[Freeswitch-users] Prevent conversion from rfc2833 to SIP INFO

mayamatakeshi mayamatakeshi at gmail.com
Tue Feb 9 00:17:48 UTC 2021

On Mon, Feb 8, 2021 at 5:50 PM mayamatakeshi <mayamatakeshi at gmail.com>

> Hi,
> I have a gateway inviting FS with an SDP without payload telephone-event
> (rfc2833).
> This channelA is bridged to another channelB that does support rfc2833.
> I noticed that in this case FS converts the rfc28333 digit from channelB
> to SIP INFO to channelA.
> The gateway doesn't support SIP INFO, but does support rfc2833 even if it
> doesn't advertise it (at least it does support it for outgoing calls from
> what was reported to me).
> I tried to use this before the bridge:
>        <action application="set" data="pass_rfc2833=true"/>
> but behavior didn't change.
> Is there anything else I can try?
Thinking again, since the telephone-event was not negotiated at channelA
side, of course FS cannot relay the RFC2833 packet.
So I am thinking in detect rfc2833 digits at channelB and send them as
inband tones at channelA using:
  send_dtmf <dtmf digits>[@<tone_duration>]
But I'm worried with the possibility of overlapping tones at channelA.
Does anyone know if send_dtmf has queueing behavior, meaning if I call it
multiple times in rapid succession, one operation will not interrupt the
previous one?
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