So, my fellow network admin and I have painted ourselves into a little corner. Years ago, we’ve set up a trixbox (v2.8.0.4), and for the most part, it still runs great. The phones are all Polycom and Snom, and everything is perfect
But…
This network we manage for a client has grown. Business has picked up, so more phones and PCs have been added. Now, we’re running out of IPs on this class C network. The phones and PCs are all on the same subnet, and now we’re probably down to about 5 free addresses left. IP conflicts are starting to be a problem when the users there take it upon themselves to add equipment.
So, I’ve have drawn up in my head to create another subnet, a 10.10.10.x/24 network, and make this the phone only network. So far, all I’ve done is added a subinterface to trixbox, a eth0:1 entry. It’s pingable from any 10.10.10.x device. I’ve also started to create a 10.10.10.0 DHCP scope with mac reservations using the macs from the phones and will activate it once I think I have things figured out.
But I do have a few questions:
First:
How do the phones communicate with the server and then to the outside world? Do the phones relay voice through the server directly, and then the server sends sip packets through our SIP trunk? Or do the phones receive a token from the server and then the phones send packets through the SIP trunk itself with the server monitoring it all? The trixbox server will still retain it’s original IP address on the 10.18.157.x along with the new 10.10.10.1 secondary IP, as the provider’s sip trunk LAN port is 10.18.157.x… Also, is there any routing I would need to enable on the trixbox, and/or routing entries?
Second. I read elsewhere that for trixbox, I would add entries into the sip_general_custom.conf file, a bindport = 5060 entry and a bindaddr = 0.0.0.0 entry. But, would I need to comment out the #sip_general_custom.conf entry in sip.conf and then reboot? Also, sip.conf doesn’t have much in it, with no bindport or bindaddr entries. However, sip.conf.0 does contain bindport and bindaddr entres, showing both 5060 and 0.0.0.0. Does Trixbox currently use the sip.conf.0 as a running file, and would this mean it is already listening to any available IPs?
Sorry for so many questions for a first time poster. But, we want to have the legwork done and functioning before we commit to an after hours, probably all nighter of work, changing TFTP entries on every phone, checking, testing, etc.