Telephony integration is the means by which the telephony server and the Messaging system exchange control information about calls. Telephony integration is achieved when a call is presented to a voice port and information about the call is supplied to the Messaging system. This information includes the nature of the call, called party information, and the calling party number.
Messaging supports only Session Initiation Protocol (SIP) integration. SIP is an important technology for establishing real-time audio and multimedia calls in a converged IP network environment.
Using SIP, the telephony server and the application servers are connected to the local area network (LAN). All exchange of information, such as call information, signaling information, and voice data, happens over voice channels through the LAN.
You can connect the Messaging system to the telephony server, using the SES proxy or Session Manager, through SIP trunks. The SIP trunks are administered as part of a trunk group on the telephony server. You can also directly connect the Messaging system to the telephony server without using the SES proxy or Avaya Session Manager. Consult your ATAC or Sales Engineer representative for these types of integration.
Messaging interoperates with AudioCodes gateway to work with telephony servers and integrations that are not supported. You need one AudioCodes gateway for each application server. If you need more than one gateway to meet the port capacity, you will need extra application servers. For the Messaging configurations with a cluster of application servers, you can configure a single AudioCodes gateway for load balancing of the traffic across multiple application servers.