM Failure: 5000SCCAN? nDigital Loss Group: 18Preferred Minimum Session Refresh Interval(sec): 600Disconnect Supervision - In? yBG; Reviewed:RRR m/d/yOut? ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.11 of 59Colt CM70 SM

On Page 3, set the Numbering Format field to private. This allows delivery of CLI in formatsother than E.164 with leading “ ”. In test, CLIs were sent as Communication Manager extensionnumbers and were reformatted by the Session Manager in an Adaptation described in Section6.4. This format was successfully verified in the network.add trunk-group 1TRUNK FEATURESACA Assignment? nPage3 of21Measured: noneMaintenance Tests? yNumbering Format: privateUUI Treatment: service-providerReplace Restricted Numbers? nReplace Unavailable Numbers? nOn Page 4 of this form: Set Support Request History to y. Set Send Diversion Header to y. Note – History-Info and Diversion headers may notboth be required but were sent during compliance testing. Set the Telephone Event Payload Type to 100 to match the value preferred by Colt (thisPayload Type is not applied to calls from SIP end-points). Set the Identity for Calling Party Display to From to ensure that where CLI forincoming calls is withheld, it is not displayed on Communication Manager extension.add trunk-group 1Page4 of21PROTOCOL VARIATIONSMark Users as Phone?Prepend ' ' to Calling/Alerting/Diverting/Connected Number?Send Transferring Party Information?Network Call Redirection?nnynSend Diversion Header? ySupport Request History? yTelephone Event Payload Type: 100Convert 180 to 183 for Early Media?Always Use re-INVITE for Display Updates?Identity for Calling Party Display:Block Sending Calling Party Location in INVITE?Accept Redirect to Blank User Destination?Enable Q-SIP?nnFromnnnNote: - The above screenshot shows Network Call Redirection set to n. This was temporarilyset to y for some of the last tests that involved testing of 302 Moved Temporarily and REFERmessages. When set, REFER messages are sent that are not acted on by the Colt SIP Trunk andso are unnecessary additional signalling.BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.12 of 59Colt CM70 SM

5.7. Administer Calling Party Number InformationUse the change private-unknown-numbering command to configure Communication Managerto send the calling party number in the format required. In test, calling party numbers were sentas Communication Manager extension numbers to be modified in the Session Manager.Adaptations are used in Session Manager to format the number as described in Section 6.4.These calling party numbers are sent in the SIP From, Contact and PAI headers as well as theDiversion header for forwarded calls. The numbers are displayed on display-equipped PSTNtelephones with any reformatting performed in the network.change private-numbering 0Page1 of2NUMBERING - PRIVATE FORMATExt ExtLen Code4 2TrkGrp(s)1PrivatePrefixTotalLen4Total Administered: 1Maximum Entries: 5405.8. Administer Route Selection for Outbound CallsIn the test environment, the Automatic Route Selection (ARS) feature was used to routeoutbound calls via the SIP trunk to the Colt SIP Trunk. The single digit 9 was used as the ARSaccess code providing a facility for telephone users to dial 9 to reach an outside line. Use thechange feature-access-codes command to configure a digit as the Auto Route Selection (ARS)- Access Code 1.change feature-access-codesPageFEATURE ACCESS CODE (FAC)Abbreviated Dialing List1 Access Code:Abbreviated Dialing List2 Access Code:Abbreviated Dialing List3 Access Code:Abbreviated Dial - Prgm Group List Access Code:Announcement Access Code: *69Answer Back Access Code:Attendant Access Code:Auto Alternate Routing (AAR) Access Code: 8Auto Route Selection (ARS) - Access Code 1: 9Access Code 2:BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.1 of1013 of 59Colt CM70 SM

Use the change ars analysis command to configure the routing of dialled digits following thefirst digit 9. A small sample of dial patterns are shown here as an example. Furtheradministration of ARS is beyond the scope of this document. The example entries shown willmatch outgoing calls to numbers beginning 0. Note that exact maximum number lengths shouldbe used where possible to reduce post-dial delay. Calls are sent to Route Pattern 1.change ars analysis 0PageARS DIGIT ANALYSIS TABLELocation: allDialedString00011827000TotalMin pubupubupubuNodeNum1 of2Percent Full: 0ANIReqdnnnnnUse the change route-pattern x command, where x is an available route pattern, to add the SIPtrunk group to the route pattern that ARS selects. In this configuration, route pattern 1 is used toroute calls to trunk group 1. Numbering Format is applied to CLI and is used to set TDMsignalling parameters such as type of number and numbering plan indicator. This doesn’t havethe same significance in SIP calls and during testing it was set to unk-unk.change route-pattern 1Page1 ofPattern Number: 1Pattern Name: Session ManagerSCCAN? nSecure SIP? nUsed for SIP stations? nGrp FRL NPA Pfx Hop Toll No. InsertedNoMrk Lmt List Del DigitsDgts1: 102:3:4:5:6:1:2:3:4:5:6:BCC VALUE TSC CA-TSC0 1 2 M 4 WRequesty y y y y n ny y y y y n ny y y y y n ny y y y y n ny y y y y n ny y y y y n nBG; Reviewed:RRR rITC BCIE Service/Feature PARM Sub Numbering LARDgts onerestnoneSolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.14 of 59Colt CM70 SM

5.9. Administer Incoming Digit TranslationThis step configures the settings necessary to map incoming DDI calls to CommunicationManager extensions. The incoming digits sent in the INVITE message from Colt can bemanipulated as necessary to route calls to the desired extension. During test, the incoming DDInumbers were changed in the Session Manager to Communication Manager Extension numberusing an Adaptation as described in Section 6.4. When done this way, there is no requirement forany incoming digit translation in Communication Manager. If incoming digit translation isrequired, use the change inc-call-handling-trmt trunk-group x command where x is the TrunkGroup defined in Section 5.6.change inc-call-handling-trmt trunk-group 1INCOMING CALL HANDLING TREATMENTService/NumberNumberDel InsertFeatureLenDigitspublic-ntwrkPage1 of3Note: One reason for configuring the enterprise in this way is to allow the use of the extensionnumber as a common identifier with other network elements within the enterprise such as voicemail.5.10. EC500 ConfigurationWhen EC500 is enabled on a Communication Manager station, a call to that station will generatea new outbound call from Communication Manager to the configured EC500 destination,typically a mobile phone. The following screen shows an example EC500 configuration for theuser with station extension 2396. Use the command change off-pbx-telephone station-mappingx where x is Communication Manager station. The Station Extension field will automatically populate with station extension. For Application enter EC500. Enter a Dial Prefix (e.g., 9) if required by the routing configuration. For the Phone Number enter the phone that will also be called (e.g. 0035389434nnnn). Set the Trunk Selection to 1 so that Trunk Group 1 will be used for routing. Set the Config Set to 1.change off-pbx-telephone station-mapping 2391STATIONS WITH OFF-PBX TELEPHONE INTEGRATIONStationExtension2391Application DialCCPrefixEC500-Phone Number0035389434nnnnTrunkSelectionarsPage1 ofConfigSet13DualModeNote: The phone number shown is for a mobile phone in the Avaya Lab. To use facilities forcalls coming in from EC500 mobile phones, the number received in Communication Managermust exactly match the number specified in the above table.Save Communication Manager configuration by entering save translation.BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.15 of 59Colt CM70 SM

6. Configuring Avaya Aura Session ManagerThis section provides the procedures for configuring Session Manager. The Session Manager isconfigured by opening a web browser to the System Manager. The procedures include thefollowing areas: Log in to Avaya Aura System Manager Administer SIP domain Administer Locations Administer Adaptations Administer SIP Entities Administer Entity Links Administer Routing Policies Administer Dial Patterns Administer Application for Avaya Aura Communication Manager Administer Application Sequence for Avaya Aura Communication Manager Administer SIP Extensions6.1. Log in to Avaya Aura System ManagerAccess the System Manager using a web browser and entering http:// FQDN /SMGR, where FQDN is the fully qualified domain name of System Manager. Log in using appropriatecredentials (not shown) and the Home tab will be presented with menu options shown below.BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.16 of 59Colt CM70 SM

6.2. Administer SIP DomainTo add the SIP domain that will be used with Session Manager, select Routing from the Hometab menu and in the resulting tab select Domains from left hand menu. Click the New button tocreate a new SIP domain entry. In the Name field enter the domain name of the enterprise site ora name agreed with Colt; this will be the same as specified in the Authoritative Domain specifiedin the IP Network Region on Communication Manager. Refer to Section 5.3 for details. In test,avaya.com was used. Optionally, a description for the domain can be entered in the Notes field(not shown). Click Commit to save changes.Note: If the existing domain name used in the enterprise equipment does not match that used inthe network, a Session Manager Adaptation can be used to change it (see Section 6.4).BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes 2015 Avaya Inc. All Rights Reserved.17 of 59Colt CM70 SM

6.3. Administer LocationsLocations can be used to identify logical and/or physical locations where SIP Entities reside forthe purposes of bandwidth management. One location is added to the sample configuration forall of the enterprise SIP entities. On the Routing tab select Locations from the left hand menu(not shown). Under General, in the Name field, enter an informative name for the location.Scroll to the bottom of the page and under Location Pattern, click Add, then enter an IPAddress Pattern in the resulting new row, * is used to specify any number of allowed charactersat the end of the string. Below is the location configuration used for the test enterprise.BG; Reviewed:RRR m/d/ySolution & Interoperability Test Lab Application Notes

Avaya Avaya Aura Session Manager 7.0.0.0.700007 Avaya Aura System Manager 7.0.0.0.16266 Avaya Aura Communication Manager 7.0-441 Build 0.22477 Avaya Session Border Controller for Enterprise 7.-21-6602 Avaya G430 Media Gateway 37.19. Avaya 96x0 Phone (SIP) 2_6_14_5 Avaya 9608 Phone (SIP) 7.0.0 R39 Avaya 96x0 Phone (H.323) 3.230A

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