300-085 · Question #26
A user has reported that keys pressed when accessing Cisco Unity Express are not being acknowledged by the system. This configuration is found on the voice-enabled IOS router: dial-peer voice 6800…
The correct answer is A. dtmf-relay sip-notify. When a SIP-based dial-peer is used to connect to Cisco Unity Express and DTMF keystrokes are not recognized, the dial-peer requires the dtmf-relay sip-notify command to transmit tones via SIP NOTIFY messages.
Question
A user has reported that keys pressed when accessing Cisco Unity Express are not being acknowledged by the system. This configuration is found on the voice-enabled IOS router:
dial-peer voice 6800 voip destination-pattern 5... session protocol sipv2 session target ipv4:10.3.6.127 codec g711ulaw no vad Which configuration command is needed to complete the configuration and solve the problem?
Options
- Adtmf-relay sip-notify
- Bdtmf-relay rtp-nte
- Cdtmf-relay cisco-rtp
- Ddtmf-relay h245-signal
How the community answered
(30 responses)- A73% (22)
- B10% (3)
- C3% (1)
- D13% (4)
Why each option
When a SIP-based dial-peer is used to connect to Cisco Unity Express and DTMF keystrokes are not recognized, the dial-peer requires the dtmf-relay sip-notify command to transmit tones via SIP NOTIFY messages.
Because the dial-peer uses session protocol sipv2, DTMF digits must be relayed using SIP NOTIFY messages per RFC 2833 signaling over SIP. The dtmf-relay sip-notify command instructs the router to encode each DTMF digit as an out-of-band SIP NOTIFY event, which Cisco Unity Express understands and processes correctly - without this, the codec carries audio in-band but DTMF events are lost.
dtmf-relay rtp-nte sends DTMF via Named Telephone Events in RTP packets (RFC 2833), but this method requires mutual negotiation in SDP and is not the standard method used for CUE SIP integrations in this topology.
dtmf-relay cisco-rtp is a Cisco-proprietary RTP-based DTMF method and is incompatible with SIP session protocol dial-peers.
dtmf-relay h245-signal is an H.323 signaling method and cannot be applied to a SIP (sipv2) dial-peer.
Concept tested: DTMF relay configuration for SIP dial-peers connecting to CUE
Source: https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/1842-dtmf-voice-gateways.html
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