SIP trunk Flash Hook configuration

Hi

I have a Voip device that is connected as SIP extension (pjsip) and a SIP trunk (pjsip) that is talking to an FXO GW.

When I click on the recall button on the Voip device it send telephony event DTMF 16 to the PBX, however when the PBX send it to the GW via the SIP trunk it sending the Flash Hook signal with SIP INFO (signal=!).
I would like to change the configuration on the PBX to send the Flash hook, as received from the phone (RFC2833/4733). I have set the config of the DTMF on the trunk to RFC 4733 (under pjsip settings; Advanced), but this does not make a difference.

I am using FreePBX 16.0.33

Could anyone please suggest where am I going wrong or where/how this can be changed.

Thanks

Is the gateway offering telephony events, and including 16 in the valid range?

Very good point , which I should have checked, but just checked and it does

In the 200OK (from the GW)-

Media Attribute (a): fmtp:101 0-15,16

Flash over RFC4733 is not supported in PJSIP currently.

Shame

btw, could I configure a different SIP trunk type that can support it?

Thanks

No, neither chan_sip or chan_pjsip support it and there is no other SIP channel driver in Asterisk.

Thanks

I can’t say I have tried it but theoretically, because DualToneMultiFrequency fits ‘inband’ on g711 and the decode is a simple one of 4x4 matrix. What gets decoded when DTMF is forced to ‘inband’ and you send a simple 'D" (1633 + 941 , event 16) and don’t rely on RFC4733 ?

A little more ‘off the reservation’

https://www.cloudacm.com/?p=3147

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