# Troubleshooting DTMF digits not detected in Webex Contact Center flows

Canonical: https://warmtransfer.net/knowledge/wxcc-dtmf-troubleshooting

Last verified: 2026-10-02

Webex Contact Center's PSTN media and signalling requirements list RFC2833 as the DTMF method for calls arriving at Webex Contact Center, and require those calls to arrive using the G.711 uLaw or G.711 aLaw codec[^35][^34]. Cisco states that calls are routed into Webex Contact Center using Webex Calling PSTN services, which all new Webex Contact Center subscriptions include[^36].

## How it works

### PSTN ingress requirements

Webex Contact Center lists RFC2833 as the DTMF method for calls arriving at Webex Contact Center[^35]. Calls must arrive using G.711 uLaw or G.711 aLaw[^34]. Any other codec, such as G.729, iLBC or Opus, must be transcoded before the call reaches Webex Contact Center media edge locations[^29]. On PSTN ingress, Webex Contact Center supports SIP Delayed Offer and does not support SIP Early Offer[^46]. Webex Contact Center does not support SIP Early Media on PSTN ingress[^27]. It specifies a 20 ms packetization timer in the SIP SDP for PSTN ingress[^37].

See also [Webex Contact Center voice channel and PSTN options](https://warmtransfer.net/knowledge/webex-cc-voice-channels).

### RFC 2833 and RFC 4733 telephone-events

RFC 2833 has been obsoleted by RFC 4733 and RFC 4734[^38]. RFC 4733 registers DTMF named events under the media type audio/telephone-event, which appears in SDP as telephone-event[^42]. RFC 2833 NTE payload types and attributes are negotiated between the two ends at call setup in the SDP body of the SIP messages[^30]. Under RFC 4733, if no events parameter is received, the sender should assume the receiver supports DTMF events 0-15 and no others[^39]. RFC 4733 assigns event codes 0-9 to digits 0-9, 10 to *, 11 to #, and 12-15 to A-D[^41].

The RTP timestamp marks the start of an event, and the several RTP packets of one event carry the same timestamp[^44]. RFC 4733 says the final packet of each event should be sent 3 times, so a capture normally shows 3 end packets for one key press[^40]. RFC 4733 notes that some implementations send telephone-events and encoded audio for the same instant, and recommends that gateways render only the telephone-event payload once it is received[^43].

## Gateway and SBC configuration

### Cisco Local Gateway on IOS XE

Cisco's Local Gateway on IOS XE configuration puts dtmf-relay rtp-nte on the inbound/outbound dial-peer that faces Webex Calling (dial-peer 100)[^24]. The same configuration also puts dtmf-relay rtp-nte on the SIP PSTN trunk dial-peer (dial-peer 200)[^21]. In Cisco's Local Gateway TDM PSTN example, the loop-back dial-peers 10, 11 and 12 each carry dtmf-relay rtp-nte[^22]. For an existing Unified CM environment, Cisco's Local Gateway configuration puts dtmf-relay rtp-nte on the UCM-facing dial-peers 300 and 400[^23].

### CUBE DTMF relay methods

On CUBE, the dtmf-relay keyword rtp-nte selects RFC 2833 RTP Named Telephone Events, which CUBE classes as an in-band method[^45]. CUBE also lists G.711 audio, meaning audible tones carried in the voice stream, as an in-band DTMF method that is enabled by default[^7]. CUBE's out-of-band DTMF relay methods are sip-notify (Cisco proprietary), sip-kpml (RFC 4730 KPML over SUBSCRIBE/NOTIFY) and sip-info (SIP INFO with application/dtmf-relay)[^9].

WarmTransfer's reading of the sources is that, because RFC2833 is the only DTMF method Webex Contact Center lists, a BYO PSTN path that delivers digits only as SIP INFO, KPML, unsolicited NOTIFY or in-band audio needs its gateway or SBC to convert them to RFC2833 before Webex Contact Center[^31][^35]. If an endpoint supports none of the DTMF relay mechanisms configured on CUBE, an MTP or transcoder is required[^8].

Cisco says audible in-band DTMF works reliably only if the call uses G.711 uLaw or aLaw, because compressing codecs distort the tones and can make them unrecognisable to the receiver[^18]. Cisco warns that transcoding a call from a low-bit-rate codec to G.711 most likely distorts in-band DTMF tones as well[^20].

Cisco devices use RTP payload types 96 and 97 for fax by default, and if a third-party device uses 96 or 97 for DTMF, Cisco recommends changing the fax payload type on both dial-peers or using asymmetric payload dtmf[^10].

## Diagnosing digits that are not detected

We infer that a Menu or Collect Digits activity that always takes its no-input path on a BYO PSTN entry point, even though callers press keys, matches carrier digits not arriving as RFC2833 telephone-events, and that the SDP and RTP should be checked first[^28].

On the signalling side, confirm that telephone-event is negotiated in the SDP, since NTE payload types and attributes are agreed at call setup in the SDP body[^30][^42]. Cisco's CUBE DTMF tech note uses debug ccsip messages to show SIP DTMF capability advertisement and debug voip rtp session named-event to show RFC 2833 digit transmission[^19]. CUBE gives show sip-ua calls, show sip-ua calls dtmf-relay sip-info, show sip-ua history dtmf-relay kpml and show sip-ua history dtmf-relay sip-notify for checking DTMF relay[^11].

In a packet capture, Cisco's tech note uses the Wireshark display filter rtpevent to isolate RFC 2833 DTMF packets, which Wireshark shows with protocol RTP EVENT[^33]. In Cisco's packet-capture example, RFC 2833 DTMF packets carry payload type telephone-event (96), a dynamic payload type in the range 96-127, while voice packets carry G.711 PCMA (8)[^32]. A well-formed key press normally shows 3 end packets[^40], with the packets of one event sharing the same RTP timestamp[^44].

See also [Building IVR menus and collecting digits in Webex Contact Center flows](https://warmtransfer.net/knowledge/wxcc-ivr-menu-setup).

## Diagnosing doubled digits

WarmTransfer's reading of the sources is that doubled digits in a Webex Contact Center flow are consistent with one key press reaching the media edge twice, for example as an RFC 2833 event plus leaked audible tone or a relayed out-of-band copy, rather than with a flow setting[^16].

If a CUBE call goes from RTP-NTE inbound to an out-of-band method outbound without dtmf-relay rtp-nte digit-drop on the inbound dial-peer, the same digit is sent both out-of-band and in-band, and the far end reads it as a duplicate[^6]. With digit-drop configured on the inbound leg, CUBE suppresses NTE packets and relays digits only through the out-of-band method configured on the outbound leg[^5]. When both unsolicited NOTIFY and NTE are negotiated on SIP, Cisco IOS chooses unsolicited NOTIFY and suppresses the in-band RFC 2833 NTE packets to avoid double tones[^47]. Because G.711 audio is an in-band method CUBE enables by default, audible tones can travel in the voice stream alongside telephone-events[^7][^43].

## Virtual agent and AI agent DTMF settings

### Virtual Agent V2 system defaults

For the Webex Contact Center Virtual Agent V2 activity, the system-defined DTMF termination symbol is #[^56]. The system-defined DTMF no-input timeout is 5 seconds[^55]. The system-defined DTMF inter-digit timeout is 3 seconds, after which the virtual agent moves on in the conversation[^54]. Cisco states that these Virtual Agent V2 DTMF settings are defined internally, do not appear in the user interface and cannot be changed, and that they apply to Dialogflow CX, Dialogflow ES and BYOVA[^53]. Cisco lists barge-in as enabled by default, letting the caller interrupt the Virtual Agent[^52].

### Flow Designer Virtual Agent activity

In the Flow Designer Virtual Agent activity's Advanced Settings, No-Input Timeout defaults to 5 seconds and can be 1 to 30 seconds, and it applies to voice or DTMF input[^50]. Max No-Input Attempts defaults to 3 and can be 0 to 9[^49]. Inter-digit Timeout defaults to 3 seconds and can be 0 to 30 seconds[^48]. The Terminator Symbol that ends caller input can be # or *[^51].

See also [Webex Contact Center Flow Designer architecture](https://warmtransfer.net/knowledge/webex-cc-flow-designer).

### Webex AI Agent Studio

For autonomous AI agents in Webex AI Agent Studio, the DTMF Digit Timeout between 2 digits defaults to 5 seconds and can be 2 to 10 seconds[^1]. The termination character that ends caller DTMF input can be # or *[^3]. Max length for caller DTMF input defaults to 16 characters and can be 8 to 32[^2].

### Dialogflow CX

Dialogflow CX processes DTMF only through telephony integrations, where the key presses are converted to text end-user input[^15]. Dialogflow CX DTMF can be enabled at agent level (Agent settings, Speech and IVR), at flow or page level, and at parameter level[^12]. If the pause between DTMF digits exceeds the interdigit timeout duration, Dialogflow aborts DTMF processing for that sequence[^14]. The finish digit (typically #) ends DTMF input immediately and is not included in the converted input[^13].

See also [Troubleshooting virtual agent and AI Agent handoffs in Webex Contact Center](https://warmtransfer.net/knowledge/wxcc-virtual-agent-troubleshooting).

## Applicability

Applies to: Cisco Webex Contact Center, Cisco Unified Border Element, Cisco Webex Calling Local Gateway, Cisco, Cisco Webex AI Agent Studio, and Google Dialogflow CX. Deployments: on-premises, multi-tenant, and standard. Sources checked 2026-10-02. The telephone-event packet behaviour described here is defined in RFC 4733, which obsoletes RFC 2833[^38][^42].

## What remains uncertain

The default settings and exit paths of the Collect Digits activity are not covered by the sources below. The current settings and exit path labels of the Menu activity are not covered by the sources below. Known Webex Contact Center DTMF defects in Cisco bug search are not covered by the sources below. Whether received digits are visible in the Webex Contact Center flow debugger or Analyzer is not covered by the sources below. How DTMF is handled on the Webex Calling PSTN path onward into Webex Contact Center is not covered by the sources below. No-input timeout values for Webex AI Agent Studio are not covered by the sources below. DTMF test cases for BYO PSTN acceptance testing are not covered by the sources below. Any DTMF differences between the Voice Services Platform and classic media are not covered by the sources below; see also [Webex Contact Center Voice Services Platform and Voice POP](https://warmtransfer.net/knowledge/webex-cc-voice-services-platform). How Cisco's statement that the Virtual Agent V2 DTMF settings cannot be changed fits with the configurable Advanced Settings of the Flow Designer Virtual Agent activity is not covered by the sources below.

## Sources

[^1]: For autonomous AI agents in Webex AI Agent Studio, the DTMF Digit Timeout between two digits defaults to 5 seconds and can be 2 to 10 seconds. Source: [Webex AI Agent Studio Administration guide](https://help.webex.com/en-us/article/ncs9r37/Webex-AI-Agent-Studio-Administration-guide), section 'Configure the DTMF settings', Digit Timeout. Checked 2026-10-02.
[^2]: In Webex AI Agent Studio DTMF settings, Max length for caller DTMF input defaults to 16 characters and can be 8 to 32. Source: [Webex AI Agent Studio Administration guide](https://help.webex.com/en-us/article/ncs9r37/Webex-AI-Agent-Studio-Administration-guide), section 'Configure the DTMF settings', Max length. Checked 2026-10-02.
[^3]: In Webex AI Agent Studio DTMF settings, the termination character that ends caller DTMF input can be # or *. Source: [Webex AI Agent Studio Administration guide](https://help.webex.com/en-us/article/ncs9r37/Webex-AI-Agent-Studio-Administration-guide), section 'Configure the DTMF settings', Termination Character. Checked 2026-10-02.
[^4]: Cisco's BYOVA sample code README lists DTMF in the conceptual contract both BYOVA transport variants implement, alongside session start, audio, events, transfer and end. Source: [CiscoDevNet/webex-contact-center-provider-sample-code](https://github.com/CiscoDevNet/webex-contact-center-provider-sample-code), bring-your-own/virtual-agent/README.md, integration variants description. Checked 2026-10-02.
[^5]: With digit-drop configured on the inbound leg, CUBE suppresses NTE packets and relays digits only through the out-of-band method configured on the outbound leg. Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', digit-drop description. Checked 2026-10-02.
[^6]: If a CUBE call goes from RTP-NTE inbound to an out-of-band method outbound without dtmf-relay rtp-nte digit-drop on the inbound dial-peer, the same digit is sent both out-of-band and in-band, and the far end reads it as a duplicate. Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', note on digit-drop for RTP-NTE to out-of-band interworking. Checked 2026-10-02.
[^7]: CUBE lists G.711 audio, meaning audible tones carried in the voice stream, as an in-band DTMF method that is enabled by default. Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', in-band methods list, G711 audio entry. Checked 2026-10-02.
[^8]: If an endpoint supports none of the DTMF relay mechanisms configured on CUBE, an MTP or transcoder is required. Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', interworking section. Checked 2026-10-02.
[^9]: CUBE's out-of-band DTMF relay methods are sip-notify (Cisco proprietary), sip-kpml (RFC 4730 KPML over SUBSCRIBE/NOTIFY) and sip-info (SIP INFO with application/dtmf-relay). Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', list of out-of-band DTMF relay methods. Checked 2026-10-02.
[^10]: Cisco devices use RTP payload types 96 and 97 for fax by default, and if a third-party device uses 96 or 97 for DTMF, Cisco recommends changing the fax payload type on both dial-peers or using asymmetric payload dtmf. Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', payload type note. Checked 2026-10-02.
[^11]: CUBE gives show sip-ua calls, show sip-ua calls dtmf-relay sip-info, show sip-ua history dtmf-relay kpml and show sip-ua history dtmf-relay sip-notify for checking DTMF relay. Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', verification section. Checked 2026-10-02.
[^12]: Dialogflow CX DTMF can be enabled at agent level (Agent settings, Speech and IVR), at flow or page level, and at parameter level. Source: [DTMF for telephony integrations](https://cloud.google.com/dialogflow/cx/docs/concept/dtmf), section on enabling DTMF. Checked 2026-10-02.
[^13]: In Dialogflow CX, the finish digit (typically #) ends DTMF input immediately and is not included in the converted input. Source: [DTMF for telephony integrations](https://cloud.google.com/dialogflow/cx/docs/concept/dtmf), section on DTMF settings, finish digit. Checked 2026-10-02.
[^14]: In Dialogflow CX, if the pause between DTMF digits exceeds the interdigit timeout duration, Dialogflow aborts DTMF processing for that sequence. Source: [DTMF for telephony integrations](https://cloud.google.com/dialogflow/cx/docs/concept/dtmf), section on DTMF settings, interdigit timeout duration. Checked 2026-10-02.
[^15]: Dialogflow CX processes DTMF only through telephony integrations, where the key presses are converted to text end-user input. Source: [DTMF for telephony integrations](https://cloud.google.com/dialogflow/cx/docs/concept/dtmf), page introduction. Checked 2026-10-02.
[^16]: Doubled digits in a Webex Contact Center flow are consistent with one key press reaching the media edge twice, for example as an RFC 2833 event plus leaked audible tone or a relayed out-of-band copy, rather than with a flow setting (inferred). Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', digit-drop note; read with RFC 4733 Section 2.5.2.2. Checked 2026-10-02.
[^17]: Flow Designer's voice activities include Collect Digits, Menu and Send Digits. Source: [Build and manage flows with Flow Designer](https://help.webex.com/en-us/article/nhovcy4/Build-and-manage-flows-with-Flow-Designer), activity headings list under voice activities. Checked 2026-10-02.
[^18]: Cisco says audible in-band DTMF works reliably only if the call uses G.711 uLaw or aLaw, because compressing codecs distort the tones and can make them unrecognisable to the receiver. Source: [Configure DTMF Relay on CUBE](https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/200412-DTMF-Relay-and-Interworking-on-CUBE.html), in-band DTMF section. Checked 2026-10-02.
[^19]: Cisco's CUBE DTMF tech note uses debug voip rtp session named-event to show RFC 2833 digit transmission and debug ccsip messages to show SIP DTMF capability advertisement. Source: [Configure DTMF Relay on CUBE](https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/200412-DTMF-Relay-and-Interworking-on-CUBE.html), troubleshooting and debug sections. Checked 2026-10-02.
[^20]: Cisco warns that transcoding a call from a low-bit-rate codec to G.711 most likely distorts in-band DTMF tones as well. Source: [Configure DTMF Relay on CUBE](https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/200412-DTMF-Relay-and-Interworking-on-CUBE.html), in-band DTMF section. Checked 2026-10-02.
[^21]: Cisco's Local Gateway configuration also puts dtmf-relay rtp-nte on the SIP PSTN trunk dial-peer (dial-peer 200). Source: [Configure Local Gateway on Cisco IOS XE for Webex Calling](https://help.webex.com/en-us/article/jr1i3r/Configure-Local-Gateway-on-Cisco-IOS-XE-for-Webex-Calling), section 'Configure Local Gateway with a SIP PSTN trunk', dial-peer voice 200 example. Checked 2026-10-02.
[^22]: In Cisco's Local Gateway TDM PSTN example, the loop-back dial-peers 10, 11 and 12 each carry dtmf-relay rtp-nte. Source: [Configure Local Gateway on Cisco IOS XE for Webex Calling](https://help.webex.com/en-us/article/jr1i3r/Configure-Local-Gateway-on-Cisco-IOS-XE-for-Webex-Calling), section 'Configure Local Gateway with a TDM PSTN trunk', loop-back dial-peers 10, 11, 12. Checked 2026-10-02.
[^23]: Cisco's Local Gateway configuration for an existing Unified CM environment puts dtmf-relay rtp-nte on the UCM-facing dial-peers 300 and 400. Source: [Configure Local Gateway on Cisco IOS XE for Webex Calling](https://help.webex.com/en-us/article/jr1i3r/Configure-Local-Gateway-on-Cisco-IOS-XE-for-Webex-Calling), section 'Configure Local Gateway with an existing Unified CM environment', dial-peers 300 and 400. Checked 2026-10-02.
[^24]: Cisco's Local Gateway on IOS XE configuration puts dtmf-relay rtp-nte on the inbound/outbound dial-peer that faces Webex Calling (dial-peer 100). Source: [Configure Local Gateway on Cisco IOS XE for Webex Calling](https://help.webex.com/en-us/article/jr1i3r/Configure-Local-Gateway-on-Cisco-IOS-XE-for-Webex-Calling), section 'Configure Webex Calling registration-based trunk', dial-peer voice 100 example. Checked 2026-10-02.
[^25]: Cisco's 2021 Webex Contact Center tech note shows the Flow Designer Menu activity with Entry Timeout and Unmatched Entry exit paths, which it loops back into the menu. Source: [Configure Counter in Menu Block for Webex Contact Center Flows](https://www.cisco.com/c/en/us/support/docs/contact-center/webex-contact-center/217379-configure-counter-in-menu-block-for-webe.html), configuration section, Menu block exit paths. Checked 2026-10-02.
[^26]: Cisco's tech note limits Menu retries with a counter: a Set Variable increments it ({{Timeout+1}}) on the timeout and unmatched paths, and a Condition ({{Timeout >= n}}) sends the call to Play Message and Disconnect Contact once the limit is reached. Source: [Configure Counter in Menu Block for Webex Contact Center Flows](https://www.cisco.com/c/en/us/support/docs/contact-center/webex-contact-center/217379-configure-counter-in-menu-block-for-webe.html), configuration steps, Set Variable and Condition blocks. Checked 2026-10-02.
[^27]: Webex Contact Center does not support SIP Early Media on PSTN ingress. Source: [Set up voice channels for Webex Contact Center](https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center), section 'PSTN Media and Signaling Requirements for Webex Contact Center', SIP Early Media row. Checked 2026-10-02.
[^28]: A Menu or Collect Digits activity that always takes its no-input path on a BYO PSTN entry point, even though callers press keys, matches carrier digits not arriving as RFC2833 telephone-events, and the SDP and RTP should be checked first (inferred). Source: [Set up voice channels for Webex Contact Center](https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center), section 'PSTN Media and Signaling Requirements for Webex Contact Center', DTMF row. Checked 2026-10-02.
[^29]: Webex Contact Center requires any codec other than G.711, such as G.729, iLBC or Opus, to be transcoded before the call reaches Webex Contact Center media edge locations. Source: [Set up voice channels for Webex Contact Center](https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center), section 'PSTN Media and Signaling Requirements for Webex Contact Center', codec transcoding note. Checked 2026-10-02.
[^30]: RFC 2833 NTE payload types and attributes are negotiated between the two ends at call setup in the SDP body of the SIP messages. Source: [Configure DTMF Relay on CUBE](https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/200412-DTMF-Relay-and-Interworking-on-CUBE.html), RTP-NTE section. Checked 2026-10-02.
[^31]: Because RFC2833 is the only DTMF method Webex Contact Center lists, a BYO PSTN path that delivers digits only as SIP INFO, KPML, unsolicited NOTIFY or in-band audio needs its gateway or SBC to convert them to RFC2833 before Webex Contact Center (inferred). Source: [Set up voice channels for Webex Contact Center](https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center), section 'PSTN Media and Signaling Requirements for Webex Contact Center', DTMF row, read with the CUBE DTMF Relay chapter. Checked 2026-10-02.
[^32]: In Cisco's packet-capture example, RFC 2833 DTMF packets carry payload type telephone-event (96), a dynamic payload type in the range 96-127, while voice packets carry G.711 PCMA (8). Source: [Identify DTMF Events in a Packet Capture](https://www.cisco.com/c/en/us/support/docs/voice/dual-tone-multifrequency-dtmf/223118-identify-dtmf-events-in-a-packet.html), section comparing normal RTP packets with DTMF packets. Checked 2026-10-02.
[^33]: Cisco's tech note uses the Wireshark display filter rtpevent to isolate RFC 2833 DTMF packets, which Wireshark shows with protocol RTP EVENT. Source: [Identify DTMF Events in a Packet Capture](https://www.cisco.com/c/en/us/support/docs/voice/dual-tone-multifrequency-dtmf/223118-identify-dtmf-events-in-a-packet.html), section on finding DTMF packets in Wireshark. Checked 2026-10-02.
[^34]: Webex Contact Center requires calls to arrive using the G.711 uLaw or G.711 aLaw codec. Source: [Set up voice channels for Webex Contact Center](https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center), section 'PSTN Media and Signaling Requirements for Webex Contact Center', Codec row. Checked 2026-10-02.
[^35]: Webex Contact Center's PSTN media and signalling requirements list RFC2833 as the DTMF method for calls arriving at Webex Contact Center. Source: [Set up voice channels for Webex Contact Center](https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center), section 'PSTN Media and Signaling Requirements for Webex Contact Center', DTMF row. Checked 2026-10-02.
[^36]: Cisco states that calls are routed into Webex Contact Center using Webex Calling PSTN services, which all new Webex Contact Center subscriptions include. Source: [Webex Contact Center Architecture](https://help.webex.com/en-us/article/utqcm7/Webex-Contact-Center-Architecture), section on voice ingress services. Checked 2026-10-02.
[^37]: Webex Contact Center specifies a 20 ms packetization timer in the SIP SDP for PSTN ingress. Source: [Set up voice channels for Webex Contact Center](https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center), section 'PSTN Media and Signaling Requirements for Webex Contact Center', Packetization Timer row. Checked 2026-10-02.
[^38]: RFC 2833 has been obsoleted by RFC 4733 and RFC 4734. Source: [Information on RFC 2833: RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals](https://www.rfc-editor.org/info/rfc2833), info page, 'Obsoleted by' field. Checked 2026-10-02.
[^39]: Under RFC 4733, if no events parameter is received, the sender should assume the receiver supports DTMF events 0-15 and no others. Source: [RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals](https://www.rfc-editor.org/rfc/rfc4733.html), Section 2.5.1.1. Checked 2026-10-02.
[^40]: RFC 4733 says the final packet of each event should be sent three times, so a capture normally shows three end packets for one key press. Source: [RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals](https://www.rfc-editor.org/rfc/rfc4733.html), Section 2.5.1.4. Checked 2026-10-02.
[^41]: RFC 4733 assigns event codes 0-9 to digits 0-9, 10 to *, 11 to #, and 12-15 to A-D. Source: [RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals](https://www.rfc-editor.org/rfc/rfc4733.html), Section 3.2, DTMF named events table. Checked 2026-10-02.
[^42]: RFC 4733 registers DTMF named events under the media type audio/telephone-event, which appears in SDP as telephone-event. Source: [RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals](https://www.rfc-editor.org/rfc/rfc4733.html), Section 2.1. Checked 2026-10-02.
[^43]: RFC 4733 notes that some implementations send telephone-events and encoded audio for the same instant, and recommends that gateways render only the telephone-event payload once it is received. Source: [RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals](https://www.rfc-editor.org/rfc/rfc4733.html), Section 2.5.2.2. Checked 2026-10-02.
[^44]: Under RFC 4733 the RTP timestamp marks the start of an event, and the several RTP packets of one event carry the same timestamp. Source: [RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals](https://www.rfc-editor.org/rfc/rfc4733.html), Section 2.2.1. Checked 2026-10-02.
[^45]: On CUBE, the dtmf-relay keyword rtp-nte selects RFC 2833 RTP Named Telephone Events, which CUBE classes as an in-band method. Source: [Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/dtmf-relay.html), chapter 'DTMF Relay', list of in-band DTMF relay methods. Checked 2026-10-02.
[^46]: Webex Contact Center supports SIP Delayed Offer and does not support SIP Early Offer on PSTN ingress. Source: [Set up voice channels for Webex Contact Center](https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center), section 'PSTN Media and Signaling Requirements for Webex Contact Center', SIP Early Offer and SIP Delay Offer rows. Checked 2026-10-02.
[^47]: When both unsolicited NOTIFY and NTE are negotiated on SIP, Cisco IOS chooses unsolicited NOTIFY and suppresses the in-band RFC 2833 NTE packets to avoid double tones. Source: [Configure DTMF Relay on CUBE](https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/200412-DTMF-Relay-and-Interworking-on-CUBE.html), SIP section on preference when multiple DTMF methods are negotiated. Checked 2026-10-02.
[^48]: In the Flow Designer Virtual Agent activity, Inter-digit Timeout defaults to 3 seconds and can be 0 to 30 seconds. Source: [Build and manage flows with Flow Designer](https://help.webex.com/en-us/article/nhovcy4/Build-and-manage-flows-with-Flow-Designer), activity 'Virtual Agent', Advanced Settings, Inter-digit Timeout. Checked 2026-10-02.
[^49]: In the Flow Designer Virtual Agent activity, Max No-Input Attempts defaults to 3 and can be 0 to 9. Source: [Build and manage flows with Flow Designer](https://help.webex.com/en-us/article/nhovcy4/Build-and-manage-flows-with-Flow-Designer), activity 'Virtual Agent', Advanced Settings, Max No-Input Attempts. Checked 2026-10-02.
[^50]: In the Flow Designer Virtual Agent activity's Advanced Settings, No-Input Timeout defaults to 5 seconds and can be 1 to 30 seconds; it applies to voice or DTMF input. Source: [Build and manage flows with Flow Designer](https://help.webex.com/en-us/article/nhovcy4/Build-and-manage-flows-with-Flow-Designer), activity 'Virtual Agent', Advanced Settings, No-Input Timeout. Checked 2026-10-02.
[^51]: In the Flow Designer Virtual Agent activity, the Terminator Symbol that ends caller input can be # or *. Source: [Build and manage flows with Flow Designer](https://help.webex.com/en-us/article/nhovcy4/Build-and-manage-flows-with-Flow-Designer), activity 'Virtual Agent', Advanced Settings, Terminator Symbol. Checked 2026-10-02.
[^52]: Cisco lists barge-in as enabled by default, letting the caller interrupt the Virtual Agent. Source: [Configure Virtual Agent-Voice in Webex Contact Center](https://help.webex.com/en-us/article/n6gaghu/Configure-Virtual-Agent-Voice-in-Webex-Contact-Center), section listing default system-level settings. Checked 2026-10-02.
[^53]: Cisco states that the Virtual Agent V2 DTMF settings are defined internally, do not appear in the user interface and cannot be changed, and that they apply to Dialogflow CX, Dialogflow ES and BYOVA. Source: [Configure Virtual Agent-Voice in Webex Contact Center](https://help.webex.com/en-us/article/n6gaghu/Configure-Virtual-Agent-Voice-in-Webex-Contact-Center), section listing default system-level DTMF settings, introductory sentence. Checked 2026-10-02.
[^54]: For the Virtual Agent V2 activity, the system-defined DTMF inter-digit timeout is 3 seconds, after which the virtual agent moves on in the conversation. Source: [Configure Virtual Agent-Voice in Webex Contact Center](https://help.webex.com/en-us/article/n6gaghu/Configure-Virtual-Agent-Voice-in-Webex-Contact-Center), section listing default system-level DTMF settings. Checked 2026-10-02.
[^55]: For the Virtual Agent V2 activity, the system-defined DTMF no-input timeout is 5 seconds. Source: [Configure Virtual Agent-Voice in Webex Contact Center](https://help.webex.com/en-us/article/n6gaghu/Configure-Virtual-Agent-Voice-in-Webex-Contact-Center), section listing default system-level DTMF settings. Checked 2026-10-02.
[^56]: For the Webex Contact Center Virtual Agent V2 activity, the system-defined DTMF termination symbol is #. Source: [Configure Virtual Agent-Voice in Webex Contact Center](https://help.webex.com/en-us/article/n6gaghu/Configure-Virtual-Agent-Voice-in-Webex-Contact-Center), section listing default system-level DTMF settings. Checked 2026-10-02.