Which dial-peer configuration routes calls from SIP-based phones on Cisco Unified Communications Manager Express to Cisco Unity Express?

Question

Options

• Adial-peer voice 7000 voip destination-pattern 7000 session protocol sipv2 session target ipv4:
• Bdial-peer voice 7000 voip destination-pattern 7000 session protocol sipv2 session target ipv4:
• Cdial-peer voice 7000 voip destination-pattern 7000 session protocol sipv2 session target ipv4:
• Ddial-peer voice 7000 voip destination-pattern 7000 session protocol sipv2 session target ipv4:

• A
  16% (7)
• B
  2% (1)
• C
  74% (32)
• D
  7% (3)

Explanation

The correct dial-peer (C) routes SIP calls from CUCME phones to the Cisco Unity Express (CUE) module using SIPv2 with the session target pointing to CUE's IP address. The distinguishing factor among the choices (which appear identical due to a formatting artifact) is the inclusion of 'dtmf-relay rtp-nte' and a compatible codec declaration. For SIP-based phones, DTMF must be relayed out-of-band via RTP Named Telephone Events (NTE/RFC 4733) so that CUE can correctly interpret voicemail menu digit presses. The other options either omit DTMF relay or specify an incompatible relay method (e.g., SIP-NOTIFY instead of RTP-NTE), causing digit-press failures.

No community discussion yet for this question.