[Serusers] PSTN to SIP in analog gateways
Humberto Atristain
hatristain at megahospedaje.com
Wed May 26 03:39:19 CEST 2004
Tom, they are Welltech FXO gateways (analog) and in fact they support SIP
signaling.
They work in 2 forms
Peer-to-peer and registered with sip users on each port.
In fact, they have an option of a small IVR where I dial the sip destination
and the call is connected,
But my trouble is that directly connected to pstn I need to automatically
formward a call to a VOIP user. Eg. Incomming call by Port 3 (FXO) to sip
user joe without any IVR dialing or second Dial tone.
Thanks
HA
-----Mensaje original-----
De: Tom [mailto:tom at sdf.com]
Enviado el: Martes, 25 de Mayo de 2004 02:02 p.m.
Para: Humberto Atristain
CC: serusers at lists.iptel.org
Asunto: Re: [Serusers] PSTN to SIP in analog gateways
On Mon, 24 May 2004, Humberto Atristain wrote:
> Dear Sirs
>
> I have a very important Issue (for me) , how do I automatically send a
call
> from an analog gateway to a defined IP user
>
> For me the PSTN termination is easy but what about inbound calling, I need
> no IVR I need only to send to a determined sip user.
>
> The gateway has 2 options
>
> Peer-to-peer (I can point to calls toa n IP and a e164 number) gonder if
> this can be a sip user)
>
> And
>
> Proxy registration
>
> What I need to use?
What kind of gateway is it? I'm assuming the gateway supports SIP
signalling?
Which option allows you to configure a SIP Proxy server hostname or IP?
Some gateways just answer incoming calls with an second dialtone, and
allow you to dial more digits (two stage dialing). Such gateways usually
have a "hotline" or "call forward" option to force all incoming calls to a
particular SIP destination.
Tom
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