Sip trunk and busydetect=yes

Hi community,
I have asterisk with sip trunk connected to sip provider.
One gsm provider XYZ wonder me. The issue is when A call B (subscriber gsm provider XYZ) and B press button reject call, this provider send to my Asterisk in g711 voice record of Busy not any SIP signaling message “Bysy 17” and channel stay open for long time…
I am reading this topic for busy detect in fxo line this is possible

And I tried in my sip trunk settings sip.conf
in file indications.conf
set local country tones
But asterisk didn’t detect busy inband.

Is possible sip trunk detect busy in RTP stream and Hang up line?

What’s a SIP trunk. Neither the SIP RFC nor Asterisk have any concept of one.

Asterisk doesn’t detect inband signalling on anything but DAHDI. Anyone that signals failures as early media on VoIP is not intending a machine to answer.

My SIP trunk is SIP UDP, Asterisk IP <===> SIP provider IP
signaling port 5060
media 10000-20000
From my first post link I see that FXO DAHDI is capable to hear RTP Busy record and hang up line.
@david551 do you mean that my trunk is unable to detect inband signalling that why my test is failure not my configuration mistake?
Probably I can force my asterisk don’t use early media and B side will return to me Busy as a message I am wondering…

It’s only a trunk because you call it that. Asterisk does not monitor RTP streams from any SIP endpoint for in band signalling.

I don’t know any way of effectively preventing the callee using early media. I think the problem is that the service is only intended for human use.

I will try to fix this with service provider
Thank you very much

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