[asterisk-users] No DTMF on Sip Trunk?

Noah Miller noahisaacmiller at gmail.com
Thu Apr 24 16:01:18 CDT 2008


> 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.

<shrug> Semantics.  IAX connections between two asterisk boxes are
often called IAX trunks.  This is the same thing in SIP
flavor.</shrug>

Also, no offense against Digium support, but the list actually
responds more quickly at this point.  I think the Digium support staff
are in a situation of high demand and short staffing.


- Noah


>
>
>
>  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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