Gateway Configuration
It seems there are two ways to configure an spa3102's fxo port w/ pbx's, you can set the dial plan to <phone_#_of_pstn>@<freeswitch_ip> or <ext_to_dial>@<freeswitch_ip>.
From fs's perspective, what exactly is the difference here? Are there any significant differences between the two methods? Are there any best practices that should be considered?
Incoming sip did's and a zap line I had all were configured so that they entered the public context filtered by <phone_#>. Most of the examples I see for setting up the spa don't function like this but a couple do? Thanks! jlc
On Sat, Jan 9, 2010 at 9:40 AM, Joseph L. Casale <jcasale@activenetwerx.com>wrote:
It seems there are two ways to configure an spa3102's fxo port w/ pbx's, you can set the dial plan to <phone_#_of_pstn>@<freeswitch_ip> or <ext_to_dial>@<freeswitch_ip>.
From fs's perspective, what exactly is the difference here? Are there any significant differences between the two methods? Are there any best practices that should be considered?
I've only got a PAP2T (2 FXS) but from what you describe I'd say that it makes sense for the FXO port to be "phone_#_of_pstn" and FXS to be "ext_to_dial." I believe FS tries to be endpoint-type agnostic in this scenario. It's a SIP call in and gets routed in the dialplan.
Incoming sip did's and a zap line I had all were configured so that they entered the public context filtered by <phone_#>.
If a call is coming in from an actual PSTN line then hitting the public
context makes a lot of sense.
Most of the examples I see for setting up the spa don't function like this but a couple do?
As usual "it depends." However, the simplest rule of thumb would be that FXS ports are analogous to SIP users and FXO ports are analogous to SIP gateways. Out of curiosity, what is your application? -MC
I've only got a PAP2T (2 FXS) but from what you describe I'd say that it makes sense for the FXO port to be "phone_#_of_pstn" and FXS to be "ext_to_dial." I believe FS tries to be endpoint-type agnostic in this scenario. It's a SIP call in and gets routed in the dialplan.
This is what I thought as well, where I am unclear is how you register this device as a gateway then as it isn't like a regular sip provider by means of a gateway definition? The gateway definition requires a username and password for example so how do you create the basic definition to allow the spa3102 to push the call into the public context? I can make an outbound call as I just construct a dialplan like: "sofia/internal/$1@spa3102.domain.local:5060"
Incoming sip did's and a zap line I had all were configured so that they entered the public context filtered by <phone_#>. If a call is coming in from an actual PSTN line then hitting the public context makes a lot of sense.
Out of curiosity, what is your application?
We use this one pstn line as backup when voip is down for whatever reason. The company # given out is the single pstn line, its routed on busy to the sip provider for a series of other lines. As we expect the pstn to always be working, if the voip is up and more people call we get the calls. If voip is down, at least we have 1 line always working w/o any manual intervention. Outgoing always tries sip first and then routes to the pstn line if required. Thanks, jlc
On Mon, Jan 11, 2010 at 6:24 PM, Joseph L. Casale <jcasale@activenetwerx.com
wrote:
I've only got a PAP2T (2 FXS) but from what you describe I'd say that it makes sense for the FXO port to be "phone_#_of_pstn" and FXS to be "ext_to_dial." I believe FS tries to be endpoint-type agnostic in this scenario. It's a SIP call in and gets routed in the dialplan.
This is what I thought as well, where I am unclear is how you register this device as a gateway then as it isn't like a regular sip provider by means of a gateway definition? The gateway definition requires a username and password for example so how do you create the basic definition to allow the spa3102 to push the call into the public context?
Please email me off list and we'll work out the particulars and then create a more complete wiki entry. -MC
participants (2)
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Joseph L. Casale
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Michael Collins