ion protocol sipv2 session target ipv4:172.X.X.1 dtmf-relay rtp-nte dial-peer voice 710 voip description SP to CUCM-2 with SIP destination-pattern 710 voice-class codec 1 continues 206 SIP Trunking Example 10-4 continued session protocol sipv2 session target ipv4:172.X.X.2 dtmf-relay rtp-nte dial-peer voice 720 voip description SP to CUCM 3 with H.323 destination-pattern 720 voice-class codec 1 session target ipv4:172.x.x.3 dtmf-relay rtp-nte dial-peer voice 730 voip description SP to PBX-4 with SIP destination-pattern 730 voice-class codec 1 session protocol sipv2 session target ipv4:172.X.X.4 dtmf-relay rtp-nte dial-peer voice 740 voip description SP to PBX-5 with H.323 destination-pattern 740 voice-class codec 1 session target ipv4:172.x.x.5 dtmf-relay rtp-nte Contact Center SIP Trunk Interconnect There are broadly two deployment options for PSTN access in Contact Center scenarios. These exist regardless of whether the PSTN access is via traditional Time Division Multiplexing (TDM) trunks or SIP trunks: In large sites, typically centralized deployments, the PSTN ingress gateways are usu- ally dedicated to the task and the Voice Extensible Markup Language (VoiceXML) browser capabilities that Cisco Voice Portal (CVP) uses to provide interactive voice response (IVR) services residing on dedicated VoiceXML gateways that are separate from the platform where the PSTN trunks terminate. In smaller sites, typically remote offices in a distributed network architecture, the same Cisco IOS platform serves as both PSTN gateway and VoiceXML browser gateway. When SIP trunking is used for PSTN access into a contact center, the same architectures apply. The CUBE terminating the SIP trunk might be dedicated to the task or might serve as the VoiceXML browser for CVP. This is shown in Figure 10-5. Chapter 10: Deployment Scenarios 207 Contact Center d I b Contact Center d I b Figure 10-5 Contact Center SIP Trunk Deployments Example 10-5 gives the configuration of a CUBE deployed in the integrated scenario shown in Figure 10-5, tha