SIP (AST) user 1011111 and SIP (AST) user 1011112 are both based on the same Session Manager with access to a CM Evolution server. Given the following actions: 1. Session Manager searches for the…

Question

Options

• A2,1,3,5,4
• B2,3,5,4, 1
• C2,3,5,1,4
• D2, 3, 1,4, 5

• A
  5% (2)
• B
  3% (1)
• C
  77% (30)
• D
  15% (6)

Explanation

Option C (2→3→5→1→4) reflects the correct Avaya SIP call flow because the sequence follows the logical handoff chain: the INVITE is sent first (2), Session Manager must authenticate the caller and invoke sequenced applications before doing anything else (3), Communication Manager then performs full call-model processing and hands control back to Session Manager (5), only then does Session Manager look up and route to the called party (1), and finally the endpoints negotiate codecs and media flows (4) - you cannot have RTP until both legs are established.

Why the distractors fail:

• A (2,1,3,5,4) puts finding the called party (1) before authenticating the caller (3), which is impossible - SM must verify the originating user before doing anything with the destination.
• B (2,3,5,4,1) places media negotiation (4) before the called party is ever reached (1) - media cannot flow if the call hasn't been delivered yet.
• D (2,3,1,4,5) bypasses CM processing (5) before routing to the called party (1), skipping the full call-model loop that CM must complete first.

Memory tip: Think of the flow as "Caller → Authenticate → CM Round-trip → Called → Media" - SM always handles the originating side completely (including the CM detour) before it ever touches the terminating side.

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