Digium D40 Can't Contact DPMA (no multicast)

Hi Everyone

I’ve just installed a fresh copy of FreePBX Stable-5.211.65-11 64-bit into a VMWare ESX 5.1 VM. I’ve done minimal configuration in that I’ve created a trunk + inbound + outbound route plus one extension. I’ve registered my DPMA key via the FreePBX GUI as well and ‘digium_phones license status’ shows that all is good there.

Now, I have the FreePBX server in our database on the local subnet The IP phone in question is on another network with local subnet There is an MPLS private IP network between the two locations, so those two networks can route between each other with no NAT what so ever. The phone can ping FreePBX and vice-versa without an issue.

On the phone, I’m manually specifying the FreePBX server IP of and the requests are timing out. Running tcpdump on the server side, I see UDP packets coming in like this:

15:14:15.911594 IP (tos 0x0, ttl 62, id 27830, offset 1560, flags [none], proto UDP (17), length 39) > udp
0x0000: 4500 0027 6cb6 00c3 3e11 fdeb 0a04 033e E…'l…>…>
0x0010: 0afd f922 4420 4345 5254 4946 4943 4154 …"D.CERTIFICAT
0x0020: 452d 2d2d 2d2d 0a00 0000 0000 0000 E-----…

but there doesn’t appear to be a response from Asterisk at all, hence the request fails and times out. I’ve turned on core verbosity and debug and nothing appears to come through. Sip debugging doesn’t yield anything either.

Can anyone suggest any further debugging steps?


what doesn’t sip debugging yielde? it should tell you the ip and rtp port.

I don’t see any SIP debug entries related to the IP phone. From my understanding, the Digium phones provision themselves via SIP messages, which means they should show up during a debug. And yet I see the UDP packets reach the server, but there is no response.

I should also mention that IPTables is turned off (doesn’t have any rules anyway) and fail2ban hasn’t reported any IPs as getting blocked. I’m also able to use a soft phone form the network successfully. I’m also able to get the Digium phone to register against the Asterisk server by bypassing configuration and manually configuring the SIP account.

It just seems that DPMA/Digium Configuration Server is the only component not working.

Just curious, have you tried contacting Digium support? Since you are using Digium Phones together with DPMA and the FreePBX Digium Phone Add-on module, you should be able to contact the techs and get some insight into debugging / configuration.