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Hello everyone! I'm looking for some information/help/guidance on connecting two different PBX systems. I am looking to connect a Panasonic KX-NCP500 and a FreePBX server I made from a Raspberry Pi. When I say connect all I mean is that the FreePBX and the Panasonic systems can dial each others extensions. The FreePBX server is running FreePBX 14.0.2.14 with asterisk version 13.20.0. As my understanding that on the Panasonic side I need DSP card(Which is has) and the Activation Key for SIP trunks and Extensions(Which it also has). I believe I need to create a trunk between the two systems and establish a dial plan. I know how to change all the settings on the FreePBX side, but not sure what is required or where to set these things up on the Panasonic. I read all 3 programming manuals for the PBX however I am still pretty new in the whole VOIP world. Learned most things on the fly making the RasPBX, don't know the Panasonic system very well, but trying to learn. Has anyone done this before? Does my logic on connecting the two make sense? Is there a better way to connect the two? Thanks everyone, any insight would be greatly appreciated.
Update: Seems I have a sip trunk between the two. However it's not working properly. For example when I dial extension 101 from the FreePBX it goes to the voicemail box of the extension of 101 without ringing. Does the same thing no matter what extension I choose 102 goes to 101's mailbox and so on and so forth. When I use the Panasonic PBX to dial extensions on the FreePBX.(for example 202) all I get is a busy tone. So does this sound like the issue is the trunk or issues in the dial plan?
Posted By: Randy Re: Need help connecting Panasonic and FreePBX - 04/12/18 06:03 PM
I would think that you would need to put the digits the Panasonic sees from the SIP trunk in the DID table and route them there. One side needs to be the client, one side needs to be the Server per trunk, i.e., the Panasonic has one trunk from the FreePBX going to the V-SIPGW trunk card. On the FreePBX this connection would go to the SIP Extension side.
Thanks for the reply! So I have the FreePBX set up as the following. For inbound routes I have an extension setup for the number. The DID number is 500. The outbound routes is setup as Intra-Comany, and the outbound routes dial pattern is 102. 102, 103, 104, 105, and the next box below says match pattern. As for the trunk on the freePBX side the general tab is set to default besides the name, and the dialed Number Manipulation Rules are 101 and 102. Next box says match pattern. For the trunk sip settings(outgoing) I have the trunk name and in the PEER Details I have
host=(ip of the local panasonic pbx)
user=102
type=peer
port=35060
Sip settings incoming is left blank.
I will include the log of the FreePBX when I call any of the extension on the Panasonic system.

Attached picture Call from Asterisk to Panasonic Part 1.JPG
Attached picture Call from Asterisk to Panasonic Part 2.JPG
Update: Well I figured out half of it. My dial plan in the Outbound Routes didn't work properly. I needed to put 1XX in just one line instead of listing the extensions directly. So FreePBX can now just dial any Panasonic extension and it will ring. Now have to figure out how to do the same thing on the Panasonic side so it can call the FreePBX
Posted By: OBT Re: Need help connecting Panasonic and FreePBX - 04/18/18 08:16 PM
It is working one wau because you are picking up an extension on the panasonic from the free pbx and that can dial any extension on the panasonic but because the panasonic is acting like a co it will only ring free pbx if you dial 102

You could have a disa on the free pbx and have a dial plan to dial 102 pause then dial extension on free pbx or have sip trunks on panasonic register to sip extensions on free pbx setup other pbx dialling plan on panasonic

Have you gateway trunks option in the free pbx?
This is my updated "Peer Details" that work still. I didn't need the user=102. The 102 was an extension on the Panasonic not the freePBX. I just need to somehow assign the trunk group to the SIP channel, I believe.
host=200.117.**.**
type=peer
secret=**********
port=35060
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