Transcription

Avaya Solution & Interoperability Test LabApplication Notes for Configuring OneStream SIP Trunkingwith Avaya IP Office Release 8.1 – Issue 0.1AbstractThese Application Notes describes the steps to configure Session Initiation Protocol (SIP)Trunking between OneStream Networks and Avaya IP Office Release 8.1.OneStream SIP Trunking provides PSTN access via a SIP trunk between the enterprise and theOneStream network as an alternative to legacy analog or digital trunks. This approachgenerally results in lower cost for the enterprise.Information in these Application Notes has been obtained through DevConnect compliancetesting and additional technical discussions. Testing was conducted in the Avaya Solutions andInteroperability Test Lab, utilizing OneStream SIP Trunk Services.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.1 of 41OneStream-IPO81

1. IntroductionThese Application Notes describe the steps required to configure Session Initiation Protocol(SIP) Trunking between OneStream and Avaya IP Office Release 8.1.The OneStream SIP Trunking service referenced within these Application Notes is positioned forcustomers that have an IP-PBX or IP-based network equipment with SIP functionality, but needa form of IP transport and local services to complete their solution.OneStream SIP Trunking will enable delivery of origination and termination of local, longdistance and toll-free traffic across a single broadband connection. A SIP signaling interface willbe enabled to the Customer Premises Equipment (CPE).2. General Test Approach and Test ResultsThe general test approach was to connect a simulated enterprise site to the OneStream SIPTrunking service via the public Internet and exercise the features and functionality listed inSection 2.1. The simulated enterprise site was comprised of Avaya IP Office and various Avayaendpoints.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 TestingTo verify SIP trunking interoperability, the following features and functionality were coveredduring the interoperability compliance test: Response to SIP OPTIONS queriesIncoming PSTN calls to various phone types including H.323, SIP, digital, and analogtelephones at the enterprise. All inbound PSTN calls were routed to the enterprise acrossthe SIP trunk from the service providerOutgoing PSTN calls to various phone types including H.323, SIP, digital, and analogtelephones at the enterprise. All outgoing PSTN calls were routed to the enterprise acrossthe SIP trunk from the service providerInbound and outbound PSTN calls to/from Avaya IP Office SoftphoneVarious call types including: local, long distance, outbound toll-free, and local directoryassistanceCodecs G.729A and G.711MUT.38 FaxALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.2 of 41OneStream-IPO81

Caller ID presentation and Caller ID restrictionDTMF transmission using RFC 2833Voicemail navigation using DTMF for inbound and outbound callsUser features such as hold and resume, transfer, and conferenceOff-net call forwarding and twinning2.2. Test ResultsInteroperability testing of OneStream SIP Trunking was completed with successful results for alltest cases with the exception of the observations/limitations described below. Operator and Operator Assisted calling – Not supported at the time of writing theseApplication Notes.Faxing – While T.38 faxing did work most of the time, intermittent failures wereobserved. For this reason it is recommended to use G.711 for the Fax Transport Supportsetting as described in Section 5.7.4.Call Transfers – When attempting to Transfer an inbound call back out to the PSTN(off-net transfer), OneStream responds to the SIP REFER with a “202 Accepted” and thecall transfer completes. However, directly after that a “481 Transaction / Dialog DoesNot Exist” is issued via a SIP NOTIFY message, and while the Transfer does work, ithangs on the system and uses system resources for the duration of the call.When attempting to Transfer an outbound call back out to the PSTN (off-net transfer),OneStream responds to the SIP REFER with a “406 Not Acceptable”. However, whilethe call transfers do work for the SIP endpoints in the same way as described above, theH.323 endpoints fail completely with no audio.Due to these limitations with SIP REFER it is recommended to disable the use of REFERas described in Section 5.7.1.2.3. SupportFor technical support on OneStream SIP Trunking, contact OneStream using the CustomerService links at www.OneStream.com.Avaya customers may obtain documentation and support for Avaya products by visitinghttp://support.avaya.com. In the United States, (866) GO-AVAYA (866-462-8292) providesaccess to overall sales and service support menus.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.3 of 41OneStream-IPO81

3. Reference ConfigurationFigure 1 illustrates the sample configuration used for the DevConnect compliance testing. Thesample configuration shows an enterprise site connected to OneStream SIP Trunking.Located at the enterprise site is an Avaya IP Office 500 V2. The LAN port of Avaya IP Office isconnected to the enterprise LAN while the WAN port is connected to the public network.Endpoints include an Avaya 1616 IP Telephone (with H.323 firmware), an Avaya 1140E IPTelephone (with SIP firmware), an Avaya 1408 Digital Telephone, an Avaya Analog Telephoneand an Avaya IP Office Softphone,. The site also has a Windows 2003 Server running AvayaVoicemail Pro for voicemail and the Avaya IP Office Manager to configure the Avaya IP Office.Figure 1: Avaya Interoperability Test Lab ConfigurationFor security purposes, any public IP addresses or PSTN routable phone numbers used in thecompliance test are not shown in these Application Notes. Instead, public IP addresses have beenreplaced with private addresses and all phone numbers have been replaced with numbers thatcannot be routed.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.4 of 41OneStream-IPO81

In an actual customer configuration, the enterprise site may also include additional networkcomponents between the service provider and the Avaya IP Office such as a session bordercontroller or data firewall. A complete discussion of the configuration of these devices is beyondthe scope of these Application Notes. However, it should be noted that SIP and RTP trafficbetween the service provider and the Avaya IP Office must be allowed to pass through thesedevices.4. Equipment and Software ValidatedThe following equipment and software were used for the sample configuration provided:Avaya IP Telephony Solution ComponentsEquipmentSoftwareAvaya IP Office 500 V2Release 8.1 (63)Avaya Voicemail ProRelease 8.1 (810)Avaya IP Office ManagerRelease 10.1 (63)Avaya 1616 IP Telephone (H.323)Release 1.301SAvaya 1140E IP Telephone (SIP)Release 04.03.12Avaya 1408 Digital TelephoneRelease 0.39Avaya Analog TelephoneN/AAvaya IP Office SoftphoneRelease 3.2.3.48 (67009)OneStream SIP Trunking Solution ComponentsComponentRelease?5. Configure Avaya IP OfficeAvaya IP Office is configured through the Avaya IP Office Manager PC application. From theAvaya IP Office Manager PC, select Start Programs IP Office Manager to launch theapplication. A screen that includes the following in the center may be displayed:Navigate to File Open Configuration, select the proper Avaya IP Office system from thepop-up window and log in with the appropriate credentials. The appearance of the IP OfficeManager can be customized using the View menu. In the screens presented in this section, theView menu was configured to show the Navigation pane on the left side, the Group pane in theALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.5 of 41OneStream-IPO81

center, and the Details pane on the right side. These panes will be referenced throughout theAvaya IP Office configuration. All licensing and feature configuration that is not directly relatedto the interface with the service provider (such as twinning and IP Office Softphone support) isassumed to already be in place.5.1. Licensing and Physical HardwareThe configuration and features described in these Application Notes require the IP Office systemto be licensed appropriately. If a desired feature is not enabled or there is insufficient capacity,contact an authorized Avaya sales representative.To verify that there is a SIP Trunk Channels License with sufficient capacity; click License inthe Navigation pane and SIP Trunk Channels in the Group pane. Confirm there is a validlicense with sufficient Instances (trunk channels) in the Details pane.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.6 of 41OneStream-IPO81

If Avaya IP Telephones will be used, as is the case in these Application Notes, verify the AvayaIP endpoints license. Click License in the Navigation pane and Avaya IP endpoints in theGroup pane. Confirm there is a valid license with sufficient Instances in the Details pane.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.7 of 41OneStream-IPO81

In the sample configuration, looking at the IP Office 500v2 from left to right, the first module isa Combination Card. This module has 6 Digital Stations ports, 2 Analog Station ports, 4 AnalogTrunk ports and 10 VCM channels. The VCM32 module is a Voice Compression Modulesupporting VoIP codecs. An IP Office hardware configuration with a VCM component isnecessary to support SIP trunking. The third module is a TCM 8. This module supports BCM /Norstar T-Series and M-Series telephones.The following screen shows the modules populated in the IP Office system used in the sampleconfiguration. To access such a screen, select Control Unit in the Navigation pane. The modulesappear in the Group pane. In the screen below, IP 500 V2 is selected in the Group pane,revealing additional information about the IP 500 V2 in the Details pane.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.8 of 41OneStream-IPO81

5.2. LAN2 SettingsIn the sample configuration the WAN port was used to connect the Avaya IP Office to the publicnetwork. The LAN2 settings correspond to the WAN port on the Avaya IP Office 500v2. Toaccess the LAN2 settings, first navigate to System in the Navigation Pane and then navigate tothe LAN2 LAN Settings tab in the Details Pane. Set the IP Address field to the IP addressassigned to the Avaya IP Office WAN port. Set the IP Mask field to the mask used on the publicnetwork. All other parameters should be set according to customer requirements.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.9 of 41OneStream-IPO81

On the VoIP tab in the Details Pane, check the SIP Trunks Enable box to enable theconfiguration of SIP trunks. Under RTP Keepalives set the Scope to RTP, the Initial keepalivesto Enabled and the Periodic timeout to 10. Enabling this will prevent the loss of speech path oncalls forwarded across the SIP trunk. These settings instruct Avaya IP Office to send RTPkeepalive packets every 10 seconds from the establishment of the connection. This will startmedia flowing from the far-end endpoint in those cases where the far-end endpoint is waiting toreceive media before it starts to send media of its own. The RTP Port Number Range can becustomized to a specific range of receive ports for the RTP media. Based on this setting, AvayaIP Office would request RTP media be sent to a UDP port in the configurable range for callsusing LAN2. Avaya IP Office can also be configured to mark the Differentiated Services CodePoint (DSCP) in the IP Header with specific values to support Quality of Services policies forboth signaling and media. The DSCP field is the value used for media and the SIG DSCP is thevalue used for signaling. The specific values used for the compliance test are shown in theexample below and are also the default values. All other parameters should be set according tocustomer requirements.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.10 of 41OneStream-IPO81

On the Network Topology tab in the Details Pane, configure the following parameters: Select the Firewall/NAT Type from the pull-down menu that matches the networkconfiguration. No firewall or network address translation (NAT) device was used in thecompliance test as shown in Figure 1, so the parameter was set to Open Internet.Set Binding Refresh Time (seconds) to 60. This value is used to determine thefrequency at which Avaya IP Office will send SIP OPTIONS messages to the serviceprovider.Set Public IP Address to the IP address of the Avaya IP Office WAN port.Set the Public Port to 5060.All other parameters should be set according to customer requirements.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.11 of 41OneStream-IPO81

5.3. Voicemail SettingsOn the Voicemail tab in the Details Pane, configure the SIP Settings section. The values enteredfor the SIP Name and Contact fields are used as the user part of the SIP URI in the From andContact headers for outgoing SIP trunk calls from Voicemail (e.g., Outcalling). The SIP Nameand Contact are set to one of the DID numbers assigned to the enterprise from OneStream. TheSIP Display Name (Alias) parameter can optionally be configured with a descriptive name (e.g.Voicemail). Uncheck the Anonymous box to allow Voicemail Caller ID information to thenetwork.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.12 of 41OneStream-IPO81

5.4. System Telephony SettingsOn the Telephony tab in the Details Pane, uncheck the Inhibit Off-Switch Forward/Transferbox to allow call forwarding and call transfers to the PSTN via the service provider across theSIP trunk. If for security reasons incoming calls should not be allowed to transfer back to thePSTN then leave this setting checked.5.5. Twinning Calling Party SettingsTo view or change Twinning settings, select the Twinning tab as shown in the following screen.The Send original calling party information for Mobile Twinning box is not checked in thesample configuration, and the Calling party information for Mobile Twinning is left blank.This will allow the Send Caller ID setting in Section 5.7.1 to control the calling partyinformation for Mobile Twinning and forwarded calls.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.13 of 41OneStream-IPO81

5.6. IP RouteNavigate to IP Route in the left Navigation Pane, and then right-click on the Group Pane toselect New (not shown). Create a default route with the following parameters: Set IP Address and IP Mask to 0.0.0.0.Set Gateway IP Address to the IP Address of the default router to reach OneStream.Set Destination to LAN2 from the pull-down menu.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.14 of 41OneStream-IPO81

5.7. SIP LineA SIP line is needed to establish the SIP connection between Avaya IP Office and OneStreamSIP Trunking. To create a SIP line, begin by navigating to Line in the Navigation Pane. Rightclick in the Group Pane and select New SIP Line (not shown).5.7.1. SIP Line – SIP Line TabOn the SIP Line tab in the Details Pane, configure the parameters as shown below. Leave ITSP Domain Name field blank.Set Send Caller ID to Diversion Header. With this setting and the related configurationin Section 5.5, IP Office will include the Diversion Header for calls that are directed viaMobile Twinning out the SIP Line to OneStream. It will also include the DiversionHeader for calls that are call forwarded out the SIP Line.Under REFER Support, set the Incoming and Outgoing fields to Never and uncheckRefer Support. See Section 2.2 regarding limitations observed using SIP REFER.Check the In Service box. This makes the trunk available to incoming and outgoing calls.Check the Check OOS box. IP Office will use the SIP OPTIONS method to periodicallycheck the SIP Line. The time between SIP OPTIONS sent by IP Office will use theBinding Refresh Time for LAN2, as shown in Section 5.2.Default values may be used for all other parameters.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.15 of 41OneStream-IPO81

5.7.2. SIP Line - Transport TabSelect the Transport tab. This tab was first introduced in Release 6.1. Some informationconfigured in this tab had been under the SIP Line tab in Release 6.0. Set the parameters asshown below. Set ITSP Proxy Address to the IP address(es) of the OneStream SIP proxy or proxies. Inthe compliance test, two SIP proxies were populated in this field and were prioritized viathe w3 and w2 weight parameters. Outbound calls were placed via round robin to eachproxy by setting up the 9N short code without an IP address in the Telephone Numberfield, as detailed in Section 5.8.Set Layer 4 Protocol to UDP.Set Use Network Topology Info to the network port configured in Section 5.2.Set the Send Port to 5060.Default values may be used for all other parameters.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.16 of 41OneStream-IPO81

5.7.3. SIP Line - SIP URI TabA SIP URI entry must be created to match each incoming number that Avaya IP Office willaccept on this line. Select the SIP URI tab, then click the Add button and the New Channel areawill appear at the bottom of the pane. To edit an existing entry, click an entry in the list at thetop, and click the Edit button. In the example screen below, a previously configured entry isedited. The entry was created with the parameters shown below: Set Local URI, Contact and Display Name to Use Internal Data. This setting allowscalls on this line when the SIP URI matches the number set in the SIP tab of any User asshown in Section 5.9.Associate this line with an incoming line group by entering a line group number in theIncoming Group field. This line group number will be used in defining incoming callroutes for this line. Similarly, associate the line to an outgoing line group using theOutgoing Group field. The outgoing line group number is used in defining short codesfor routing outbound traffic to this line. For the compliance test, a new incoming andoutgoing group 19 was defined that only contains this line (line 19).Set Max Calls per Channel to the number of simultaneous SIP calls that are allowedusing this SIP URI pattern.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.17 of 41OneStream-IPO81

In the sample configuration, the single SIP URI previously created shown below as Channel 1was sufficient to allow incoming calls for OneStream DID numbers destined for specific IPOffice users or IP Office hunt groups. The calls are accepted by IP Office since the incomingnumber will match the SIP Name configured for the user or hunt group that is the destination forthe call. Channels 2 and 3 display service numbers, such as a DID number routed directly tovoicemail or DID used for Mobile Call Control such as FNE00. DID numbers that IP Officeshould admit can be entered into the Local URI and Contact fields instead of Use Internal Data.The numbers 562-555-1081 and 562-555-1084 will be assigned as service numbers in theIncoming Call Routes in Section 5.10. Note that only the Incoming Line Group was populatedfor these service numbers as these numbers are only used for inbound call services.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.18 of 41OneStream-IPO81

5.7.4. SIP Line - VoIP TabSelect the VoIP tab to set the Voice over Internet Protocol parameters of the SIP line. Set theparameters as shown below. Set the Codec Selection field to Custom to allow the specific codec selection to bedifferent from the system default. To modify, click on a codec and use the arrow keys tomove it to the Selected column and to change the order of preference. For the compliancetest, G.729A and G.711MU were used.Uncheck the VoIP Silence Suppression box.Check the Re-invite Supported box.Check the Codec Lockdown box. Since both G.729A and G.711MU codecs are presentin the codec list, it is necessary to enable this setting so that G.711MU is used for faxingrather than G.729A, since G.729A is preferred in the codec list. The Use Offerer’sPreferred Codec field is automatically enabled when the Codec Lockdown box isselected.Set the Fax Transport Support to G.711. See Section 2.2 for additional faxconsiderations.Set the DTMF Support field to RFC2833. This directs Avaya IP Office to send DTMFtones using RTP events messages as defined in RFC2833.Default values may be used for all other parameters.Click the OK button at the bottom of the page (not shown).ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.19 of 41OneStream-IPO81

5.8. Short CodesDefine a short code to route outbound traffic to the SIP line. To create a short code, right-clickon Short Code in the Navigation Pane and select New (not shown). On the Short Code tab inthe Details Pane, configure the parameters as shown below. In the Code field, enter the dial string which will trigger this short code, followed by asemi-colon. In this case, 9N;. This short code will be invoked when the user dials 9followed by any number.Set Feature to Dial. This is the action that the short code will perform.Set Telephone Number to N. This field is used to construct the Request URI and Toheaders in the outgoing SIP INVITE message. The value N represents the number dialedby the user. Since there are two SIP proxies populated on the SIP line for OneStream, theIP address that would normally follow N in this field (e.g. N”@x.x.x.x”) is omitted sothat outbound calls will round-robin between the two OneStream SIP proxies.Set the Line Group Id to the outgoing line group number defined on the SIP URI tab onthe SIP Line in Section 5.7.3. This short code will use this line group when placing theoutbound call.Click the OK button (not shown).ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.20 of 41OneStream-IPO81

The simple 9N; short code previously illustrated does not provide a means of alternate routing ifthe configured OneStream SIP Line is out of service or temporarily not responding. Whenalternate routing options and/or more customized analysis of the digits following the short codeare desired, the Automatic Route Selection (ARS) feature may be used. In the following examplescreen, the short code 8N is illustrated for access to ARS. When the Avaya IP Office user dials 8plus any number N, rather than being directed to a specific Line Group Id, the call is directed to50: Main, configurable via ARS. See Section 5.11 for example ARS route configuration for 50:Main as well as a backup route.Optionally, add or edit a short code that can be used to access the SIP Line anonymously. In thescreen shown below, the short code *67N; is illustrated. This short code is similar to the 9N;short code except that the Telephone Number field begins with the letter W, which means“withhold” the outgoing calling line identification. In the case of the SIP Line to OneStreamdocumented in these Application Notes, when a user dials *67 plus any number N, IP Office willinclude the user’s telephone number in the P-Asserted-Identity (PAI) header along with“Privacy: Id”. OneStream will allow the call due to the presence of a valid DID in the PAIheader, but will prevent presentation of the caller id to the called PSTN destination.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.21 of 41OneStream-IPO81

The following screen illustrates a short code that acts like a feature access code rather than ameans to access a SIP Line. In this case, the Code FNE31 is defined for Feature FNE Serviceto Telephone Number 31 (Mobile Call Control). This short code will be used as means to allowa OneStream DID to be programmed to route directly to this feature, via inclusion of this shortcode as the destination of an Incoming Call Route. See Section 5.10. This feature is used toprovide dial tone to twinned mobile devices (e.g., cell phone) directly from IP Office; once dialtone is received the user can perform dialing actions including making calls and activating ShortCodes.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.22 of 41OneStream-IPO81

5.9. UserConfigure the SIP parameters for each user that will be placing and receiving calls via the SIPline defined in Section 5.7. To configure these settings, first navigate to User in the NavigationPane, and then click on the user in the Group Pane to be modified. Select the SIP tab in theDetails Pane. The values entered for the SIP Name and Contact fields are used as the user partof the SIP URI in the From and Contact headers for outgoing SIP trunk calls and allow matchingof the SIP URI for incoming calls without having to enter this number as an explicit SIP URI forthe SIP line. See Section 5.7.3. The example below shows the settings for User 301. The SIPName and Contact are set to one of the DID numbers assigned to the enterprise fromOneStream. The SIP Display Name (Alias) parameter can optionally be configured with adescriptive name. If all calls involving this user and a SIP Line should be considered private,then the Anonymous box may be checked to withhold the user’s information from the network.Click the OK button (not shown).ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.23 of 41OneStream-IPO81

The following screen shows the Mobility tab for User 301. The Mobility Features and MobileTwinning boxes are checked. The Twinned Mobile Number field is configured with thenumber to dial to reach the twinned mobile telephone over the SIP trunk, in this case917205558022. Other options can be set according to customer requirements.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.24 of 41OneStream-IPO81

5.10. Incoming Call RouteAn incoming call route maps an inbound DID number on a specific line to an internal extension.This procedure can be repeated for each DID number provided by the service provider or theprocedure that follows can be used for simplicity. To create an incoming call route, right-clickIncoming Call Routes in the Navigation Pane and select New. On the Standard tab of theDetails Pane, enter the parameters as shown below. Set the Bearer Capacity to Any Voice.Set the Line Group Id to the incoming line group of the SIP line defined in Section5.7.3.Leave the Incoming Number field blank. This, in conjunction with the configuration ofthe Destination tab shown in the step below, will look to match an incoming number onthe respective Line Group ID with the DIDs populated on the User SIP tab as shown inSection 5.9 above.Default values can be used for all other fields.On the Destinations tab, set the Destination field to “.” by typing a period in the field (do notuse the pull down menu). Click the OK button (not shown).ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes 2013 Avaya Inc. All Rights Reserved.25 of 41OneStream-IPO81

Incoming Call Routes for other direct mappings of DID numbers to IP Office users listed inFigure 1 are omitted here, but can be configured in the same fashion.In the screen shown below, the incoming call route for Incoming Number 5625551081 isillustrated. The Line Group Id is 19, matching the Incoming Group field configured in the SIPURI tab in Section 5.7.3.When configuring an Incoming Call Route, the Destination field can be manually configuredwith a number such as a short code, or certain keywords available from the pull-down menu. Forexample, the following Destinations tab for an incoming call route contains the DestinationFNE31 entered manually. FNE31 is the short code for FNE Service, as shown in Section 5.8.An incoming call to 562-555-1081 will be delivered directly to internal dial tone from the IPOffice, allowing the caller to perform dialing actions including making calls and activating ShortCodes. The incoming caller ID must match the Twinned Mobile Number entered in the UserMobility tab (Section 5.9); otherwise the IP Office responds with a 486 Busy Here and the callerwill hear a busy tone.ALW; Reviewed:SPOC mm/dd/2013Solution & Interoperability Test Lab Application Notes

(SIP) Trunking between OneStream and Avaya IP Office Release 8.1. The OneStream SIP Trunking service referenced within these Application Notes is positioned for customers that have an IP-PBX or IP-based network eq