[Freeswitch-users] start_dtmf_generate not working with opus

[Michael Jerris]

before you go any further i’d check to see if this is a bug we’ve already fixed by testing newer code instead of an older release.

> On Jan 23, 2018, at 7:24 AM, Varghese Paul <varghesepaul87 at gmail.com> wrote:
> 
> 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
> 
> Varghese Paul 
> 
> <opus.conf.xml>_________________________________________________________________________