Hi all,

I am facing a issue with start_dtmf_generate when i use opus codec and SIP INFO dtmf.

Caller ( opus/dtmf ) ------------------------ Freeswitch ----------------------------- Calleee ( ulaw/inband)

Caller is using jssip webrtc and dtmf is in INFO. For inband dtmf generation i am using start_dtmf_generate in caller channel.

The setup is working when i use ulaw or ilbc or alaw codec. When i use webrtc with OPUS codec i am seeing the inband dtmf generation is not working. Webrtc with ulaw codec and dtmf INFO is working fine.

Version : FreeSWITCH Version 1.6.17~64bit ( 64bit)

I am attaching the opus.conf.xml for the reference.

Any idea, why start_dtmf_generate is not working with opus codec ?. Do i need to set any setting for opus ?.

Regards