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?
Please help
HA
PSTN gateways usually send INVITE messages with the request URI containing sip:<called_number>@proxy, so all you have to do is create an alias <called_number>@proxy -> <real_user>@proxy
Jan.
On 24-05 22:38, 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?
Please help
HA
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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
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@sdf.com] Enviado el: Martes, 25 de Mayo de 2004 02:02 p.m. Para: Humberto Atristain CC: serusers@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
-
Humberto Atristain -
Jan Janak -
Tom