[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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