Ribbon SBC Q21 9.3 Genesys Voice Platform 8.1

Document Overview

This document provides a configuration guide for Ribbon Session Border Controller (SBC) Q21 series when connecting with the Genesys Voice Platform.

• For additional information on Genesys Voice Platform, please visit http://www.genesys.com/
• For additional information on Ribbon SBC, please visit http://ribboncommunication.com. 

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound call flows between Ribbon SBC Q21 and Genesys Voice Platform.

Audience

This technical document is intended for telecommunication engineers with the purpose of configuring the Ribbon SBC Q21 series  with the Genesys Voice Platform. This configuration requires the navigation of a third-party server and the Ribbon SBC Q21 Command Line Interface (CLI). Understanding the basic concepts for IP/Routing, SIP, RTP, and TLS are also required for completing the configuration and any necessary troubleshooting.

Requirements

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

Reference Configuration

The following reference configuration shows connectivity between third-party and Ribbon Q21.

Topology

Support

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

1. SIP Server Configuration

The following configuration shows how objects are configured by default:

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. SIP Trunk Configuration

Define a SIP trunk DN to represent all SIP calls arriving from the Ribbon SBC internal interface to the SIP Server. Configure the following options under the TServer section of the Trunk DNs.

3. SIP Extension  

Under SIPSwitch, define the DNs Extension types with the following options in the TServer section for the various SIP endpoints. These will register to the SIP Server.

4. Route Point

In SIPSwitch, create a DNs Routing Point type. This type should match the Request URI user part.

Route Point

5. IVR Profile

The following displays the content from the configuration file esttt.conf:

6. SIP DN Configuration

7. 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
dn8 = 2086041020,name="DN1",mkcall="2086041021"
dn9 = 2086041021,name="DN2"
dn15 = 2086041022,name="DN3"
dn25 = 2076041025,name="RDN"
dn10 = 1011,pool="shared"
dn11 = 1012,pool="shared",script="annc=(PROMPT=(\"1\"=(INTERRUPTABLE=1,ID=1)))"
dn12 = 1013,pool="shared",script="collect=(MAX_DIGITS=4,RESET_DIGITS=11,BACKSPACE_DIGITS=22,TOTAL_TIMEOUT=1000) play=music/on_hold"

SBC Q21 Configuration

#Basic Q21 Configuration
cli vnet add public
cli vnet edit public ifname eth2 gateway 10.35.141.65

cli vnet add private
cli vnet edit private ifname eth3 gateway 10.35.141.97

cli realm add public1
cli realm edit public1 rsa 10.35.141.94 mask 255.255.255.224 vnet public
cli realm edit public1 medpool 1 imr on emr on
cli realm edit public1 sipauth all

cli realm add private1
cli realm edit private1 rsa 10.35.141.124 mask 255.255.255.224 vnet private
cli realm edit private1 medpool 2 imr on emr on
cli realm edit private1 sipauth all

cli iedge add Genesys 0
cli iedge edit Genesys 0 type sipproxy realm private1 static 10.35.176.111 dtg togen newsrcdtg gentrunk
cli iedge edit Genesys 0 invitenosdp yes
cli iedge edit Genesys 0 uri 10.35.176.111

cli iedge add gen_trunk 0
cli iedge edit gen_trunk 0 realm public1 type sipgw static 10.35.8.146 newsrcdtg togen dtg gentrunk
cli iedge edit gen_trunk 0 removeVIDEOfromSDP enable
cli iedge edit gen_trunk 0 invitenosdp yes

cli realm edit public1 proxy_regid Genesys proxy_uport 0

#Message manipulation rule which changes m-line with video for SIP 200OK
cli fmm trigger add blankvideo-mline-t sdp-body
cli fmm trigger edit blankvideo-mline-t code is(200)
cli fmm trigger edit blankvideo-mline-t sdp.section.type is("media")
cli fmm trigger edit blankvideo-mline-t sdp.line.type is("m")

cli fmm action add modmline modify
cli fmm action edit modmline blankvideo-mline-t.sdp.line.value substitute(blankvideo-mline-t.sdp.line.value,"(video 0 RTP/AVP)","$1 34")

cli fmm rule add modvideo
cli fmm rule edit modvideo condition while(blankvideo-mline-t) actions modmline

cli fmm profile add modvidp
cli fmm profile edit modvidp rules modvideo