Try with <action application="set" data="pass_rfc2833=true"/> before bridge On Wed, 12 Oct 2016, 07:31 Yu Boot, <yu@yu-boot.ru> wrote:
I find this:
<action application="start_dtmf_generate"/>
It works like I've wanted, but any call that is routed via this extension has audio issues like stuttering, dropouts etc. :(
11.10.2016 11:32, Benjamin Cropley пишет:
<action application="set" data="dtmf_type=rfc2833"/>
good luck :)
On Tue, Oct 11, 2016 at 9:16 AM, Yu Boot <yu@yu-boot.ru> wrote:
Yes. But unfortunately a remote side can't work with SIP INFO either. How to force FS to generate inband DTMF to B-leg?
11.10.2016 10:32, Dragos Oancea пишет:
By default FS should pass RFC2833 without any change. Perhaps you don't have a "telephone-event" payload in the SDP answer from the other side (B-leg) ?
Dragos
On 11/10/2016 08:17, Yu Boot wrote:
FS receives DTMF tones from A-leg in RFC2833 mode, but generates SIP INFO to B-leg instead of RFC2833. How to fix this?
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