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Table of Contents

 

Document Overview

This document provides a configuration guide for Sonus SBC 5XX0 Series (Session Border Controller) when connecting to Genesys Voice Platform.

This configuration guide supports features documented in the Genesys Voice Platform guides.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound calls flows between Sonus SBC 5XX0 and Genesys Voice Platform.

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. There will be steps that require navigating third-party as well as the Sonus SBC Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and for troubleshooting, if necessary.

This configuration guide is offered as a convenience to Sonus customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate, but are presented without warranty of any kind, express or implied, and are provided "AS IS." Users must take full responsibility for the application of the specifications and information in this guide.


Requirements

The following equipment and software were used for the sample configuration provided:

 

 

Equipment

Software Version

Sonus Networks 

Sonus SBC 5200
BMC
BIOS
ConnexIP OS
SonusDB
EMA
SBX

V05.01.01-R000
V02.16.00
V02.06.00
V03.01.01-R000
V05.01.01-R000
V05.01.01-R000
V05.01.01-R000

Third-party Equipment



Genesys SIP Server8.1.100.98
Genesys GVP MCP

8.1.504.93

Polycom SoundPoint IP 501 SIP

2.1.3


Reference Configuration

The following reference configuration shows connectivity between third-party and Sonus SBC 5XX0.

Support

For any questions regarding this document or the content herein, please contact your maintenance and support provider.

 

Third-Party Product Features

The testing was executed with the Genesys test plan. The following features were tested:

Inbound calls 

Outbound Calls

Music on hold

Call Transfers

Other scenarios

Remote Agent behind SBC scenarios

Verify License

SBC-POL-RTU

 

 

Genesys Voice Platform Configuration

The following new configurations are included in this section:

  1. Accessing Genesys Tools and Interfaces
  2. Creating SIP Switch in Genesys Administrator
  3. SIP Server Configuration in Genesys Administrator
  4. Genesys Media Server Deployment
  5. Stat Server Configuration

1. Accessing Genesys Tools and Interfaces

This section provides the configuration required for the Genesys components.

Genesys is configured using several different tools and interfaces. The tools and interfaces used in this document are shown below to include their location and method of access.

To access these items, a Remote Desktop Connection (RDC) to the Genesys server is required. The username, password, and IP address of the system to be accessed should be provided by the person(s) installing the Genesys system.

Once logged onto the Genesys system, click the Start button and look for the installed applications shown below. If the applications are not visible on the Start Menu, find them using the search box just above the Start button.

 

Steps can be performed using multiple tools. For example, starting or stopping an application can be performed in the Genesys Administrator as well as the Solution Control Interface.

Note that there are many option parameters. Type the name of the option in the filter "notice that it filters in real time". Some options can be set at both Application and Switch/DN levels. The option setting at the DN level takes precedence over the Application-level setting. See the Genesys SIP Server Deployment Guide for details. 

 


2. Creating SIP Switch in Genesys Administrator





 

By default, it does not have this permission. You must grant “Full Control” permission for the System account for the all DNs on the corresponding switch. It is done for all DNs at once by changing the permissions for the system account on the DN folder in the switch object. Or, you can start SIP Server under another account that has change permission on the necessary DNs.

With this full control access, the SIP Server Switch grants DNs like Extension to update their options like “contact” when a new SIP register message is received from end points moving to a new IP location

 

3. SIP Server Configuration in Genesys Administrator

Follow these steps to configure the SIP Server to monitor SIP Server Switch resources, such as SIP extensions/SIP end points registered to SIP Server. The SIP Server also monitors various route points and notifies URS whenever the call arrives on the Route Point.

Back to Top

4. Genesys Media Server Deployment

Follow these steps to configure a Media Server deployment.

Media Server platform consists of Resource Manager and Media Control Platform applications in the Genesys Voice Platform product suite. To deploy, Media Control Platform and optionally a Resource Manager are required to be installed. When installed, Resource Manager serves as the ingress point to Media services and provides a MCP resource as a media service to the network/calling side.

  1. Install and configure MCP (Media control Platform) using the Genesys Media Server Deployment Guide.
  2. Within the MCP application’s Connections tab, add connections to SNMP Master Agent, Message Server, and Reporting Server (optional).

    The connections to applications are added for the following reasons:

    Message Server - To ensure that component log information reaches the Log database and can be viewed in the Solution Control Interface (SCI)

    Reporting Server - (Optional) To ensure that these components detect the Reporting Server to which they are sending reporting data.

    SNMP Master Agent - To ensure that alarm and trap information is captured.

  3. Verify VoIP service DN of type=msml as specified in section Creating SIP Switch in Genesys Administrator to support SIP Server-Media Server MSML interactions to support treatments and conferencing capabilities.
  4. To play music on hold (MOH) and music treatments, verify the following options are set in MCP and SIP Server:

    MCP->msml-> play.basepath = file://$InstallationRoot$ (this is the installation folder of Media Server. After this is, it will automatically look for the music sub folder).

    “MOH” and music treatments are located in the “music” folder.

    The ‘announcement” folder should contain ‘prompt’ files with proper IDs to support. Used in the URS Routing Strategies as mentioned in chapter 0.

    SIP Server->TServer->msml-support=true

  5. Install and configure Resource Manager as per Genesys Media Server Deployment Guide.

    Note: If SIP Server and Resource Manager are on the same machine and within the Resource Manager application, then the default SIP listening port number should be increased by 100 so the Resource Manager listening port is set to 5160 and the SIP Server application listens on port 5060. Make the necessary port changes within Resource Manager’s sip, proxy, register, subscription, and monitor sections.

  6. Within the Resource Manager application’s Connections tab, add connections to SNMP Master Agent, Message Server, and Reporting Server (optional).

    The connections to applications are added for the following reasons:

    Message Server - To ensure that component log information reaches the Log database and can be viewed in the Solution Control Interface (SCI).

    Reporting Server - To ensure that these components detect the Reporting Server to which they are sending reporting data. (Optional).

    SNMP Master Agent - To ensure that alarm and trap information is captured.

  7. Within the Integrating Media Control Platform with the Resource Manager, click the Media Control Platform Application object. TheConfiguration tab appears.

    Click the Options tab, and use the View drop-down list to select Show options in groups...

    Select sip to find the routeset option.

    In the Value field, type the following:

    <sip:IP_RM:SIPPort_RM;lr>

     

    Where IP_RM is the IP address of the Resource Manager, and SIPPort_RM is the SIP port of the Resource Manager—typically, 5060.

     

    Note: You must include the angle brackets in the Value field in the sip.routeset and sip.securerouteset parameters.

     

    In the Value field of the securerouteset option, type the following:

    <sip:IP_RM:SIPSecurePort_RM;lr>

  8. G.729 media codec is not configured by default as a supported codec or as a codec that can be transcoded. This support can be enabled by adding “g729” as one of the values to the mpc.codec and mpc.transcoders space separated list.  The G.729 media codec was not provisioned in this Genesys deployment and is only mentioned here for completeness.

    Example:

    mpc.transcoders=PCM GSM G726 G729

    mpc.codec=g729 pcmu pcma g726 gsm h263 h263-1998 h264 telephone-event

     

    Alternately Media Server (specifically MCP component) can be configured to respond a multiple codec offer request with a single codec response. This feature support is available starting with MCP 8.1.4 release.

    This setting can be enabled by setting mpc.answerwithonecodec=1 (Default=0 – MCP responds to multiple codec offer with a multiple codec response list).

  9. This step is optional and is only required if multiple media control platform (MCP) instances are deployed and need to be controlled by Resource Manager for load balancing.

 

Log in to Genesys Administrator.

  1. On the Provisioning tab, click Voice Platform>Resource Groups.

  2. On the Details pane tool bar, click New.

  3. The Resource Group Wizard opens to the Welcome page.

  4. On the Resource Manager Selection page, add the Resource Manager Application object for which you want to create the group. On the Group Name and Type page: enter MCPGroup or any custom name without spaces. Select type as Media Control Platform.

  5. On the Tenant Assignments page, add the child tenant to which the Resource Group will be assigned.

  6. On the Group Properties page, enter the information as specified below for the Resource Group that you are configuring.

 

Note: For the Media Control Platform group, the Max.Conference Size and Max.Conference Count, and the Geo-location options are optional.

For a complete list of resource-group options and their descriptions, refer to  the Genesys Voice Platform User’s Guide.

In this step, you create a default IVR Profile that can be used to accept calls other than those specified in the dialing plans.

  1. Log in to Genesys Administrator.
  2. On the Provisioning tab, select Voice Platform>IVR Profiles.
  3. In the Tasks panel, click Define New IVR Profile. The IVR Profile Wizard opens to the Welcome page.
  4. On the Service Type page, enter the name of the default IVR Profile, IVR_App_Default.
  5. Select either Conference or Announcement from the drop-down list. (Only one service type per IVR Profile is supported.)
  6. If you selected Conference, on the Service Properties page, enter a conference ID number.
  7. If you selected Announcement, on the Service Properties page, enter the URL of the announcement, for example, http://webserver/hello.wav.
  8. Click Finish.
  9. In the gvp.general section of the Tenant’s Annex tab, set the default-application to this default IVR Profile name – IVR_App_Default.

This completes installation and configuration of Media Server. Make sure Resource Manager and MCP are started successfully.

Back to Top

5. Stat Server Configuration

This section explains configuration of Stat Server that connects with T-Servers/SIP Servers and maintains agent and/or extension status which is used by URS during call routing.

  1. Install and configure Stat Server per the Genesys Framework Stat Server Deployment Guide.
  2. Add connections to SIP Server, Message Server to perform real-time monitoring of the SIP agent status.

6. Universal Routing Configuration in Genesys Administrator

This section explains how to configure a Universal Routing Configuration (URS) to support execution of call routing on SIP Server.

  1. Install and configure Universal Routing Server per the Genesys Universal Routing Deployment Guide.
  2. Add connections to Message Server, Stat Server, and SIP Server.
  3. Add connection to SIP Server to monitor events received by SIP Server for various route points and extensions on the SIP Server Switch.
  4. Add connection to Stat Server to query Stat Server for routing calls to available and ready agents.


Use any of the strategies below to test your configuration. 

7. URS Routing Strategies

This section shows examples of five URS routing strategies used during testing. 


 



 

 

 

 

Sonus SBC 5xx0 Series Configuration

 

configure
#DSP Resources
set system mediaProfile compression 90 tone 10
commit
#Element Routing Priority profile
set profiles callRouting elementRoutingPriority TG_ERP entry localOperator 0 entityType trunkGroup
set profiles callRouting elementRoutingPriority TG_ERP entry nationalType 0 entityType trunkGroup
set profiles callRouting elementRoutingPriority TG_ERP entry internationalType 0 entityType trunkGroup
commit
#Ip Signaling Profiles
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes flags endToEndBye enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags dialogEventPackage enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags info enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags refer enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags statusCode3xx enable
set profiles signaling ipSignalingProfile GENESYS_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile GENESYS_IPSP egressIpAttributes flags disable2806Compliance enable
commit
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags disableMediaLockDown enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags reQueryPsxOnRegisterRefresh enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags endToEndBye enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags relayDataPathModeChangeFromOtherCallLeg enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags dialogEventPackage enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags info enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags options enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags refer enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags statusCode3xx enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags updateWithoutSdp enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags authcodeHeaders enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes sipHeadersAndParameters flags endToEndAck enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes flags sameCallIdForRequiredAuthorization enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes privacy transparency enable
set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags sendSdpInSubsequent18x enable
set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags suppress183WithoutSdp enable
set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags suppress183For3xxRedirectResponse enable sendSdpIn200OkIf18xReliable enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags noPortNumber5060 enable
commit
#Trusted Side Configuration
#IP Interface Group
set addressContext default ipInterfaceGroup TRUSTED ipInterface TRUSTED portName pkt0
set addressContext default ipInterfaceGroup TRUSTED ipInterface TRUSTED ipAddress 10.35.177.246
set addressContext default ipInterfaceGroup TRUSTED ipInterface TRUSTED prefix 26
set addressContext default ipInterfaceGroup TRUSTED ipInterface TRUSTED state enabled
commit
#IP Static Route
set addressContext default staticRoute 0.0.0.0 0 10.35.177.193 TRUSTED TRUSTED preference 100
commit
#zone
set addressContext default zone TRUSTED id 3
commit
#SIP signaling port
set addressContext default zone TRUSTED sipSigPort 3 ipInterfaceGroupName TRUSTED
set addressContext default zone TRUSTED sipSigPort 3 ipAddressV4 10.35.177.247
set addressContext default zone TRUSTED sipSigPort 3 portNumber 5060
set addressContext default zone TRUSTED sipSigPort 3 transportProtocolsAllowed sip-udp
set addressContext default zone TRUSTED sipSigPort 3 mode inService
set addressContext default zone TRUSTED sipSigPort 3 state enabled
commit
#IP Peer
set addressContext default zone TRUSTED ipPeer GENESYS ipAddress 10.35.176.111
set addressContext default zone TRUSTED ipPeer GENESYS ipPort 5060
set addressContext default zone TRUSTED ipPeer GENESYS authentication intChallengeResponse enabled
set addressContext default zone TRUSTED ipPeer GENESYS authentication incInternalCredentials enabled
commit
#SIP trunk group
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS policy callRouting elementRoutingPriority TG_ERP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS policy signaling ipSignalingProfile GENESYS_IPSP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS media mediaIpInterfaceGroupName TRUSTED
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS ingressIpPrefix 10.35.176.111 32
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS mode inService
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS state enabled
commit
#Untrusted Side Configuration
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED ceName LITTLE
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED portName pkt2
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED ipAddress 10.35.177.150
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED prefix 26
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED mode inService
set addressContext default ipInterfaceGroup UNTRUSTED ipInterface UNTRUSTED state enabled
commit
#IP Static Route
set addressContext default staticRoute 0.0.0.0 0 10.35.177.129 UNTRUSTED UNTRUSTED preference 100
commit
#zone
set addressContext default zone UNTRUSTED id 4
#SIP signaling port
set addressContext default zone UNTRUSTED sipSigPort 4 ipInterfaceGroupName UNTRUSTED
set addressContext default zone UNTRUSTED sipSigPort 4 ipAddressV4 10.35.177.151
set addressContext default zone UNTRUSTED sipSigPort 4 portNumber 5060
set addressContext default zone UNTRUSTED sipSigPort 4 transportProtocolsAllowed sip-udp,sip-tcp
set addressContext default zone UNTRUSTED sipSigPort 4 mode inService
set addressContext default zone UNTRUSTED sipSigPort 4 state enabled
commit
#IP Peer
set addressContext default zone TRUSTED ipPeer PSTN ipAddress 10.35.8.146
set addressContext default zone TRUSTED ipPeer PSTN ipPort 5060
set addressContext default zone TRUSTED ipPeer PSTN surrogateRegistration authUserName 2086041020
set addressContext default zone TRUSTED ipPeer PSTN surrogateRegistration regAuthPassword 123456
set addressContext default zone TRUSTED ipPeer PSTN surrogateRegistration sendCredentials challengeForAnyMessage
set addressContext default zone TRUSTED ipPeer PSTN authentication intChallengeResponse enabled
set addressContext default zone TRUSTED ipPeer PSTN authentication incInternalCredentials enabled
commit
#SIP trunk groups
#PSTN trunk
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN policy callRouting elementRoutingPriority TG_ERP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN policy signaling ipSignalingProfile GENESYS_IPSP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN media mediaIpInterfaceGroupName UNTRUSTED
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN ingressIpPrefix 10.35.8.146 32
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN mode inService
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-PSTN state enabled
commit
#RDN trunk
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN policy callRouting elementRoutingPriority TG_ERP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN policy signaling ipSignalingProfile ACCESS_IPSP
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN signaling registration requireRegistration required
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN media mediaIpInterfaceGroupName UNTRUSTED
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN ingressIpPrefix 0.0.0.0 0
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN mode inService
set addressContext default zone TRUSTED sipTrunkGroup TWO-WAY-SIP-GENESYS-RDN state enabled
commit
#Global Configuration
#Route Labels
set global callRouting routingLabel TO_TWO_WAY_GENESYS routingLabelRoute 0 routeType trunkGroup
set global callRouting routingLabel TO_TWO_WAY_GENESYS routingLabelRoute 0 trunkGroup TWO-WAY-SIP-GENESYS
set global callRouting routingLabel TO_TWO_WAY_GENESYS routingLabelRoute 0 ipPeer GENESYS
commit
set global callRouting routingLabel TO_TWO_WAY_PSTN routingLabelRoute 0 routeType trunkGroup
set global callRouting routingLabel TO_TWO_WAY_PSTN routingLabelRoute 0 trunkGroup TWO-WAY-SIP-PSTN
set global callRouting routingLabel TO_TWO_WAY_PSTN routingLabelRoute 0 ipPeer PSTN
commit
#Routes
set global callRouting route trunkGroup TWO-WAY-SIP-GENESYS LITTLE standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_TWO_WAY_PSTN
set global callRouting route trunkGroup TWO-WAY-SIP-GENESYS-RDN LITTLE standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_TWO_WAY_GENESYS
set global callRouting route trunkGroup TWO-WAY-SIP-PSTN LITTLE standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_TWO_WAY_GENESYS
commit
exit

 

Test Results

S.NoProcedureObservationResultComment
Inbound calls     
1Inbound Call to Agent released by caller Pass 
2Inbound Call to Agent released by agent Pass 
3Inbound Calls rejected Pass 
4Inbound Call abandoned Pass 
5Inbound Call to Route Point with Treatment Pass 
6Interruptible Treatment Pass 
7IVR (Collect Digit) Treatment Pass 
8Inbound Call routed by using 302 out of SIP Server signaling path Pass 
Outbound Calls    
91PCC Outbound Call from SIP Endpoint to external destination Pass 
103PCC Outbound Call to external destination Pass 
111PCC Outbound Call Abandoned Pass 
Music on hold    
12Caller is put on hold and retrieved by using RFC 2543 method Pass 
13T-Lib-Initiated Hold/Retrieve Call with MOH using RFC 3264 method Pass 
Call Transfers    
143PCC 2 Step Transfer to internal destination by using re-INVITE method Pass 
153PCC Alternate from consult call to main call Pass 
161PCC Unattended (Blind) transfer using REFER Pass 
171PCC Attended Transfer to external destination Pass 
183PCC Two Step Conference to external party Pass 
193PCC (same as 1PCC) Single-Step Transfer to another agent Pass 
203PCC Single Step Transfer to external destination using REFER Pass 
213PCC Single Step Transfer to internal busy destination using REFER Pass 
Other scenarios    
22Early Media for Inbound Call to Route Point with Treatment Pass 
23Early Media for Inbound Call with Early Media for Routed to Agent Pass 
24Inbound call routed outbound (Remote Agent) using INVITE without SDP Pass 
25Call Progress Detection Pass 
26Out of Service detection. Checking MGW live status Pass 
27SIP Authentication for outbound calls Not Supported 
28SIP Authentication for incoming calls Pass 
Remote Agent behind SBC scenarios    
29T-Lib-Initiated Answer/Hold/Retrieve Call for Remote SIP endpoint which supports the BroadSoft SIP Extension Event Package Pass 
303PCC Outbound Call from Remote SIP endpoint to external destination Pass 
313PCC 2 Step Transfer from Remote SIP endpoint to internal destination Pass 
321PCC Attended Transfer from Remote SIP endpoint to external destination Pass 
 

Conclusion

These Application Notes describe the configuration steps required for Sonus SBC 5XX0 to successfully interoperate with Genesys Voice Platform. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results.

 

 

Appendix A

The following new configurations are included in this section:

  1. SIP Server standard configuration
  2. DN Configuration
  3. EpiPhone configuration

1. SIP Server standard configuration

Information below displays how objects are configured by default.

SIP Server Application CME Options TServer section

SIP-hold-rfc3264=true

router-timeout=30

default-dn=

blind-transfer-enabled=true

resource-management-by-rm=true

msml-support=true

sip-enable-moh=true


2. DN Configuration 

 

Name

Number

Name in CME

CME Options TServer section

Comment

MGW-TRUNK

MGW-TRUNK

MGW-TRUNK

refer-enabled=true

contact=<TSE_CONTACT>

oos-check=10

oos-force=5

oosp-transfer-enabled=true

sip-replaces-mode=2

TSE

Ext-DN1

Ext-DN2

21001

21002

N/A

N/A

 

SIP-DN1

SIP-DN2

7101

7102

7101

7102

refer-enabled=false

ring-tone-on-make-call=false

make-call-rfc3725-flow=1

contact=*

 

SIP-RDN

7200

7200

refer-enabled=true

ring-tone-on-make-call=false

make-call-rfc3725-flow=1

contact=*

sip-cti-control=talk,hold

SIP endpoint which supports the BroadSoft SIP Extension Event Package.

SIP-UNKN

7777

N/A

N/A

 

RP

5000

5000

 

 

RP1

5001

5001

 

 

RP2

5002

5002

 

 

SVC_MSML

SVC_MSML

SVC_MSML

prefix=msml=

contact=<MS_CONTACT>

service-type=msml

subscription-id= Environment

MS

3. EpiPhone configuration


Content of configuration file esttt.conf:

 [TcCM]

site1 = UTE_HOME

connect-on-startup  = true

open-log-on-startup = false

log-to-file = epi-phone.log

#------------------------------------------------------------------------------

[UTE_HOME]

server = (host=<SIP_SERVER_HOST_IP>,port=<SIP_SERVR_TLIB_PORT>)

sip-register = false

dn1 = 7101,name="Alice",mkcall="7102"

dn2 = 7102,name="Bob"

dn3 = 7200,name="John"

dn4 = 5000,pool="shared"

dn5=5001,pool="shared",script="annc=(PROMPT=(\"1\"=(INTERRUPTABLE=1,ID=1)))"

dn6=5002,pool="shared",script="collect=(MAX_DIGITS=4,RESET_DIGITS=11,BACKSPACE_DIGITS=22,TOTAL_TIMEOUT=100)"