[Freeswitch-users] Modems

[Rupa Schomaker]

http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo

On Tue, Aug 10, 2010 at 2:17 AM, Ken Gillett <ken at ukgb.net> wrote:

> Well I did try a Linksys/Cisco/Sipura 3102 but it's a configuration
> nightmare. The problem mainly is as I said, I cannot figure how VOIP -> PSTN
> bridging can be achieved whatever the gateway device.
>
> A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the
> call. From that piece of data it gets the SIP domain and the ID of the user.
> So how is the gateway able to be inserted in this process. The SIP client
> has no knowledge of this device and no way to include that third piece of
> data into its process.
>
> Obviously such gateways do exist, but I do not yet understand how the
> process works. Some advice on the basic process flow would therefore assist
> me to set up what I need irrespective of what devices I am using. Sorry to
> be so ignorant about this, but anyone able and willing to help with an
> explanation?
>
>
> On 10 Aug 2010, at 04:23, Rupa Schomaker wrote:
>
> > I'm pretty sure the zoom does not support sip originated calls to the FXO
> port.  It's FXO port is strictly used as failover or selectable via dialplan
> when the call originates from the FXS port (eg: dial 9 first to get FXO).
> >
> > Try:
> >
> > cisco 3102
> > audiocodes
> > grandstream
> >
> > for atas that support full FXS/FXO operation.
> >
> > On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett <ken at ukgb.net> wrote:
> > I am actually trying this out with a Zoom 5801 which with an FXO and FXS
> port and the ability to bridge in both directions can apparently do what I
> require, but I cannot get my head around what I am even trying to get it to
> do. And this is before I've even thought about bringing FreeSwitch into the
> equation.
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
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-- 
-Rupa
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