) between the
Asterisk server and VoIP Gateway.
Any helpful hints on where to look?
-Matt
--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760
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ly introduced b2bua module is brought to bear on
> that
> somewhat, but classically, OpenSIPS is a proxy.
>
> -- Alex
>
> Matthew S. Crocker wrote:
>
> > Hello,
> >
> > I'm brand new to OpenSIPS, just going through the make process now.
>
>
own Acme for
> that.
> I've had great luck with Edgewater Networks' "Edgemarc" devices, for
> example. That's just one. There are many.
>
>
> - Jeff
>
>
>
> On 8/20/09 2:49 PM, "Matthew S. Crocker"
> wrote:
>
> >
&
ing an RTP stream with a
mediaproxy/recorder would be a bonus.
-Matt
--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760
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know I can route the call through Asterisk and have it do a quiet answer and
listen for the modem sounds. I'd like to avoid using Asterisk for all RTP
traffic and only use it for the fax gateway traffic (i.e. once it has been
determined to be a fax Asterisk steps in and handled the T38 -
to something able to handle
> the
> fax - of course, the b2b will bridge the existing leg (towards PSTN)
> and
> the new leg.
>
> Regards,
> Bogdan
>
>
> Matthew S. Crocker wrote:
> > Can OpenSIPS make routing decisions based on the SDP information in
> an I
t see the need of Asterisk - maybe with some changes, the
> b2b
> >> module will be able to handle this - see my prev email.
> >>
> >> Regards,
> >> Bogdan
> >>
> >>
> >>> -Brett
> >>>
> >>>
> >
e with.. anything? It doesn't need to be an AVP.. just
> looking for any sortable data structure.
>
> Thanks,
> Brett
>
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