[Freeswitch-users] Modems

[Ken Gillett]

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

_/_/_/_/_/_/_/_/