Configuring Avaya Voice Portal with F5 BIG-IP Local

FS; Reviewed: SPOC 9/2/2008

Solution & Interoperability Test Lab Application Notes ©2008 Avaya Inc. All Rights Reserved.

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Avaya Solution & Interoperability Test Lab

Configuring Avaya Voice Portal with F5 BIG-IP Local Traffic Manager and Cisco 3825 Integrated Services Router/Voice Gateway – Issue 1.0

Abstract

These Application Notes describe the configuration steps required to enable routing of calls between Avaya Voice Portal and the PSTN through a SIP infrastructure consisting of the F5 BIG-IP Local Traffic Manager (LTM) and the Cisco 3825 Integrated Services Router/Voice Gateway. This solution allows Avaya Voice Portal to receive calls from the PSTN and transfer calls to the PSTN. The Cisco 3825 interfaces to the PSTN via a T1/ISDN interface. The F5 BIG-IP LTM supports load balancing and failover of the incoming SIP traffic to Avaya Voice Portal.



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1. Introduction These Application Notes describe the configuration steps required to enable calls between Avaya Voice Portal and the PSTN through a SIP infrastructure consisting of the F5 BIG-IP Local Traffic Manager (LTM) and the Cisco 3825 Integrated Services Router/Voice Gateway (also referred to as “F5 BIG-IP LTM” and “Cisco 3825,” respectively). This solution allows Avaya Voice Portal to receive calls from the PSTN and transfer calls to the PSTN. The Cisco 3825 interfaces to the PSTN via a T1/ISDN interface. Refer to Figure 1 for an illustration of the test configuration. The following features were covered during the interoperability testing of Avaya Voice Portal, F5 BIG-IP LTM, and Cisco 3825 in a SIP environment:

T1/ISDN network interface between Cisco 3825 and simulated PSTN Basic calls from PSTN to Avaya Voice Portal Modification of the dialed number (DNIS) by the Cisco 3825 to a Voice Portal

Application ID SIP call routing by the F5 BIG-IP LTM between the Cisco 3825 and Avaya Voice Portal Load balancing and failure recovery features of the F5 BIG-IP LTM in routing incoming

calls to Avaya Voice Portal Call transfers from Avaya Voice Portal to PSTN, including blind, supervised, and

bridged transfers DTMF tones / RFC 2833 support G.711 mu-law

Figure 1: SIP-to-PSTN Interworking with Avaya Voice Portal, F5 BIG-IP LTM and Cisco 3825



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2. Equipment and Software Validated The following equipment and software were used for the sample configuration:

Equipment Software

Avaya Voice Portal Voice Portal Management System (VPMS) Media Processing Platform (MPP)

4.1.0.1.2710 4.0.0.1.0005

Cisco 3825 Integrated Services Router/Voice Gateway 2.4(15)T4 (IP Voice)

F5 BIG-IP Local Traffic Manager BIG-IP 9.4.4 Build 73.1 Hotfix HF1

Nuance Speech Server Nuance OpenSpeech Recognizer Nuance RealSpeak

3.0 4.0

Application Server – HTTP Server running on Windows 2003 Server Service Pack 2

Table 1: Equipment and Software



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3. Configure Avaya Voice Portal This section covers the administration of Avaya Voice Portal, including:

Configuring a SIP connection Adding MPP servers Configuring the VoIP audio format Adding a speech server • Adding applications Starting the MPP servers

Avaya Voice Portal is configured via the Voice Portal Management System (VPMS) web interface. Access the web interface at http://<ip-addr>/VoicePortal, where <ip-addr> is the IP address of the VPMS. Log in using the Administrator user role. The screen shown in Figure 2 is displayed. Note: The navigation steps to each screen are displayed at the top of each screen.

Figure 2: VPMS Main Screen



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Configure the SIP Connection. To configure a SIP connection, navigate to the VoIP Connections screen and then click on the SIP tab. Click Add. The SIP Connection form is displayed as shown in Figure 3. Configure the parameters as described below:

Select TCP as the Proxy Transport Select the “Yes” radio button for Enable Specify the IP address of the F5 BIG-IP LTM in the Proxy Server Address field Set the Proxy Server Port to 5060 for TCP

Click on Continue (not shown), and additional fields will be displayed:

Set Listener Port to 5060 for TCP Set the SIP Domain (e.g., avaya.com) Set the Maximum Simultaneous Calls. In this example, a maximum of 40 calls is

supported among two MPP servers. Accept the default values for the other fields

Click on Save.

Figure 3: SIP Connection



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Add the Media Processing Platform (MPP) Servers. Add an MPP server by navigating to the MPP Servers screen and clicking on Add. Specify a descriptive Name and enter the IP address of the MPP server in the Host Address field. Click on Continue and specify the Maximum Simultaneous Calls supported by the MPP server. Check the box Trust this certificate within the MPP Certificate window, and click on Save. Figure 4 shows the configuration for the first MPP server. Repeat these steps for the second MPP server.1

Figure 4: MPP Server

1 Although two MPP servers were used in this configuration to illustrate the load balancing and fail-over capabilities of the F5 BIG-IP LTM, only one MPP is required when deploying an Avaya Voice Portal system.



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Configure the VoIP Audio Format. Continue from the previous step in the MPP Servers screen and click on VoIP Settings. The Cisco 3825 was configured for G.711 mu-law, so the MPP Native Format field in Figure 5 is set to audio/basic for mu-law. Click on Save.

Figure 5: VoIP Settings



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Add an Automatic Speech Recognition (ASR) Server. To configure an ASR server, click on Speech Servers in the left pane, select the ASR tab, and then click Add. Set the Engine Type to the appropriate value. In this configuration, a Nuance ASR server was used so the engine type was set to Nuance. Set the Network Address field to the IP address of the speech server, Total Number of Licensed ASR Resources to the appropriate value, and select the desired Languages to be supported. The other fields can be left at their default values. Click on Save.

Figure 6: ASR Speech Servers



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Add a text-to-speech (TTS) Server. To configure the (TTS) server, click on Speech Servers in the left pane, select the TTS tab, and then click Add. Set the Engine Type to the appropriate value. In this configuration, a Nuance TTS server was used so the engine type was set to Nuance. Set the Network Address field to the IP address of the speech server, Total Number of Licensed TTS Resources to the appropriate value, and select the desired Languages to be supported. The other fields can be left at their default values. Click on Save.

Figure 7: TTS Server



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Add an Application. On the Applications page, click on Add. Specify a Name for the application (e.g., Intro), set the MIME Type field to the appropriate value (e.g., VoiceXML), and set the VoiceXML URL field to point to a VoiceXML application on the application server. Next, specify the type of ASR and TTS servers to be used by the application (e.g., Nuance) and the Application Launch Type of “Inbound.” Then specify the called number that invokes the application. In the sample configuration, the Cisco 3825 changes the called number to 1011111. As shown in Figure 8, a called number range has been configured, so that all called numbers in the range will launch this application. Click on Save (not shown).

Figure 8: Application



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Start the MPP Servers. Start the MPP servers from the MPP Manager page shown in Figure 9. Select each MPP and then click the Start button. The Mode of each MPP should be Online and the State should be Running.

Figure 9: MPP Manager

4. Configure F5 BIG-IP Local Traffic Manager This section provides the steps required for the F5 Big-IP LTM to support the session management functions described in Section 1, including configuration of:

Network interfaces Default IP route IRules for SIP message modification to support interoperability Custom server health monitor Server Pools SIP NAT (SNAT) pools Virtual servers Separate SIP inbound/outbound transport types

To implement these steps via the web interface, a management IP interface must be configured on the F5 LTM using the front panel buttons and display. These application notes assume that this has already been done using the Quick Start instructions in Reference [2], and that the management interface was set to 192.45.122.102. They also assume that the product has already



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been properly licensed and login/password configuration performed. Set the URL in a web browser to this IP address, log in, and perform the following steps. Note: Any fields not mentioned in the configuration steps should be left at their default values. All configurations in this section are automatically saved in the file /config/bigip.conf. The bigip.conf file for the sample configuration is shown in Appendix A. Configure Network Interfaces. As shown in Figure 1, the F5 LTM acts as a SIP session manager between branch locations and the main data center of an enterprise. Each branch contains a Cisco 3825 that converts incoming PSTN calls to SIP and routes them via UDP to the “PSTN” virtual server interface of the F5 BIG-IP LTM in the data center (10.3.3.100). The F5 BIG-IP LTM then routes the SIP calls over its “VP” virtual server interface (192.45.122.100) via TCP to Avaya Voice Portal. There are two steps to configure the network interfaces:

1. Define VLANs – these are used to logically associate self IP interfaces with physical interfaces on the F5 BIG-IP LTM. Define two VLANs, one for the PSTN side interface (physical interface 1.2), and one for the Avaya Voice Portal side interface (1.1). Click on VLANs under Network in the left pane, and then click Create…. Shown below are the settings for the PSTN side. Enter a Name and Tag, and select one of the physical Interfaces under Resources by clicking on the << button next to the Untagged box. Click on Finished.

Figure 10: Adding a VLAN



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Figure 11 shows the result of configuring both VLANs.

Figure 11: VLANs Configured

2. Define the self IP interfaces for the PSTN side (10.3.3.102) and the Avaya Voice

Portal side (192.45.122.102), and associate them with the VLANs. Figure 12 shows the settings for the PSTN side. Click on Self IPs under Network in the left pane, and then click Create…. Enter the IP Address and Netmask, and select the appropriate VLAN, as defined in the previous step. Click on Finished.

Figure 12: Adding a Self IP



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Figure 13 shows the result of defining both self IPs.

Figure 13: Self IPs Configured

Configure Default IP Route. Click on Routes under Network in the left pane, and then click on Add….Type defaults to “Default Gateway”. Enter the IP address in the Resource section.

Figure 14: Default IP Route



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Configure any required iRules. iRules provide a flexible way to optionally specify how SIP messages should be modified by the F5 BIG-IP LTM as they are routed between different SIP network devices, and can resolve SIP implementation differences in multi-vendor or custom solutions. In the sample configuration, five iRules were defined, as shown in Table 2. iRule Name Description

via2 Splits up a compound Via header into the appropriate number of individual Via headers

incoming-route Insert TCP transport parameter in messages routed to Avaya Voice Portal outgoing-route Insert UDP transport parameter in messages routed to Cisco 3825 map_single_DNIS Maps incoming number to an Avaya Voice Portal application ID map_range_DNIS Maps incoming number range to an Avaya Voice Portal application ID

Table 2: iRule Descriptions To define an iRule, click on iRules under Local Traffic in the left pane, and then click Create…. Specify a Name for the iRule, type the specification text in the Definition window, and click on Finished. The screen in Figure 15 shows the definition for “via2”. Repeat this step for the remaining two iRules. Definitions all iRules listed in Table 2 are shown in Appendix A. Information on how to specify iRules is available at http://devcentral.f5.com.

Figure 15: Adding an iRule



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Configure Custom Health Monitors. The F5 BIG-IP LTM uses health monitors to determine whether a server to which SIP messages can be routed is operational. Servers that are flagged as not operational will cause the F5 BIG-IP LTM to route to another operational server in the same server pool. One health monitor, “gateway-icmp” uses ICMP PING to determine server health. If standard health monitors such as this are acceptable, then this step can be skipped. In the sample configuration, it was desirable to shorten the frequency and timeout parameters of this monitor to provide rapid fail-over if a server becomes unavailable. To define this customized version, click on Monitors under Local Traffic in the left pane, and then click Create…. Specify a Name, and select “Gateway ICMP” from the Type pull-down menu. Enter an Interval of 2 and a Timeout of 2 seconds and click on Finished. Figure 16 shows configuration of the customized “avaya-gateway-icmp-monitor”.

Figure 16: Adding a Health Monitor



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Configure Server Pools. The servers with which the F5 BIG-IP LTM communicates are organized into pools. When SIP calls are routed to a server pool, additional features such as fail-over and load balancing can be specified. In the sample configuration, the PSTN side pool will consist of a single server (the Cisco 3825), whereas the Avaya Voice Portal side will contain two MPP servers. To define the Avaya Voice Portal pool, click on Pools under Local Traffic in the left pane, and then click Create…. Specify a Name of “VP_POOL” and under Health Monitors, move “avaya-gateway-icmp-monitor” into the Active window using the << button. Under the New Members section of Resources, specify the Address (192.45.122.131) and Service Port (5060) of the first MPP server and click on Add. Then enter the information for the second MPP server and click on Add again. The current set of member entries will be displayed in the text box underneath. Click on Finished. Figure 17 shows the screen for configuring this pool. Repeat this step for the PSTN side pool.

Figure 17: Adding the VP_POOL Server Pool



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Configure SIP NAT (SNAT) Pools. A SNAT pool specifies one or more IP addresses other than the “self IP” that should be used by the F5 BIG-IP LTM when routing SIP requests via a virtual server (see next step). The F5 LTM also uses this IP address in SIP message headers. Two SNAT pools are used in the sample configuration; one for routing incoming SIP requests from the PSTN to the Avaya Voice Portal side (192.45.122.100) and one for routing outgoing SIP requests from Avaya Voice Portal to the PSTN (10.3.3.100). To define the SNAT pool for the incoming requests, click on SNATs under Local Traffic in the left pane, click on the SNAT Pool List tab at the top of the right pane, and then click Create…. The form shown in Figure 18 will be displayed. Under General Properties, specify a Name (sip_snat), and under Configuration, enter the IP Address to be used (192.45.122.100), and click on Add. Then click on Finished. Repeat this step for the other SNAT pool in the sample configuration, named “sip-pstn” and using 10.3.3.100.

Figure 18: Adding a SNAT Pool Configure Virtual Servers. A virtual server controls how SIP call requests are routed to a pool of servers. A destination IP address defines where the virtual server receives requests, and designation of a server pool specifies which servers will receive those requests, as well as what load balancing and fail-over mechanisms will be employed. In the sample configuration, two virtual servers are required; one for incoming calls from the PSTN to Avaya Voice Portal, and another for outgoing calls.



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To define the virtual server for incoming calls, click on Virtual Servers under Local Traffic in the left pane, and then click Create…. The form shown in Figures 19 and 20 will be displayed. Under General Properties, specify the Name (PSTN-INCOMING) of the virtual server and the IP Address (10.3.3.100) and Service Port (5060) on which it should receive SIP requests. Under Configuration, select “sip” for SIP Profile, and the previously defined “sip_snat” for the SNAT Pool. Note that Protocol is left at the default of TCP for both sides of the virtual server. Since the Cisco 3825 uses UDP in the sample configuration, this requires a separate configuration step, described next under “Configure UDP transport for PSTN side.” Under Resources, move the previously defined iRules “via2” and “”incoming-route” from the Available box to the Enabled box, and select “sip_info” for Default Persistence Profile. Click on Finished. Repeat the above step for the virtual server that will handle outgoing calls, except for the following differences in field values: Name “PSTN-OUTGOING” SNAT Pool “sip-pstn” iRules “via2”, “outgoing-route”

Figure 19: Adding a Virtual Server



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Figure 20: Adding a Virtual Server (cont’d)



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Configure UDP transport for PSTN side. As mentioned in the previous step, the F5 BIG-IP LTM web interface assumes that the specified transport (TCP) is used for both inbound and outbound SIP signaling by the defined virtual servers. In the sample configuration, the Cisco 3825 is using UDP, and Avaya Voice Portal is using TCP. This mixed transport configuration is achieved by manually editing an F5 BIG-IP LTM configuration file, /config/bigip.conf, changing the transport for the PSTN side of each of the two virtual servers. To do this, use secure shell (ssh) or putty to log into the management interface (192.45.122.102 in the sample configuration) with the appropriate root credentials. Use a text editor such as vi to edit the file, changing the lines shown in bold in Figure 21 to those shown in bold in Figure 22 . . . . virtual PSTN-INCOMING { snatpool sip_snat pool VP_POOL destination 10.3.3.100:5060 ip protocol udp rules via2 incoming-route map_single_DNIS profiles sip tcp persist sip_info } virtual PSTN-OUTGOING { snatpool sip-pstn pool PSTN_POOL destination 192.45.122.100:5060 ip protocol tcp rules via2 outgoing-route profiles sip tcp persist sip_info } . . .

Figure 21: Excerpt of /config/bigip.conf Before Editing



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.

.

. virtual PSTN-INCOMING { snatpool sip_snat pool VP_POOL destination 10.3.3.100:5060 ip protocol udp rules via2 incoming-route map_single_DNIS profiles sip tcp serverside udp clientside persist sip_info } virtual PSTN-OUTGOING { snatpool sip-pstn pool PSTN_POOL destination 192.45.122.100:5060 ip protocol tcp rules via2 outgoing-route profiles sip tcp clientside udp serverside persist sip_info } . . .

Figure 22: Excerpt of /config/bigip.conf After Editing



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Save the file and exit the editor. Then enter the b load command to reload the modified configuration file. The output of this command should appear as shown in Figure 23. Note: After making this change, all further changes to the virtual servers should be made by editing bigip.conf. If changes are made using the web interface, then use the web interface to delete the appropriate virtual server(s) and add them back. Then repeat this step. Changes to components other than the virtual servers can still be made using the web interface. [root@f5ltm:Active] config # b load Reading configuration from /defaults/config_base.conf. Reading configuration from /config/bigip_base.conf. Reading configuration from /config/bigip_sys.conf. Reading configuration from /usr/bin/monitors/builtins/base_monitors.conf. Reading configuration from /config/profile_base.conf. Reading configuration from /config/daemon.conf. Reading configuration from /config/bigip.conf. Reading configuration from /config/bigip_local.conf. Loading the configuration ... [root@f5ltm:Active] config #

Figure 23: Result of Executing “b load” Via Command Line Interface



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5. Configure Cisco 3825 Router/Voice Gateway This section provides the configuration of the Cisco 3825. The T1/PRI interface connects to a simulated PSTN network. Figure 24 shows the T1/PRI interface definition. Figure 25 shows the SIP call routing and number translation configuration. See References [3-4] for more information. ! ! PSTN T1/PRI interface ! controller T1 2/0 framing esf linecode b8zs pri-group timeslots 1-24 ! interface Serial2/0:23 no ip address encapsulation hdlc isdn switch-type primary-5ess isdn timer T310 40000 isdn incoming-voice modem isdn T309-enable no fair-queue no cdp enable no clns route-cache !

Figure 24: Cisco 3825 Configuration – T1/PRI Interface

! ! Translation rule for changing incoming called number (DNIS) to Voice Portal Application ID ! translation-rule 10 Rule 1 732-555-1212 1011111 ! ! Dial peer to handle incoming calls, routed to F5 BIG-IP LTM dial-peer voice 732 voip service session destination-pattern 22940 progress_ind setup enable 1 translate-outgoing called 10 session protocol sipv2 session target ipv4:10.3.3.100 session transport udp incoming called-number 7......... dtmf-relay rtp-nte codec g711ulaw ! ! Dial peer to handle outbound calls to PSTN ! dial-peer voice 732 pots service session destination-pattern 732....... direct-inward-dial port 2/0:23 forward-digits 10 prefix *9 !

Figure 25: Cisco 3825 Configuration – Number Translation and Call Routing



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6. Verification Steps This section provides the verification steps that may be performed to verify that Avaya Voice Portal can establish calls with the PSTN through the F5 BIG-IP LTM and Cisco 3825.

1. From the VPMS web interface, verify that the MPP servers are online and running in the System Monitor page, shown in Figure 26.

Figure 26: Avaya Voice Portal System Monitor



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2. From the VPMS web interface, verify that the ports on the MPP servers are in service in the Port Distribution page shown in Figure27.

Figure 27: Avaya Voice Portal Port Distribution



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3. Using the F5 BIG-IP LTM web interface, click on Virtual Servers under Local Traffic in the left pane. If any status icon is red, as shown in Figure 28, determine the pool being used by that virtual server by clicking on it and then on Resources at the top of the right pane (Figure 29).

Figure 28: F5 BIG-IP LTM Virtual Server Status

Figure 29: F5 BIG-IP LTM Virtual Server/Server Pool Association



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Click on Pools under Local Traffic to determine whether that server pool is out of service (Figure 30) because its servers failed health monitor tests. Note that a virtual server and pool will remain in service as long as at least one pool member is in service.

Figure 30: F5 BIG-IP LTM Server Pool Status

Click on the pool with the red status icon, and then click on Members at the top of the right pane to see that status of the members of the server pool (Figure 31). Use the server IP addresses shown to further diagnose any network issues.

Figure 31: F5 BIG-IP LTM Server Pool Member Status



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4. From the Cisco 3825, use the show controller command to verify that the controller is up.

Cisco-3825#show controller t1 2/0 T1 2/0 is up. Applique type is Channelized T1 Cablelength is long gain36 0db No alarms detected. alarm-trigger is not set Soaking time: 3, Clearance time: 10 AIS State:Clear LOS State:Clear LOF State:Clear Version info Firmware: 20071011, FPGA: 255, spm_count = 0 Framing is ESF, Line Code is B8ZS, Clock Source is Line. Current port master clock:recovered from controller 2/0

Figure 32: Controller Status

5. From the Cisco 3825, use the show interface serial command to verify that the serial

interface is up. Cisco-3825#show interface serial 2/0:23 Serial2/0:23 is up, line protocol is up (spoofing) Hardware is DSX1 MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec, reliability 255/255, txload 1/255, rxload 1/255 Encapsulation HDLC, loopback not set Last input 00:00:03, output never, output hang never Last clearing of "show interface" counters never Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0 Queueing strategy: fifo Output queue: 0/40 (size/max) 5 minute input rate 0 bits/sec, 0 packets/sec 5 minute output rate 0 bits/sec, 0 packets/sec 16931 packets input, 73209 bytes, 0 no buffer Received 0 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 16931 packets output, 72803 bytes, 0 underruns 0 output errors, 0 collisions, 0 interface resets 0 output buffer failures, 0 output buffers swapped out 1 carrier transitions

Figure 33: Serial Interface Status



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6. From the Cisco 3825, use the show isdn status command to verify that the ISDN link is established.

Cisco-3825#show isdn status Global ISDN Switchtype = primary-ni ISDN Serial2/0:23 interface dsl 0, interface ISDN Switchtype = primary-5ess Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x807FFFFF Number of L2 Discards = 0, L2 Session ID = 0 Total Allocated ISDN CCBs = 0

Figure 34: ISDN Status

7. From a phone on the PSTN, place a call to Avaya Voice Portal. Verify that the call is

established successfully and that the proper greeting is provided. Transfer the call to another user on the PSTN. Verify that two-way audio is established between the two PSTN users.

7. Support F5 support is available from http://support.f5.com. Cisco technical support is available at http://www.cisco.com. Alternative Cisco support options are described at http://www.cisco.com/en/US/support/index.html.

8. Conclusion These Application Notes describe the configuration steps required to allow incoming and outgoing calls between Avaya Voice Portal and the PSTN through a SIP infrastructure consisting of the F5 BIG-IP Local Traffic Manager and the Cisco 3825 Integrated Services Router/Voice Gateway.

9. References This section references the product documentation that is relevant to these Application Notes. [1] Installing and Administering SIP Enablement Services, Document 03-600768, Issue 4.0, May

2007, available at http://support.avaya.com. [2] BIG-IPQuick Start Instructions, available at http://www.f5.com. [3] Cisco IOS Interface and Hardware Component Configuration Guide, available at

http://www.cisco.com. [4] Dial Peer Configuration on Voice Gateway Routers, available at http://www.cisco.com.



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10. Appendix A The following listing shows the contents of the F5 configuration file /config/bigip.conf corresponding to the sample configuration outlined in Section 4. partition Common { description "Repository for system objects and shared objects." } user root { password crypt "$1$j6hphSTL$3btcIXc0.C2RVerIwQ3Hn0" } route default inet { gateway 10.3.3.101 static } snatpool sip-pstn { members 10.3.3.100 } snatpool sip_snat { members 192.45.122.100 } shell write partition Common user admin { password crypt "$1$wKYTpgcZ$KPCOo9jj6ncsVudAwqDes1" description "Admin User" id 0 group 500 home "/home/admin" shell "/bin/false" role administrator in all } user f5emsvr { password crypt "!!" description "F5 EM Service Account" id 975 group 975 home "/root" shell "/bin/false" role guest in all } user jcollins { password crypt "$1$8BGi7c5z$XOsQulcBcH.qNJYsH723I0" description "jcollins" id 0 group 500 home "/home/jcollins" shell "/bin/bash" role administrator in all } monitor avaya-gateway-icmp-monitor { defaults from gateway_icmp interval 2 timeout 2



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} monitor sip_monitor_custom { defaults from sip debug "no" mode "tcp" } monitor test-gw-icmp { defaults from gateway_icmp interval 2 timeout 2 } monitor vp_icmp { defaults from icmp interval 2 timeout 2 } profile persist sip_info { mode sip mirror disable timeout 180 sip info "Call-ID" across services disable across virtuals disable across pools disable rule none } profile sip sip { max size 65535 terminate bye enable insert via enable secure via disable insert record route disable } profile sip avaya_sip { defaults from sip max size 65535 terminate bye enable insert via enable secure via disable insert record route disable } configsync { password crypt "TO+UC?U[G_IQI\\GmIs1eg\\KmJ4;bJ`;Vcs6YOSmVHnIdVVJ" } node 10.91.2.66 { } node 10.91.2.102 { } node 192.45.122.135 { } node 192.45.122.136 { } node 192.45.122.137 { } pool PSTN_POOL {



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monitor all avaya-gateway-icmp-monitor members 10.3.3.20:5060 } pool VP_POOL { monitor all min 1 of avaya-gateway-icmp-monitor members 192.45.122.131:5060 192.45.122.132:5060 } rule incoming-route { when SIP_REQUEST { SIP::header remove Route SIP::header insert Record-route "<sip:192.45.122.100;transport=tcp;lr>" } when SIP_RESPONSE_SEND { SIP::header remove Record-route SIP::header insert Record-route "<sip:10.3.3.100;transport=udp;lr>" } } rule map_range_DNIS { when RULE_INIT { # Forward mapping # Format of map array # range(lowest:highest) translate-number array set map { 9000001:9000009 9000000 1011111:1011122 1011112 } # Reverse mapping is created on-the-fly and kept for this period of time (second) set tmout 600 } when SIP_REQUEST { set n [findstr [SIP::header "To"] "sip:" 4 "@"] foreach { r m } [array get ::map] { set l [getfield $r ":" 1] set u [getfield $r ":" 2] if { $n >= $l && $n <=$u } { SIP::uri [regsub $n [SIP::uri] $m] set tmp [SIP::header "To"] SIP::header insert "To" [regsub $n $tmp $m] SIP::header remove "To" 1 session add uie [SIP::call_id] $n $::tmout break } } } when SIP_RESPONSE_SEND { set m [findstr [SIP::header "To"] "sip:" 4 "@"] set n [session lookup uie [SIP::call_id]] if { $n != "" } { SIP::uri [regsub $m [SIP::uri] $n] set tmp [SIP::header "To"]



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SIP::header insert "To" [regsub $m $tmp $n] SIP::header remove "To" 1 set tmp [SIP::header "Contact"] if { $tmp != "" } { SIP::header remove "Contact" SIP::header insert "Contact" [regsub $m $tmp $n] } } } } rule map_single_DNIS { when RULE_INIT { array set a2b { 1011111 1011112 } array set b2a { 1011112 1011111 } } when SIP_REQUEST { set a [findstr [SIP::header "To"] "sip:" 4 "@"] log local0. "SIP REQUEST Method = [SIP::method] A= $a" if { [ info exists ::a2b($a) ] } { SIP::uri [regsub $a [SIP::uri] $::a2b($a)] set tmp [SIP::header "To"] SIP::header insert "To" [regsub $a $tmp $::a2b($a)] SIP::header remove "To" 1 } else { log local0. "can't find match number" #pass as is } #log local0. "[SIP::uri] [SIP::to]" } when SIP_RESPONSE { set b [findstr [SIP::header "To"] "sip:" 4 "@"] log local0. "SIP REQUEST Method = [SIP::method] A= $b" if { [ info exists ::b2a($b) ] } { SIP::uri [regsub $b [SIP::uri] $::b2a($b)] set tmp [SIP::header "To"] SIP::header insert "To" [regsub $b $tmp $::b2a($b)] SIP::header remove "To" 1 set tmp [SIP::header "Contact"] if { $tmp != "" } { SIP::header remove "Contact" SIP::header insert "Contact" [regsub $b $tmp $::b2a($b)] } } else { log local0. "can't find match number" #pass as is } }



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} rule outgoing-route { when SIP_REQUEST { SIP::header remove Route `SIP::header insert Record-route "<sip:10.3.3.100;transport=udp;lr>" } when SIP_RESPONSE_SEND { SIP::header remove Record-route SIP::header insert Record-route "<sip:192.45.122.100;transport=tcp;lr>" } } rule via2 { when SIP_RESPONSE { set via0 [SIP::via 0] set comma [string first "," $via0] if {$comma != -1} { SIP::header remove Via 0 SIP::header insert Via [string range $via0 \ [expr $comma + 1] [string length $via0]] SIP::header insert Via [string range $via0 0 [expr $comma - 1]] } } } virtual PSTN-INCOMING { snatpool sip_snat pool VP_POOL destination 10.3.3.100:5060 ip protocol udp rules via2 incoming-route map_range_DNIS profiles sip tcp serverside udp clientside persist sip_info } virtual PSTN-OUTGOING { snatpool sip-pstn pool PSTN_POOL destination 192.45.122.100:5060 ip protocol tcp rules via2 outgoing-route profiles sip tcp clientside udp serverside



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persist sip_info } node * monitor icmp



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Configuring Avaya Voice Portal with F5 BIG-IP Local