[Freeswitch-users] start_dtmf_generate is not working with opus code

[Andrew Paul]

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
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