Avaya

Avaya Solution & Interoperability Test Lab
Application Notes for Configuring the Avaya Aura Solution for Midsize Enterprise 6.1 with the AT&T Mobility SIP Trunk Service in Puerto Rico Issue 1.0
Abstract
These Application Notes describe the procedures for configuring Session Initiation Protocol (SIP) Trunking between the service provider AT&T Mobility in Puerto Rico and an Avaya Aura SIP-enabled enterprise solution. The Avaya solution consists of a single server containing the Avaya Aura Solution for Midsize Enterprise 6.1 Template, an Avaya Media Gateway and different types of endpoints. The AT&T Mobility SIP Trunk Service in Puerto Rico provides PSTN access via a SIP trunk between the enterprise and the AT&T network as an alternative to legacy analog or digital trunks. This approach generally results in lower cost for the enterprise. AT&T Mobility in Puerto Rico is a member of the Avaya DevConnect Service Provider program. Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab.
MAA; Reviewed: SPOC 1/19/2012
Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved.
1 of 67 ATT-PR-AASME61
Table of Contents
1. Introduction ............................................................................................................................... 4 2. General Test Approach and Results........................................................................................... 4 2.1. Interoperability Compliance Testing.................................................................................... 4 2.2. Test Results ......................................................................................................................... 5 2.3. Support................................................................................................................................. 6 3. Reference Configuration............................................................................................................. 6 3.1. Midsize Enterprise Solution Components ... 8 4. Equipment and Software Validated ..........................................................................................10 5. Configure Communication Manager.........................................................................................10 5.1. Licensing and Capacity....................................................................................................11 5.2. System Features.................................................................................................................. 12 5.3. IP Node Names................................................................................................................... 13 5.4. Codecs................................................................................................................................. 13 5.5. IP Network Region ............................................................................................................ 14 5.6. Signaling Group.................................................................................................................. 15 5.7. Trunk Group....................................................................................................................... 17 5.8. Calling Party Information................................................................................................... 19 5.9. Inbound Routing................................................................................................................. 20 5.10. Outbound Routing............................................................................................................ 21 6. Configure Avaya Aura Session Manager............................................................................. 24 6.1. System Manager Login and Navigation ............................................................................ 25 6.2. SIP Domains ...................................................................................................................... 26 6.3. Locations ............................................................................................................................27 6.4. Adaptations ........................................................................................................................ 29 6.5. SIP Entities ........................................................................................................................ 30 6.6. Entity Links....................................................................................................................... 34 6.7. Routing Policies ................................................................................................................ 36 6.8. Dial Patterns....................................................................................................................... 37 7. Configure Avaya Aura Session Border Controller ............................................................... 39 7.1. Installation Wizard.............................................................................................................. 39 7.1.1. Network Settings ...........................................................................................................40 7.1.2. Session Border Controller Data .................................................................................... 41 7.2. Post Installation Configuration .......................................................................................... 43 7.2.1. Options Frequency .................................................................................................... 43 7.2.2. Media Ports ... 45 7.2.3. Blocked Headers.46 7.2.4. Diversion Header Domain............................................................................................. 48 7.2.5. Request URI .. 52 7.2.6. Refer-to Header . 53 7.2.7. Save the Configuration . 56 8. AT&T Mobility SIP Trunk Service Configuration ........................................................................ 57 9. Verification and Troubleshooting .............................................................................................. 57
MAA; Reviewed: SPOC 1/19/2012 Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved. 2 of 67 ATT-PR-AASME61
10. Conclusion................................................................................................................................ 58 11. References...................................................................................................................................... 59 Appendix A: Avaya Aura SBC Configuration File ......................................................................... 60
1. Introduction
These Application Notes describe the steps to configure Session Initiation Protocol (SIP) Trunking between the AT&T Mobility SIP Trunk Service in Puerto Rico and the Avaya Aura SIP-enabled enterprise solution. The Avaya solution consists of a HP Proliant DL360 server running the Avaya Aura Solution for Midsize Enterprise Template, release 6.1. Multiple Avaya Aura applications are delivered as part of the Template, running as virtual machines on top of System Platform. An Avaya Media Gateway and various Avaya SIP, H.323, digital and analog endpoints are also part of the solution. The AT&T Mobility SIP Trunk service in Puerto Rico referenced within these Application Notes is designed for enterprise business customers. Customers using this service with the Avaya Aura SIP-enabled enterprise solution should be able to place and receive PSTN calls via a broadband WAN connection and the SIP protocol. This converged network solution is an alternative to traditional PSTN trunks such as analog and/or ISDN-PRI. During the next pages and for brevity in these Application Notes, the service providers name AT&T Mobility in Puerto Rico will be abbreviated and referred as AT&T Mobility or just as AT&T.
2. General Test Approach and Test Results

A simulated enterprise site containing all the equipment for the Avaya Aura Solution for Midsize Enterprise was installed at the Avaya Solution and Interoperability Lab. The enterprise site was configured to connect to the AT&T Mobility SIP Trunk service by means of a broadband connection to the public Internet.
2.1. Interoperability Compliance Testing

To verify SIP trunking interoperability, the following features and functionality were covered during the interoperability compliance test: Incoming PSTN calls to various phone types. Phone types included H.323, SIP, digital, and analog telephones at the enterprise. All inbound calls from PSTN were routed to the enterprise across the SIP trunk from the service provider. Outgoing PSTN calls from various phone types. Phone types included H.323, SIP, digital, and analog telephones at the enterprise. All outbound calls to PSTN were routed from the enterprise across the SIP trunk to the service provider. Inbound and outbound PSTN calls to/from Avaya one-X Communicator (1XC) soft phones Avaya one-X Communicator supports two modes (Road Warrior and Telecommuter). Each supported mode was tested. Avaya one-X Communicator also supports two signaling protocols: H.323 and SIP. Each supported protocol was tested. Various call types, including: local, long distance, international, outbound toll-free, emergency (911) and local directory assistance (411, 611).
MAA; Reviewed: SPOC 1/19/2012 Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved. 4 of 67 ATT-PR-AASME61
Codecs G729A and G.711MU and proper codec negotiation. DTMF tone transmissions passed as out-of-band RTP events as per RFC 2833. Caller ID presentation and Caller ID restriction. Voicemail navigation for inbound and outbound calls. User features such as hold and resume, transfer, and conference. Off-net call forwarding and mobility (extension to cellular). Routing inbound PSTN calls to call center agent queues. Network Call Redirection using SIP REFER for transfer of inbound call back to PSTN.
Items not supported or not tested included the following: Operator services such as dialing 0 or 0 + 10 digits are not supported in this offer by AT&T in Puerto Rico. Inbound toll-free are supported but were not tested as part of the compliance test.
2.2. Test Results

Interoperability testing of the AT&T Mobility SIP Trunk Service with the Avaya Aura SIPenabled enterprise solution was completed with successful results with the exception of the observations and limitations described below: Call Display on PSTN transferred calls: Call display was not properly updated on the PSTN phone to reflect the true connected party on calls that are transferred to the PSTN from the enterprise. After the call transfer was completed, the PSTN phone showed the party that initiated the transfer instead of the actual connected party. Network Call Redirection: When a Communication Manager vector is programmed to redirect an inbound call to a PSTN number before answering the call in the vector, AT&T will send an ACK to the 302 Moved Temporarily SIP message from the enterprise, but it will not redirect the call to the new party in the Contact header of the 302 message. The initiator of the inbound call hears silence. Network call redirection works successfully when the Communication Manager vector is programmed to redirect the inbound call to a PSTN number after answering the call first in the vector (using SIP REFER message for network call redirection instead of the 302 message). Network Call Redirection using REFER with redirected part Busy: In the testing environment, when an inbound call was made to the enterprise, to a vector redirecting the call to another PSTN endpoint that was busy, the caller will hear a busy tone, but AT&T will not return a 486 Busy Here, preventing any additional processing of the call by Communication Manager, like the routing of the call to a local agent on the enterprise. SIP User to User Information: When a Communication Manager vector is programmed to send User-to-User Information (UUI) to a remote party, the information is generated and included in the REFER header sent to AT&T, but the UUI is not passed to the destination SIP endpoint.
T.38 Fax: Even though incoming T.38 fax calls to the Enterprise worked successfully, outbound T.38 fax calls failed to complete. Thus, T.38 Fax should not be used with this solution.
2.3. Support
At the time of writing these Application Notes, software load 6.1.0.0.2580 of the Midsize Enterprise Solution Template, the Session Border Controller functionality is being introduced by Avaya under a Beta trial program. Customers who want to implement SBC functionality with their Midsize Enterprise Solution server, as described in these Application Notes, should contact Scott Larson with the Avaya Global Product Introduction Team at (303)538-2407 to enroll in this program. For technical support on the Avaya products described in these Application Notes visit http://support.avaya.com. For technical support on the AT&T Mobility SIP Trunk Services offer, call the AT&T Mobility Network Operations Center at 787-717-9900.
3. Reference Configuration
Figure 1 illustrates the sample Avaya Aura SIP-enabled enterprise solution connected to the AT&T Mobility SIP Trunk Service through a public Internet WAN connection, which is the configuration used for the Compliance Testing. For security purposes, private addresses are shown in these Application Notes for the Public SBC and the ITSP network interfaces, instead of the real public IP addresses used during the tests. Also PSTN routable phone numbers used in the compliance test have been changed to nonroutable ones.
Figure 1: Avaya Aura SIP Enterprise Solution connecting to AT&T Mobility SIP Trunk Service.
3.1. Midsize Enterprise Solution Components

The Avaya Aura Solution for Midsize Enterprise Release 6.1 Template delivers the following applications as virtual machines running on System Platform 6.0.3: Communication Manager 6.0.1 Communication Manager Messaging 6.0.1 Session Manager 6.1 System Manager 6.1 Presence Services 6.1 Utility Services 6.1 Application Enablement Services 6.1 Session Border Controller 6.0.2
These Application Notes will not cover the software installation of System Platform and the loading of the Midsize Enterprise Template. For more information and step by step instructions on the software installation of System Platform, the Midsize Enterprise Template and initial configuration, see [1]. The screenshot shows the list of the applications installed and running on the server, as seen from the Virtual Machine Management screen in System Platform.
Note that Application Enablement and Presence Services are installed as part of the Midsize Enterprise Template, but since these applications were not used during the compliance testing, the configuration of these services is not covered in these Application Notes.
The other Avaya components used to create the simulated customer site included: Avaya G450 Media Gateway Avaya 9600-Series IP Telephones (H.323 and SIP) Avaya one-X Communicator soft phones (H.323 and SIP) Avaya digital and analog telephones
The public interface of the Avaya Aura SBC is located at the edge of the Enterprise, connecting to the outside network. The SBC private interface connects to the enterprise network. All SIP and RTP traffic entering or leaving the enterprise flows through the SBC. In this way, the SBC can protect the enterprise against any SIP-based attacks. The SBC also provides network address translation at both the IP and SIP layers. The transport protocol between the Avaya Aura SBC and AT&T Mobility across the public IP network is UDP. The transport protocol between the Avaya Aura SBC and the enterprise Session Manager across the enterprise IP network is TCP. A separate trunk was created between Communication Manager and Session Manager to carry the service provider traffic. This was done so that any trunk or codec settings required by the service provider could be applied only to this trunk, without affecting other enterprise SIP traffic. In addition, this trunk carried both inbound and outbound traffic. For inbound calls, the calls flow from the service provider to the SBC, then to Session Manager. Session Manager uses the configured dial patterns (or regular expressions) and routing policies to determine the recipient of the call, in this case the Communication Manager. Once the call arrives at Communication Manager, further incoming call treatment, such as incoming digit translations and class of service restrictions may be performed. Outbound calls to the PSTN were first processed by Communication Manager for outbound feature treatment such as automatic route selection and class of service restrictions. Once Communication Manager selected the proper SIP trunk, the call was routed to Session Manager. The Session Manager once again used the configured dial patterns (or regular expressions) and routing policies to determine the route to the SBC for egress to the AT&T network. Since Puerto Rico is a country member of the North American Numbering Plan (NANP), the user dialed 10 digits for local calls, and 11 (1 + 10) or 10 digits for other calls between the NANP.
4. Equipment and Software Validated

The following equipment and software were used for the sample configuration provided: Component Avaya Avaya Aura Solution for Midsize Enterprise on HP Proliant DL360 G7 Server. Avaya Aura Communication Manager Avaya Aura Communication Manager Messaging Avaya Aura System Manager Avaya Aura Session Manager Avaya Aura Session Border Controller Avaya G450 Media Gateway Avaya 96xx Series IP Telephones (H.323) Avaya 96xx Series IP Telephones (SIP) Avaya 96x1 Series IP Telephones (H.323) Avaya 96x1 Series IP Telephones (SIP) Avaya one-X Communicator (H.323, SIP) Avaya 2420 Digital Telephone Avaya 6210 Analog Telephone AT&T Puerto Rico SIP Trunking
Acme-Packet Net-Net 4250 SBC BroadWorks Soft Switch Nortel CS2K PSTN Gateway
Version 6.1.0.0.2580 (System Platform 6.0.3) R016x.00.1.510.1 vcm-016-00.1.510.1 6.1.0.0.7345-6.1.5.106 6.1.2.0.612004 6.0.2.0.2 31.19.2 Avaya one-X Deskphone Edition 3.1 Avaya one-X Deskphone Edition SIP 2.6.2 Avaya one-X Deskphone Edition 6.0 SP5 Avaya one-X Deskphone Edition SIP 6.0.2 6.1.1.02-SP1-32858 n/a n/a Firmware SC6.1.0 MR-9 GA (Build 938) R17 CVM11
5. Configure Communication Manager.

This section describes the procedure for configuring Communication Manager for the AT&T Mobility SIP Trunk Service. A SIP trunk is established between Communication Manager and Session Manager for use by signaling traffic to and from AT&T. It is assumed the general installation and administration of Communication Manager, Avaya G450 Media Gateway and Session Manager has been previously completed and is not discussed here. The Communication Manager configuration was performed using the System Access Terminal (SAT). Some screens in this section have been abridged and highlighted for brevity and clarity in presentation. Note that the IP addresses and phone numbers shown throughout these Application Notes have been edited so that the actual IP addresses of the network elements and public PSTN numbers are not revealed.
5.1. Licensing and Capacity

Use the display system-parameters customer-options command to verify that the Maximum Administered SIP Trunks value on Page 2 is sufficient to support the desired number of simultaneous SIP calls across all SIP trunks at the enterprise including any trunks to and from the service provider. The example shows that 12000 licenses are available and 263 are in use. The license file installed on the system controls the maximum values for these attributes. If a required feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales representative to add additional capacity.
5.2. System Features

Use the change system-parameters features command to set the Trunk-to-Trunk Transfer field to all to allow incoming calls from the PSTN to be transferred to another PSTN endpoint. If for security reasons, incoming calls should not be allowed to transfer back to the PSTN then leave the field set to none.
On Page 9 verify that a text string has been defined to replace the Calling Party Number (CPN) for restricted or unavailable calls. This text string is entered in the two fields highlighted below. The compliance test used the value of anonymous for both.
5.3. IP Node Names

Use the change node-names ip command to verify that node names have been previously defined for the IP addresses of Communication Manager (procr) and the Session Manager Security module (SM). These node names will be needed for defining the service provider signaling groups in Section 5.6.
5.4. Codecs
Use the change ip-codec-set command to define a list of codecs to use for calls between the enterprise and the service provider. For the compliance test, ip-codec-set 2 was used for this purpose. The AT&T SIP Trunk Service supports codecs G.729A and G.711MU, in this order of preference. Enter G.729A and G.711MU in the Audio Codec column of the table. Default values can be used for all other fields.
Since T.38 fax testing was not reliable, it is recommended to disable T.38 Fax by setting the Fax Mode field to off on Page 2.
5.5. IP Network Region

Create a separate IP network region for the service provider trunk group. This allows for separate codec or quality of service settings to be used (if necessary) for calls between the enterprise and the service provider versus calls within the enterprise or elsewhere. For the compliance test, IP Network Region 2 was chosen for the service provider trunks. Use the change ip-networkregion 2 command to configure region 2 with the following parameters: Set the Authoritative Domain field to match the SIP domain of the enterprise. In this configuration, the domain name is avayalab2.com as assigned to the shared test environment in the Avaya test lab. This domain name appears in the From header of SIP messages originating from this IP region. Note that a Session Manager Adaptation (Section 6.4) is used to convert this shared domain name to the specific domain expected by AT&T. Enter a descriptive name in the Name field. Enable IP-IP Direct Audio (shuffling) to allow audio traffic to be sent directly between IP endpoints without using media resources in the Avaya Media Gateway. Set both Intra-region and Inter-region IP-IP Direct Audio to yes. This is the default setting. Shuffling can be further restricted at the trunk level on the Signaling Group form. Set the Codec Set field to the IP codec set defined in Section 5.4. Default values can be used for all other fields.
On Page 4, define the IP codec set to be used for traffic between region 2 and region 1 (the rest of the enterprise). Enter the desired IP codec set in the codec set column of the row with destination region (dst rgn) 1. Default values may be used for all other fields. The example below shows the settings used for the compliance test. It indicates that codec set 2 will be used for calls between region 2 (the service provider region) and region 1 (the rest of the enterprise).
5.6. Signaling Group

Use the add signaling-group command to create a signaling group between Communication Manager and the Session Manager for use by the service provider trunk. This signaling group is used for inbound and outbound calls between the service provider and the enterprise. For the compliance test, signaling group 2 was used for this purpose and was configured using the parameters highlighted below. Set the Group Type field to sip. Set the IMS Enabled field to n. This specifies the Communication Manager will serve as an Evolution Server for the Session Manager. Set the Transport Method to the recommended default value of tls (Transport Layer Security). Set the Near-end Listen Port and Far-end Listen Port to a valid unused port instead of the default well-known port value. (For TLS, the well-known port value is 5061). This is necessary so the SM can distinguish this trunk from the trunk used for other enterprise SIP traffic. For ease of troubleshooting, the compliance test was conducted with the Transport Method set to tcp and the Near-end Listen Port and Far-end Listen Port set to 5062. (For TCP, the well-known port value is 5060). Set the Peer Detection Enabled field to y. The Peer-Server field will initially be set to Others and cannot be changed via administration. Later, the Peer-Server field will automatically change to SM once Communication Manager detects its peer is a Session Manager. Set the Near-end Node Name to procr. This node name maps to the IP address of the Communication Manager as defined in Section 5.3. Set the Far-end Node Name to SM. This node name maps to the IP address of Session Manager as defined in Section 5.3. Set the Far-end Network Region to the IP network region defined for the service provider in Section 5.5. Set the Far-end Domain to the domain of the enterprise.
Set the DTMF over IP field to rtp-payload. This value enables Communication Manager to send DTMF transmissions using RFC 2833 Set Direct IP-IP Audio Connections to y. This field will enable media shuffling on the SIP trunk allowing Communication Manager to redirect media traffic directly between the SIP trunk and the enterprise endpoint. If this value is set to n, then the Avaya Media Gateway will remain in the media path of all calls between the SIP trunk and the endpoint. Depending on the number of media resources available in the Avaya Media Gateway, these resources may be depleted during high call volume preventing additional calls from completing. Default values may be used for all other fields.
5.7. Trunk Group

Use the add trunk-group command to create a trunk group for the signaling group created in Section 5.6. For the compliance test, trunk group 2 was configured using the parameters highlighted below. Set the Group Type field to sip. Enter a descriptive name for the Group Name. Enter an available trunk access code (TAC) that is consistent with the existing dial plan in the TAC field. Set the Service Type field to public-ntwrk. Set the Signaling Group to the signaling group shown in the previous step. Set the Number of Members field to the number of trunk members in the SIP trunk group. This value determines how many simultaneous SIP calls can be supported by this trunk. Default values were used for all other fields.
On Page 2, verify that the Preferred Minimum Session Refresh Interval is set to a value acceptable to the service provider. This value defines the interval that re-INVITEs must be sent to keep the active session alive. For the compliance test, the default value of 600 seconds was used.
On Page 3, set the Numbering Format field to private. This field specifies the format of the calling party number (CPN) sent to the far-end. Beginning with Communication Manager 6.0, public numbers are automatically preceded with a + sign when passed in the SIP From, Contact and P-Asserted Identity headers. The addition of the + sign impacted interoperability with AT&T Mobility. Thus, the Numbering Format was set to private and the Numbering Format in the route pattern was set to unk-unk (see Section 5.10).
Set the Replace Restricted Numbers and Replace Unavailable Numbers fields to y. This will allow the CPN displayed on local endpoints to be replaced with the value set in Section 5.2, if the inbound call enabled CPN block. On Page 4, set the Network Call Redirection field to y. This enables the use of the SIP REFER method for calls transferred back to the PSTN. Set the Send Diversion Header field to y. This is needed to support call forwarding of inbound calls back to the PSTN and some Extension to Cellular (EC500) call scenarios. Set the Support Request History field to n. Set the Telephone Event Payload Type to 101, and Convert 180 to 183 for Early Media to y, the values preferred by AT&T. Default values were used for all other fields.
5.8. Calling Party Information

The calling party number is sent in the SIP From, Contact and PAI headers. Since private numbering was selected to define the format of this number (Section 5.7), use the change private-numbering command to create an entry for each extension which has a DID assigned. The DID numbers are provided by the SIP service provider. Each DID number is assigned in this table to one enterprise internal extension or Vector Directory Numbers (VDNs), and they are used to authenticate the caller with the Service Provider. In the sample configuration, 5 DID numbers were assigned for testing. These 5 numbers were mapped to 5 extensions, 55001 to 55005. These same 10-digit numbers were used in the outbound calling party information on the service provider trunk when calls were originated from these 5 extensions.
In a real customer environment, normally the DID number is comprised of the local extension plus a prefix. If this is true, then a single private numbering entry can be applied for all extensions. In the example below, all stations with a 5-digit extension length, beginning with 5, will send the calling party number as the Private Prefix plus the extension number.
5.9. Inbound Routing

In general, the incoming call handling treatment form for a trunk group can be used to manipulate the digits received for an incoming call if necessary. Since Session Manager is present, Session Manager can be used to perform digit conversion using an Adaptation, and digit manipulation via the Communication Manager incoming call handling table may not be necessary. If the DID number sent by AT&T is unchanged by Session Manager, then the DID number can be mapped to an extension using the incoming call handling treatment of the receiving trunk group. Use the change inc-call-handling-trmt command to create an entry for each DID.
In a real customer environment, where the DID number is normally comprised of the local extension plus a prefix, a single entry can be applied for all extensions, like in the example below.
5.10. Outbound Routing

In these Application Notes, the Automatic Route Selection (ARS) feature is used to route outbound calls via the SIP trunk to the service provider. In the sample configuration, the single digit 9 is used as the ARS access code. Enterprise callers will dial 9 to reach an outside line. This common configuration is illustrated below with little elaboration. Use the change dialplan analysis command to define a dialed string beginning with 9 of length 1, as a feature access code (fac).
Use the change feature-access-codes command to configure 9 as the Auto Route Selection (ARS) Access Code 1.
Use the change ars analysis command to configure the routing of dialed digits following the first digit 9. The example below shows a subset of the dialed strings tested as part of the compliance test. See Section 1.1 for the complete list of call types tested. All dialed strings are mapped to route pattern 2 which contains the SIP trunk group to the service provider.
The route pattern defines which trunk group will be used for the call and performs any necessary digit manipulation. Use the change route-pattern command to configure the parameters for the service provider trunk route pattern in the following manner. The example below shows the values used for route pattern 2 for the compliance test. Pattern Name: Enter a descriptive name. Grp No: Enter the outbound trunk group for the SIP service provider. For the compliance test, trunk group 2 was used. FRL: Set the Facility Restriction Level (FRL) field to a level that allows access to this trunk for all users that require it. The value of 0 is the least restrictive level. Pfx Mrk: 1 The prefix mark (Pfx Mrk) of one will prefix any FNPA 10-digit number with a 1 and leave numbers of any other length unchanged. This will ensure 1 + 10 digits are sent to the service provider for the long distance North American Numbering Plan (NANP) numbers. All HNPA 10 digit numbers are left unchanged. Numbering Format: unk-unk All calls using this route pattern will use the private numbering table. See setting of the Numbering Format in the trunk group form for full details in Section 5.7. LAR: none. If desired, one or more alternate Communication Manager trunks can be listed in the route pattern so that the Look-Ahead Routing (LAR) next setting can route-advance to attempt to complete the call using alternate trunks, should there be no response or an error response is received from the far-end. For the compliance test, since only one trunk group was used, the default value none was selected.
6. Configure Avaya Aura Session Manager

This section provides the procedures for configuring Session Manager. The procedures include adding the following items: SIP domain Logical/physical Location that can be occupied by SIP Entities Adaptation module to perform URI manipulations. SIP Entities corresponding to Communication Manager, the SBC and Session Manager Entity Links, which define the SIP trunk parameters used by Session Manager when routing calls to/from SIP Entities Routing Policies, which control call routing between the SIP Entities Dial Patterns, which govern to which SIP Entity a call is routed
It may not be necessary to create all the items above when creating a connection to the service provider, since some of them would have already been defined as part of the initial Midsize Enterprise Solution template installation. This includes entries such as certain SIP domains, locations, SIP entities, and Session Manager itself. However, each item should be reviewed to verify the configuration.
6.1. System Manager Login and Navigation

Session Manager configuration is accomplished by accessing the browser-based GUI of System Manager, using the URL https://<ip-address>/SMGR, where <ip-address> is the IP address of System Manager. Log in with the appropriate credentials and click on Log On (not shown). The screen shown below is then displayed.
Most of the configuration items are performed in the Routing Element. Click on Routing in the Elements column to bring up the Introduction to Network Routing Policy screen.
6.2. SIP Domains

Create a SIP domain for each domain for which Session Manager will need to be aware in order to route calls. For the compliance test, this includes the enterprise domain, avayalab2.com, and the AT&T domain, aslab.centixvoip.net. The enterprise SIP domain was previously created during the Midsize Enterprise template installation, and this entry was already populated. To add the service provider domain, Navigate to Routing Domains in the left-hand navigation pane (Section 6.1) and click the New button in the right pane (not shown). In the new right pane that appears (shown below), fill in the following: Name: Type: Notes: Enter the domain name. Select sip from the pull-down menu. Add a brief description (optional).
Click Commit. The screen below shows the entry for the enterprise domain
The screen below shows the entry for the AT&T test domain.
6.3. Locations
Locations can be used to identify logical and/or physical locations where SIP Entities reside for purposes of bandwidth management and call admission control. To add a location, navigate to Routing Locations in the left-hand navigation pane and click the New button in the right pane (not shown). In the General section, enter the following values. Use default values for all remaining fields: Name: Enter a descriptive name for the location. Notes: Add a brief description (optional). In the Location Pattern section, click Add and enter the following values. Use default values for all remaining fields: IP Address Pattern: An IP address pattern used to identify the location. Notes: Add a brief description (optional). The screen below shows the location Miami, created during the template installation. This location includes all equipment on the Enterprise subnet, 10.5.5.0. Click Commit to save any changes made, if any.
Note that call bandwidth management parameters should be set per customer requirements. Repeat the preceding procedure to create a separate Location for the AT&T SIP Trunk. Displayed below is the screen for addition of the AT&T Puerto Rico Location, which specifies the inside IP address for the AA-SBC. Click Commit to save.
6.4. Adaptations
Session Manager can be configured with Adaptation modules that modify SIP messages before or after routing decisions have been made. A generic module DigitConversionAdapter supports digit conversion of telephone numbers and specific headers of SIP messages. Other Adaptation modules are built on this generic module, and can modify other headers to permit interoperability with third party SIP products. For the compliance test, the adaptation AT&T In/Out was created. It will be assigned to the SIP Entity for the Avaya Aura SBC, later in this document. This adaptation uses the DigitConversionAdapter generic module and specifies the following two parameters:
iosrcd=avayalab2.com. This parameter replaces the domain of the PAI header on inbound requests with the value of the enterprise domain, avayalab2.com. This parameter must match the value used for the Far-end Domain setting on the Communication Manager signaling group form in Section 5.6. osrcd=aslab.centixvoip.net. This parameter enables the outbound source domain to be overwritten with aslab.centixvoip.net. For outbound PSTN calls from the enterprise to AT&T, the domain portion of the PAI header of outgoing requests will now contain aslab.centixvoip.net, as expected by AT&T.
The screen below shows the adaptation AT&T In/Out created for the compliance test. All other fields were left with their default values.
6.5. SIP Entities

A SIP Entity must be added for Session Manager and for each SIP telephony system connected to it, which includes Communication Manager and the SBC. Navigate to Routing SIP Entities in the left navigation pane and click on the New button in the right pane (not shown). In the General section, enter the following values. Use default values for all remaining fields: Name: Enter a descriptive name. FQDN or IP Address: Enter the FQDN or IP address of the SIP Entity that is used for SIP signaling. Type: Select Session Manager for Session Manager, CM for Communication Manager, Other for the SBC. Adaptation: This field is only present if Type is not set to Session Manager If applicable, select the Adaptation name created in Section 6.4 that will be applied to this entity. Location: Select one of the locations defined previously. Time Zone: Select the time zone for the location above.
The following screen shows the Session Manager SIP Entity, created during the template installation. The FQDN or IP Address is the address of the Session Manager signaling interface (virtual SM-100).
To define the ports used by Session Manager, scroll down to the Port section of the SIP Entity Details screen. This section is only present for Session Manager SIP entities. In the Port section, click Add and enter the following values. Use default values for all remaining fields: Port: Protocol: Default Domain: Port number on which the Session Manager can listen for SIP requests. Transport protocol to be used to send SIP requests. The domain used for the enterprise.
Defaults can be used for the remaining fields. Click Commit to save.
The following screen shows the addition of the SIP Entity for Communication Manager. In order for Session Manager to send SIP service provider traffic on a separate entity link to Communication Manager, the creation of a separate SIP entity for Communication Manager is required. This SIP Entity should be different to the one created during the Template installation for use with all other enterprise SIP traffic. The FQDN or IP Address field is set to the IP address of the procr interface in Communication Manager.
The following screen shows the addition of the Avaya Aura SBC Entity. The FQDN or IP Address field is set to the IP address of the SBC private network interface (see Figure 1). The location is set to the one defined for SBC in Section 6.3. For Adaptation field, select the adaptation module AT&T In/Out previously defined in Section 6.4.
6.6. Entity Links

A SIP trunk between Session Manager and a telephony system is described by an Entity Link. Two Entity Links were created; one to the Communication Manager for use only by service provider traffic and one to the SBC. To add an Entity Link, navigate to Routing Entity Links in the left navigation pane and click on the New button in the right pane (not shown). Fill in the following fields in the new row that is displayed: Name: SIP Entity 1: Protocol: Port: Enter a descriptive name. Select the Session Manager from the drop-down menu. Select the transport protocol used for this link. Port number on which Session Manager will receive SIP requests from the far-end. Select the name of the other system from the drop-down menu. Port number on which the other system receives SIP requests from the Session Manager. Check this box. Note: If this box is not checked, calls from the associated SIP Entity specified in Section 6.5 will be denied.
SIP Entity 2: Port: Trusted: Click Commit to save.
The following screens illustrate the Entity Links to Communication Manager and the SBC. It should be noted that in a customer environment the Entity Link to Communication Manager would normally use TLS. For the compliance test, TCP was used to facilitate troubleshooting since the signaling traffic would not be encrypted. The protocol and ports defined here must match the values used on the Communication Manager signaling group form in Section 5.6. Entity Link to Communication Manager
Entity Link to the SBC:
The following screen shows the complete list of Entity Links. Note that only the highlighted links were created for the compliance test, and are the ones relevant to these Application Notes. Other links appearing on this screen were automatically created at the time of the Midsize Enterprise template installation.
6.7. Routing Policies

Routing policies describe the conditions under which calls will be routed to the SIP Entities specified in Section 6.5. Two routing policies must be added: one for Communication Manager and one for the SBC. To add a routing policy, navigate to Routing Routing Policies in the left navigation pane and click on the New button in the right pane (not shown). The following screen is displayed. In the General section, enter the following values: Name: Notes: Enter a descriptive name. Add a brief description (optional).
In the SIP Entity as Destination section, click Select. The SIP Entity List page opens (not shown). Select the appropriate SIP entity to which this routing policy applies and click Select. The selected SIP Entity displays on the Routing Policy Details page as shown below. Use default values for remaining fields. Click Commit to save. The following screens show the Routing Policies for Communication Manager and the SBC.
6.8. Dial Patterns

Dial Patterns are needed to route specific calls through Session Manager. For the compliance test, dial patterns were needed to route calls from Communication Manager to AT&T and vice versa. Dial Patterns define which route policy will be selected for a particular call based on the dialed digits, destination domain and originating location. To add a dial pattern, navigate to Routing Dial Patterns in the left navigation pane and click on the New button in the right pane (not shown). Fill in the following, as shown in the screens below: In the General section, enter the following values: Pattern: Enter a dial string that will be matched against the Request-URI of the call. Min: Enter a minimum length used in the match criteria. Max: Enter a maximum length used in the match criteria. SIP Domain: Enter the destination domain used in the match criteria. Notes: Add a brief description (optional). In the Originating Locations and Routing Policies section, click Add. From the Originating Locations and Routing Policy List that appears (not shown), select the appropriate originating location for use in the match criteria. Lastly, select the routing policy from the list that will be used to route all calls that match the specified criteria. Click Select. Default values can be used for the remaining fields. Click Commit to save. Two examples of the dial patterns used for the compliance test are shown below. The first example shows that 11 digit dialed numbers that begin with 1 uses route policy Outgoing to AT&T.
The second example shows that a 10 digit number starting with 787111, to domain avayalab2.com and originating from the AT&T Puerto Rico location, will use route policy Incoming to CM Trunk 2. This number falls in the DID range assigned to the enterprise by AT&T. AT&T Puerto Rico is selected for the Originating Location because these calls come from the SBC, which resides in that location.
7. Configure Avaya Aura Session Border Controller

This section describes the configuration of the Avaya Aura SBC. This configuration is done in two parts: The first part is done during the Avaya Aura Solution for Midsize Enterprise template software installation via the installation wizard, which is invoked during the loading of the template. The second part of the configuration is done after the installation is complete using the SBC web interface. The resulting SBC configuration file is shown in Appendix A.
7.1. Installation Wizard

During the installation of the Midsize Enterprise template, the installation wizard will prompt the installer for information that will be used to create the initial configuration of the SBC. These Application Notes will not cover the use of the installation wizard in its entirety, but will include screens that are presented as part of the wizard related to the SBC configuration. For a complete reference in the use of the Midsize Enterprise installation wizard, see [1].
7.1.1. Network Settings

The first screen of the installation wizard is the Network Settings screen. Here is where IP Addresses, Hostnames and Domains are assigned to all Virtual Machines. For the Session Border Controller, fill in the fields as described below: IP Address: Hostname: Domain: Enter the IP address of the SBC management interface (Eth0). Enter a host name for the SBC. Enter the enterprise network domain. Note that this could be different, like in the test scenario, than the enterprise SIP domain.
7.1.2. Session Border Controller Data

On the SBC/Session Border Controller Data screen, fill in the fields as described below and shown in the following screen: In the Configure section check Yes for Do you wish to configure Session Border Controller? In the SIP Service Provider Data section: Service Provider: From the pull-down menu, select the name of the service provider to which the SBC will connect. This will allow the wizard to select a configuration file customized for this service provider. At the time of the compliance test, a customized configuration file did not exist for AT&T Mobility in Puerto Rico. Thus, Generic was chosen instead and further customization was done manually after the wizard was complete. Port: Enter the port number that the service provider uses to listen for SIP traffic. SIP Proxy Enter the AT&T provided IP address of the service provider SIP IP Address1: Proxy. If the service provider has multiple proxies, enter the primary proxy on this screen and additional proxies can be added after installation. Signaling/Media Enter the AT&T provided subnet where signaling/media traffic Network1: will originate. If signaling/media traffic can originate from multiple networks, enter one network address on this screen and additional networks can be added after installation. Signaling/Media: Enter the netmask corresponding to Signaling/Media Network1. Netmask1:
In the SBC Network Data section: Private IP Address: Enter the IP address of the private side of the SBC (Eth5). Private Net Mask: Enter the netmask associated with the private network to which the SBC connects. Private Gateway: Enter the default gateway of the private network. Public IP Address: Enter the IP address of the public side of the SBC (Eth4). Public Net Mask: Enter the netmask associated with the public network to which the SBC connects. Public Gateway: Enter the default gateway of the public network. In the Enterprise SIP Server section: IP Address1: Enter the IP address of the Enterprise SIP Server to which the SBC will connect. In the case of the compliance test, this is the IP address of the Session Manager SIP signaling interface. Transport1: From the pull-down menu, select the transport protocol to be used for SIP traffic between the SBC and Session Manager.
Note: Physical interface Eth5 of the Midsize Enterprise server maps to logical interface Eth1 of the SBC. Similarly, physical interface Eth4 of the server maps to logical interface Eth2 of the SBC.
7.2. Post Installation Configuration

The installation wizard configures the Session Border Controller for use with the service provider chosen in Section 7.1.2. Since a Generic provider was selected in the installation wizard, additional manual changes must also be performed. These changes are performed by accessing the browser-based GUI of the Session Border Controller, using the URL https://<ipaddress>, where <ip-address> is the management IP address configured in Section 7.1.1. Log in with the proper credentials.
7.2.1. Options Frequency

To set the frequency of the OPTIONS messages sent from the SBC to the service provider, first navigate to vsp enterprise servers sig-gateway Telco. Click Show Advanced.
Scroll down to the Routing section of the form. Enter the desired interval in the ping-interval field. For the compliance test, 300 seconds was used. Click Set at the top of the form (shown in previous screen).
7.2.2. Media Ports

To adjust the port range assigned to media streams leaving the outside interface of the SBC, to match the range specified by AT&T for the compliance test of 50000 to 54999, navigate to cluster box interface eth2 ip outside. On the right pane, navigate to media-ports and click Configure.
On the next screen, set the value for base-port to 50000, and the count to 4999. Click Set to complete the configuration.
7.2.3. Blocked Headers

The P-Location and Alert-Info headers are sent in SIP messages from the Session Manager to the AT&T network. These headers contain private IP addresses and SIP Domains from the enterprise, which should not be propagated outside of the enterprise boundaries. These headers were simply removed (blocked) from both requests and responses for outbound calls. To create a rule for blocking a header on an outbound call, first navigate to vsp session-config-pool entry ToTelco header-settings. Click Edit blocked-header
In the right pane that appears, click Add. In the blank fields, enter the name of the header to be blocked. After all the blocked headers are added, click OK. The screen below shows the P-Location and the Alert-Info headers configured to be blocked for the compliance test.
The list of blocked headers for outbound calls will appear in the right pane as shown below. Click Set to complete the configuration.
7.2.4. Diversion Header Domain

Avaya Aura Session Manager can adapt the domain in various SIP headers such as the Request- URI, History-Info and P-Asserted-Identity. As described in these Application Notes, Session Manager was used to adapt the domain in the PAI headers of both incoming and outbound requests, allowing the interoperability between the enterprise domain avayalab2.com and the service providers aslab.centixvoip.net. In the sample configuration, the domain portion of the Diversion header was not altered by the Session Manager adaptations. To allow diverted calls to be processed properly, the SBC was used to convert the domain in the Diversion header from the enterprise domain to the AT&T expected aslab.centixvoip.net. Navigate to vsp session-config-pool entry ToTelco header-settings. The screen below shows the configuration before making changes for the Diversion header. Click Add alteredheader.
In the number field, enter an appropriate unused number. Since this is the first altered-header rule, number 1 was used. For the source-header field, enter Diversion. In the source-field area, enter the following: type: value: Choose selection from the drop-down menu Either enter a value to match directly, or click the regular expression link for assistance in creating the proper value. In the sample configuration, the rule shown will match on avayalab2.com appearing in the Diversion header. Enter the domain to appear in the host portion of the Diversion header, in place of avayalab2.com. For the compliance test, AT&T expected aslab.centixvoip.net.
replacement:
In the destination area, enter Diversion. Select host from the type drop-down menu, since it is the host portion of the Diversion header that the rule should replace. Click the Create button.
Additional configuration can be applied to the altered-header rule using the screen shown below. In the sample configuration, the defaults were retained. Click the Set button.
The following screen shows a summary of the altered-header rule configured in this section. It also shows the blocked-header rule configured in Section 7.2.3.
7.2.5. Request URI

On incoming calls to the enterprise, AT&T will always send the same pilot DID number on the user portion of the request-line of any incoming INVITE, and the actual number dialed in the user portion of the To header. Since Session Manager routes the calls based on the number contained in the request-uri, it is necessary to modify the user portion of the request-uri sent to Session Manager, to replace the pilot number with the actual number being called. Navigate to vsp session-config-pool entry ToPBX. Click on request-uri-specification.
On the user field, select to-uri from the drop-down menu, instead of the default requesturi. By making this change, the call is allowed to be routed to the correct destination by Session Manager, and ultimately by Communication Manager. Click Set to complete the configuration
7.2.6. Refer-To Header

This section presents a sample configuration that will cause the SBC to modify the host portion of the Refer-To header in a REFER message, replacing the enterprise domain with the value expected by AT&T. It should be noted that similar results could have been achieved by using additional adaptations at a more global level in Session Manager. Since this manipulation was needed only for specific Network Call Redirection cases, it was implemented here, taking advantage of the SBC granular header modification capabilities. In the left side menu, navigate to vsp session-config-pool entry ToPBX. Click on Configure next to header-settings.
On the right panel, select Add reg-ex-header as shown below.
In the new right pane, in the number field, since this is the first reg-ex-header rule, 1 was used. Enter Refer-To in the destination field and click Create.
On the following screen that is presented, select REFER for apply-to-methods. Use defaults for all other fields. Click the Configure link to the right of create.
The following screen is presented. In the source area, type Refer-To in the enter field. In the expression field, enter a regular expression to match. In the sample configuration, <sip:(.*)@avayalab2\.com(.*)> was entered. In this expression, the first (.*) will match and store any user part of the Refer-To header. The second instance of (.*) matches and stores any UUI if present. The domain avayalab2.com is what the AA-SBC would otherwise put in the Refer-To header host part. In the replacement field, <sip:\1@aslab.centixvoip.net\2> was entered. The variable \1 is the stored user part from the original Refer-To header containing the Refer-To number, corresponding to the first instance of (.*) from the expression. The variable \2 is any stored UUI from the original Refer-To header, corresponding to the second instance of (.*) from the expression. After completing the source, expression and replacement fields as appropriate, click Create.
The following screen shows the completed rule. Click Set to complete the configuration
7.2.7. Save the Configuration

To save the configuration, begin by clicking on Configuration in the left pane to display the configuration menu. Next, select Update and save configuration.
8. AT&T Mobility SIP Trunk Service Configuration

Information about how to establish the AT&T Mobility SIP Trunk Service in Puerto Rico can be obtained by contacting an AT&T Mobility sales representative. AT&T Mobility is responsible for the configuration of the AT&T Mobility SIP Trunk service in their network. To establish service, the customer will need to provide AT&T with the IP address used to reach the SBC at the enterprise. AT&T will provide the customer with the necessary information to configure the SIP connection from the enterprise site to the AT&T network, including: IP address of the AT&T SIP proxy. AT&T SIP domain. CPE SIP domain. Supported codecs. DID numbers Port numbers used for signaling and media.
This information is used to complete the Communication Manager, Session Manager, and the SBC configuration discussed in the previous sections.
9. Verification and Troubleshooting

This section provides verification steps that may be performed in the field to verify that the solution is configured properly. This section also provides a list of useful troubleshooting commands that can be used to troubleshoot the solution. Verification Steps: 1. Verify that endpoints at the enterprise site can place calls to the PSTN and that the call remains active for more than 35 seconds. This time period is included to verify that proper routing of the SIP messaging has satisfied SIP protocol timers. 2. Verify that endpoints at the enterprise site can receive calls from the PSTN and that the call can remain active for more than 35 seconds. 3. Verify that the user on the PSTN can end an active call by hanging up. 4. Verify that an endpoint at the enterprise site can end an active call by hanging up.
Troubleshooting: 1. Session Border Controller: Call Logs - On the web user interface of the SBC, the Call Logs tab can provide useful diagnostic or troubleshooting information. 2. Communication Manager: list trace station <extension number> Traces calls to and from a specific station. list trace tac <trunk access code number> Trace calls over a specific trunk group. status signaling-group <signaling group number> Displays signaling group service state. status trunk <trunk group number> Displays trunk group service state. status station <extension number> Displays signaling and media information for an active call on a specific station. 3. Session Manager: traceSM -x Session Manager command line tool for traffic analysis. Login to the Session Manager management interface to run this command. Call Routing Test - The Call Routing Test verifies the routing for a particular source and destination. To run the routing test, navigate to Home Elements Session Manager System Tools Call Routing Test. Enter the requested data to run the test.
10. Conclusion
AT&T Mobility SIP Trunk Service in Puerto Rico passed compliance testing. These Application Notes describe the configuration necessary to connect the above service to the Avaya Aura Solution for Midsize Enterprise 6.1. The AT&T Mobility SIP Trunk Service is a SIP-based Voice over IP solution for customers ranging from small businesses to large enterprises. AT&T Mobility SIP Trunk Service provides a flexible, cost-saving alternative to traditional hardwired telephony trunks.
11. References
This section references the documentation relevant to these Application Notes. Additional Avaya product documentation is available at http://support.avaya.com. [1] Avaya Aura Solution for the Midsize Enterprise (ME) 6.1 Intelligent Workbook. July 2011. [2] Installing and Configuring Avaya Aura Solution for Midsize Enterprise, Release 6.1, Issue 3.1, July 2011. [3] Avaya Aura Solution for the Midsize Enterprise Release Notes, Release 6.1, July 2011. [4] Installing and Configuring Avaya Aura System Platform, Release 6.0.3, February 2011. [5] Administering Avaya Aura System Platform, Release 6.0.3, February 2011. [6] Administering Avaya Aura Communication Manager, June2010, Document Number 03300509. [7] Avaya Aura Communication Manager Feature Description and Implementation, June 2010, Document Number 555-245-205. [8] Installing and Upgrading Avaya Aura System Manager, Release 6.1, November 2010. [9] Installing and Configuring Avaya Aura Session Manager, April 2011, Document Number 03-603473. [10] Administering Avaya Aura Session Manager, November 2010, Document Number 03603324. [11] Avaya Aura Session Border Controller System Administration Guide, September 2010. [12] Avaya one-X Deskphone H.323 Administrator Guide Release 6.1, May 2011, Document Number 16-300698. [13] Avaya one-X Deskphone SIP Administrator Guide Release 6.1, December 2010, Document Number 16-603838. [14] Administering Avaya one-X Communicator, October 2011. [15] Using Avaya one-X Communicator, Release 6.1, October 2011. [16] RFC 3261 SIP: Session Initiation Protocol, http://www.ietf.org/. [17] RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals, http://www.ietf.org/
Appendix A: Avaya Aura SBC Configuration File

# # Copyright (c) 2004-2011 Acme Packet Inc. # All Rights Reserved. # # File: /cxc/cxc.cfg # Date: 17:11:58 Wed 2011-10-19 # config cluster config box 1 set hostname me-sbc.sil.miami.avaya.com set timezone America/New_York set name me-sbc.sil.miami.avaya.com set identifier 00:ca:fe:51:81:77 config interface eth0 config ip mgmt set ip-address static 10.5.5.19/24 config ssh set mode ssh-2 return config snmp set trap-target 10.5.5.11 162 set trap-filter generic set trap-filter dos set trap-filter sip set trap-filter system return config web set ciphers TLS_RSA_WITH_AES_128_CBC_SHA,TLS_DHE_RSA_WITH_AES_128_CBC_SHA,TLS_DHE_DSS_WIT H_AES_128_CBC_SHA,SSL_RSA_WITH_3DES_EDE_CBC_SHA,SSL_DHE_RSA_WITH_3DES_EDE_CBC _SHA,SSL_DHE_DSS_WITH_3DES_EDE_CBC_SHA,TLS_RSA_WITH_AES_256_CBC_SHA,TLS_DHE_R SA_WITH_AES_256_CBC_SHA,TLS_DHE_DSS_WITH_AES_256_CBC_SHA return config web-service set protocol https 8443 set authentication certificate "vsp\tls\certificate ws-cert" return config icmp return config routing config route Default set gateway 10.5.5.254 return config route Static0 set destination network 192.11.13.4/30 set gateway 10.5.5.10 return config route Static1 set admin disabled return config route Static2 set admin disabled return config route Static3 MAA; Reviewed: SPOC 1/19/2012 Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved. 60 of 67 ATT-PR-AASME61
set admin disabled return config route Static4 set admin disabled return config route Static5 set admin disabled return config route Static6 set admin disabled return config route Static7 set admin disabled return config route MgmtDefault set gateway 10.5.5.254 return return return return config interface eth1 config ip inside set ip-address static 10.5.5.20/24 config sip set udp-port 5060 "" "" any 0 set tcp-port 5060 "" "" any 0 set tls-port 5061 "" "" TLS 0 "vsp\tls\certificate aasbc.p12" return config icmp return config media-ports return config routing config route Default set gateway 10.5.5.254 return return return return config interface eth2 config ip outside set ip-address static 172.16.1.5/24 config sip set udp-port 5060 "" "" any 0 return config media-ports set base-port 50000 set count 4999 return config routing config route Default set admin disabled return config route external-sip-media-1 set destination network 192.168.10.0/24 set gateway 172.16.1.254 MAA; Reviewed: SPOC 1/19/2012 Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved. 61 of 67 ATT-PR-AASME61
return return config kernel-filter config allow-rule allow-sip-udp-from-peer-1 set destination-port 5060 set source-address/mask 192.168.10.0/24 set protocol udp return config deny-rule deny-all-sip set destination-port 5060 return return return return config cli set prompt me-sbc.sil.miami.avaya.com return return return config services config event-log config file access.log set filter access info set count 3 return config file system.log set filter system info set count 3 return config file general.log set filter general info set count 3 return config file error.log set filter all error set count 3 return config file db.log set filter db debug set filter dosDatabase info set count 3 return config file management.log set filter management info set count 3 return config file peer.log set filter sipSvr info set count 3 return config file dos.log set filter dos alert set filter dosSip alert set filter dosTransport alert set filter dosUrl alert MAA; Reviewed: SPOC 1/19/2012 Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved. 62 of 67 ATT-PR-AASME61
set count 3 return config file krnlsys.log set filter krnlsys debug set count 3 return return return config master-services config database set media enabled return return config vsp set admin enabled config default-session-config config media set anchor enabled set rtp-stats enabled return config sip-directive set directive allow return config log-alert return config third-party-call-control set handle-refer-locally disabled return return config tls config default-ca set ca-file /cxc/certs/sipca.pem return config certificate ws-cert set certificate-file /cxc/certs/ws.cert return config certificate aasbc.p12 set certificate-file /cxc/certs/aasbc.p12 set passphrase-tag aasbc-cert-tag return return config session-config-pool config entry ToTelco config to-uri-specification set host next-hop return config from-uri-specification set host local-ip return config request-uri-specification set host next-hop return config p-asserted-identity-uri-specification return MAA; Reviewed: SPOC 1/19/2012 Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved. 63 of 67 ATT-PR-AASME61
config header-settings set blocked-header P-Location set blocked-header Alert-Info config altered-header 1 set source-header Diversion set source-field selection ".*avayalab2\.com" aslab.centixvoip.net set destination Diversion set destination-field host return return return config entry ToPBX config to-uri-specification set host next-hop-domain return config request-uri-specification set user to-uri set host next-hop-domain return config header-settings config reg-ex-header 1 set destination Refer-To set create Refer-To "<sip:(.*)@avayalab2\.com(.*)>" "<sip:\1@aslab.centixvoip.net\2>" set apply-to-methods REFER return return return config entry Discard config sip-directive return return return config dial-plan config route Default set priority 500 set location-match-preferred exclusive set session-config vsp\session-config-pool\entry Discard return config source-route FromTelco set peer server "vsp\enterprise\servers\sip-gateway PBX" set source-match server "vsp\enterprise\servers\sip-gateway Telco" return config source-route FromPBX set peer server "vsp\enterprise\servers\sip-gateway Telco" set source-match server "vsp\enterprise\servers\sip-gateway PBX" return return config enterprise config servers config sip-gateway PBX set domain avayalab2.com set failover-detection ping set outbound-session-config-pool-entry vsp\session-config-pool\entry ToPBX config server-pool MAA; Reviewed: SPOC 1/19/2012 Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved. 64 of 67 ATT-PR-AASME61
config server PBX1 set host 10.5.5.15 set transport TCP return return return config sip-gateway Telco set failover-detection ping set ping-interval 300 set outbound-session-config-pool-entry vsp\session-config-pool\entry ToTelco config server-pool config server Telco1 set host 192.168.10.250 return return return return return config dns config resolver config server 192.168.10.100 return return return config settings set read-header-max 8191 return return config external-services return config preferences config gui-preferences set enum-strings SIPSourceHeader Diversion set enum-strings SIPSourceHeader Refer-To return return config access config permissions superuser set cli advanced return config permissions read-only set config view set actions disabled return config users config user admin set password 0x00d8b88dfc7517d214bd8c404489c41bf575547211ad1a7521c05d24f5 set permissions access\permissions superuser return config user cust set password 0x0069d2a6686bb2d23563e1e4cd90275bd1735222619152e9d64ac2385c set permissions access\permissions read-only MAA; Reviewed: SPOC 1/19/2012 Solution & Interoperability Test Lab Application Notes 2012 Avaya Inc. All Rights Reserved. 65 of 67 ATT-PR-AASME61
return config user init set password 0x002ca274c8fd8f18d046301ba7127d77562fd2391e039de0735b0dd7b9 set permissions access\permissions superuser return config user craft set password 0x006caa3956d62ee91793108eb4b2a4fa4c6fc08f9e16c9bcbc163bd22e set permissions access\permissions superuser return config user dadmin set password 0x00d4041dc8a804e2c42bacc8258c9ee9c757b0797d4c8019ddc4f90926 set permissions access\permissions read-only return return return config features return
2012
Avaya Inc. All Rights Reserved. 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 e-mail 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 DevConnect Program at devconnect@avaya.com.

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Avaya Solution & Interoperability Test Lab

MAA; Reviewed: SPOC 1/19/2012

MAA; Reviewed: SPOC 1/19/2012

2. General Test Approach and Test Results

2.1. Interoperability Compliance Testing

2.2. Test Results

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3.1. Midsize Enterprise Solution Components

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4. Equipment and Software Validated

5. Configure Communication Manager.

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5.1. Licensing and Capacity

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5.2. System Features

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5.3. IP Node Names

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5.5. IP Network Region

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5.6. Signaling Group

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5.7. Trunk Group

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5.8. Calling Party Information

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5.9. Inbound Routing

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5.10. Outbound Routing

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6. Configure Avaya Aura Session Manager

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6.1. System Manager Login and Navigation

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6.2. SIP Domains

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6.5. SIP Entities

MAA; Reviewed: SPOC 1/19/2012

MAA; Reviewed: SPOC 1/19/2012

MAA; Reviewed: SPOC 1/19/2012

MAA; Reviewed: SPOC 1/19/2012

6.6. Entity Links

SIP Entity 2: Port: Trusted: Click Commit to save.

MAA; Reviewed: SPOC 1/19/2012

Entity Link to the SBC:

MAA; Reviewed: SPOC 1/19/2012

6.7. Routing Policies

MAA; Reviewed: SPOC 1/19/2012

6.8. Dial Patterns