[asterisk-users] No DTMF on Sip Trunk?

[Eric Wieling]

For ABE support you really should contact Digium.  BTW, there is no such 
thing as a "sip trunk".  It's a sip peer or connection or account.

Noah Miller wrote:
> Hi Jared -
> 
>>  > For the first time, I'm setting up SIP trunking between two asterisk
>>  > boxes.  The calls themselves work fine, but I'm not able to get DTMF
>>  > working.
>>
>>  If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it
>>  appears that you are), you'll need to set "rfc2833compensate=yes" in the
>>  peer or friend section of sip.conf on the Asterisk 1.4 box.
> 
> Unfortunately, this didn't work.  Maybe rfc2833compensate isn't
> available in ABE?
> 
> I think this may require inband signalling anyway, as we'll require
> non-sip (zap) devices to be able to use these sip trunks and enter
> DTMF.
> 
> Any other ideas?
> 
> Thanks!
> Noah
> 
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