pstn tones not being passed

[kieranmullen]

I am able to make outgoing valls using my asterisk/freepbx setup.. however it is not passing pstn tones correct. I had my wife call the same system through the landline and it works fine.

Any ideas?
thank you
Kieran

[vtjosh]

You need to specify inband dtmf in your configs.

You might want to add 'relaxdtmf=yes' under the general context of sip.conf.

Another thing I've seen work well is using SIPDtmfMode in extensions.conf before any call logic dealing with dtmf.

Example:
exten => s,n,SIPDtmfMode(inband)

[kieranmullen]

is SIP INFO acceptable?

[DracoFelis]

is SIP INFO acceptable?
In my experience (with a stand alone LinkSys/Sipura adapter) no.

For some reason, "In Band" seems to be the only DTMF that ViaTalk servers reliably accept. In fact, on my SPA-3000, not even "Auto" would work. If I wanted DTMF with VT, I had to force "InBand". Sadly on my adapter forcing "InBand" forces it for all providers. However, since you are using Asterisk, you should be able to force InBand DTMF for just VT (and leave other providers at whatever DTMF setting you normally prefer).

[IronHelix]

I think you can submit a ticket and have them switch you to RFC2833. Failing that though, you should be able to set dtmfmode=inband for VT and then rfc2833 or info for the other accounts, * should convert the DTMF to inband as it passes through.