FXO Hotline - Vega 50/60G


The following describes how to configure Vega50 FXO between an analog and IP PBX, in order to providing direct access between the analog PBX extensions and SIP calls from IP PBX  
 

 

Call scenario

20000 = IP PBX number used to route to the analogue PBX through the Vega.

1. IP-PBX caller calls 20000

2. IP-PBX routes call to Vega

3. Vega routes call to FXO port

4. Vega sends dial tone to IP-PBX caller

 

STEP 1: Log into the Vega gateway using the WebUI interface. The default user is admin and password is admin. 

STEP 2: As the Vega gateway WebUI loads up the Status page as shown below. 
              Once the page has finished select the "Quick Config" menu option as shown below:

STEP 3: The WebUI will present a warning message please read the message and then select "Continue" on the screen below. 

STEP 4:  Once the Quick Config UI has loaded should select the Basic Tab.  

NOTE This is where the country localisation and IP address configuration is specified

NOTE You must specify the correct country or the Vega will not be able recognizeise the network tones for you Telephone lines

 

STEP 5:  Once the Basic config is completed select the VoIP tab.  The screen shot below highlights the most important fields.

SIP Domain: this is normally the IP address or Hostname of the SIP PBX we are sending calls to and receiving calls from. 
Proxy Address: this will be the IP network destination for SIP calls from the Vega to the PBX
Registrar address: Not used
Outbound Proxy address: Not used 
Registration mode: Must be set to Off


STEP 6: The parameters to be configured by Quick config are complete  
               select the "Submit" button.

NOTE The Quick Config Submit creates a popup Click "OK" on the next window to apply the settings. 

NOTE The save process creates a  select "Continue" on this window. 

 STEP 7: The next step is to configure the Dial plan for calls from the IP-PBX.  
                This is the important section which defines the Vega calls the analogue PBX extension.

Navigate to Expert Config > Dial Plan. From the available Profiles select "modify" for the "To_FXO" profile.

 

STEP 8: Once the page has finished loading, modify the Plan IDs to configure FXO port to send PBX dial tone back to te IP-PBX caller.
              "FXO_01" - this entry defines the behaviour for the first FXO port to connect to the Analogue PBX.

                Source - enter "IF:99..,TEL:20000"

Note: "IF:99.." defines that source call from any SIP interface 
          "TEL:20000" defines the dialed number of the call that must match for the call to be routed to the defined destination below.

                Destination - enter "IF:0201,TEL:"

Note: "IF:0201" defines the destination interface to send the call that matches the source expression
          "TEL:" defines that the call will not send or dial any number, it will simply go off hook 
                    - this will connect the SIP call directly to the analogue PBX extension and then send the PBX dial tone.

This defines the SIP -> FXO behaviour in only one direction - SIP to Analogue PBX 

     

If more than one analogue PBX extension is required to be configured the same style of configuration can be used - a different source TEL number and a different destination IF need to be specified eg source "IF:99..,TEL:20001" destination "IF:0202,TEL:"

 Be sure to select the "Submit" to confirm the configuration changes ready to be applied and saved to the Vega

 

STEP 9: defines the use routing for calls from the Analogue PBX -> IP-PBX
              return to the Expert Config>Dial Planner web page and select to modify the "To_SIP" dial plan profile.

 

 

STEP 10: The next step is to modify the To_SIP profile. The standard To_SIP profile has a single entry sending all calls from analogue interfaces to SIP. This profile will forward calls from the Analogue PBX ------>Vega FXO interface -------->SIP -------->IP PBX.

     Source - enter "IF:02..,TELC:<.*>"

 

Note: "IF:02.." defines that source call from any FXO interface
          "TELC:<.*>" defines CLID of the calling party to be save and used in the destination field. 
These fields should match any call coming from the Analogue PBX to the Vega FXO extensions

    Destination - enter "IF:9901,TEL:123,TELC:<1>"

Note: "IF:9901" defines the SIP interface to send the call that matches the source expression,
          "TEL:123" defines all calls that will be sent with the DID 123 which the IP-PBX will need to route 
          "TELC:<1>" defines the CLID captured in the source description and populates the destination SIP call's From header with the captured info.

 

 

NOTE For calls inbound to FXO interface from the analogue PBX there are some specific POTS limitations which limit the conifg above.

Inbound calls from a POTS line do not have a called party telephone number as the POTS line is the terminating point for the call (but they can have CLID).

As such a call from a POTS line will only arrive at Vega with CLID information. One of the requirements of SIP is that call cannot be made without a destination telephone number > this means the Vega Dial plan must be used to define a DID number that is sent on to the SIP PBX, in our example is this given as "123" but this can be configured to what ever is required.

 

STEP 11: Make sure the config is submitted to the Vega, this concludes the Vega Configuration please select the "save" and "apply" to commit the configuration back to the Vega.

 

 

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