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. :(
<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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>
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>
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____________________________________________________________ _____________
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All the best,Ben Cropley07539 366 905
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org