- 2N® VoiceBlue Next Parameters
- IP address: 192.168.88.2
- Port: 5060
- Firmware: 03.00.03rc3
- PBX Panasonic KX-NCP500VNE
- IP address: 192.168.88.101
- IP DSP: 192.168.88.102
- Port: 5060
- Port Invite: 5060
- Firmware: 003-000
2N VoiceBlue Next settings
- SIP trunk interconnection
For the setting of the trunk between the 2N® VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed to secure communication just with traffic from your Panasonic PBX. You can specify the IP address and port from which SIP packets will be accepted.
In case you leave there 0.0.0.0 it will be open for all traffic.
- Configuration of the LCR (Least Cost Routing)
The GSM operator has e.g. in our country prefix 7 and 8 with a 9-digit number. The setting is below.
You need to create LCR rule for defined prefixes. The GSM group defines a way for the outgoing call routing. An appropriate SIM card is selected based on the GSM groups assignment.
- Configuration of GSM outgoing groups
You are able to set up different setting for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others)
- Incoming calls
For incoming calls you can define 2 groups with the different behavior and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.
In GSM incoming groups you can define the behavior for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.
You can define the list of called numbers which will be automatically dialed after DTMF dialing timeout if the customer does not press any button within the specified time. From the configuration you can see 10 seconds for DTMF dialing and after that the call will be routed to the extension 100 to your Panasonic PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters).
Panasonic NCP settings (by Masscomm)
- Connection settings
First of all we need to create a new slot as SIP Trunk (SIP Gateway Virtual type).
By default Panasonic NCP has the port 35060 opened to receive INVITE messages from the other side. We have to switch Virtual rack and V-SIPGW16 card to mode "OUS". Then we have to change the parameter in the virtual rack and card Properties of the rack->Virtual SIP Gateway (Trunk) Number of port of SIP Value: 5060. (Default value 35060)
After this change, the PBX must be restarted!
- SIP Trunk settings
The port which is going to be used has to be set as OUS.
In the main tab fill the following parameters:
Channel Attribute=Basic Channel
IP address=192.168.88.2 (in the example)
In tab Account the fields Name, Authentification ID, password, should be filled. In our case we set all parameters as VoiceBlue
We go to tab REGISTRAR and we have to set REGISTRAR CAPACITY as DISABLED. After that we will configure the server to Registrar Name: VoiceBlue; IP= 192.168.88.2.
After having followed all previous steps, the port must be set in mode INS (It should remain marked). Now we are able to do the testing of the interconnection between both devices.
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