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@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>_________________________________________________________________________