Use Webex Webinars to schedule webinars starting April 1st, 2023īeginning April 1, 2023, you can no longer use Webex Events (classic) to schedule webinars, you must use Webex Webinars instead. When SIP is used to connect a Cisco SRST system to a remote SIP-PSTN voice gateway that goes through the PSTN to a voice-mail or IVR application.The 43.4 update has important announcements, which you can review in the Announcements for the Slow Channel of Cisco Webex Meetings.When SIP is used to connect a Cisco SRST system to a remote SIP-based IVR or voice-mail application, such as Cisco Unity.The SIP DTMF relay method is needed in the following situations: You select this method in the SIP VoIP dial peer using the dtmf-relay rtp-nte command. To enable SCCP phones to send digit information to remote SIP-based IVR and voice-mail applications, Cisco SRST 3.2 and later versions provide conversion from the out-of-band SCCP digit indication to the SIP standard for DTMF relay, which is RFC 2833. DTMF Relay Using SIP Notify (Nonstandard)Ĭisco Skinny Client Control Protocol (SCCP) phones, such as those used with Cisco SRST systems, provide only out-of-band DTMF digit indications. DTMF Relay for SIP Applications and Voice MailĭTMF Relay for SIP Applications and Voice MailĭTMF relay for SIP applications can be used in two voice-mail situations:.Note The Cisco IOS Voice Configuration Library includes a standard library preface, glossary, and feature and troubleshooting documents and is located at.
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