Application Note

FS; Reviewed:
SPOC 09/07/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
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In general, a SIP telephone originates a call by sending a call request (SIP INVITE message) to
Session Manager, which then routes the call over a SIP trunk to Communication Manager for
origination services. If the call is destined for another local SIP telephone, Communication
Manager routes the call back over the SIP trunk to Session Manager for delivery to the
destination SIP telephone. If the call is destined for an H.323 or Digital telephone, then
Communication Manager terminates the call directly.
These application notes assume that Communication Manager and Session Manager are already
installed and basic configuration steps have been performed. Only steps relevant to SIP
telephone calling features will be described in this document. For further details on configuration
steps not covered in this document, consult the appropriate document in Section 10.
3. Equipment and Software Validated
The following equipment and software/firmware were used for the sample configuration
provided.
Equipment Software/Firmware
Avaya S8800 Server with G450 Media
Gateway
Avaya Aura™ Communication Manager 6.0
Service Pack 0 (Load 345, Update 18246)
Avaya S8800 Server
Avaya Aura™ Session Manager 6.0, Load
600020
Avaya Aura™ System Manager 6.0, Load
600020
Avaya 9630 IP Telephone (SIP) 2.6.0.0
Avaya 9630 IP Telephone (H.323) 3.1.1
Avaya 1603 IP Telephone (SIP) R1.0.0
Avaya 6408D+ Digital Telephone -
Modular Messaging Storage Server 5.2, Service Pack 3 Patch 1
Modular Messaging Application Server 5.2, Service Pack 3 Patch 1
Avaya 1100-series IP deskphones (SIP) 03.02.15.05
Avaya 1200-series IP deskphones (SIP) 03.02.15.05
Table 1: Equipment and Software/Firmware