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Incoming Call Fails after internet outage

asterisk
Tags: #<Tag:0x00007f749a2d4810>

#1

I was wondering if anyone else has experienced this? I kept searching and saw there was a bug on pfsense from 2011 but has been resolved.

Currently running
pfSense 2.4.4 p2
5 - Public Static IP
FreePBX14.1.1
Flowroute SIP Trunk
NAT Inbound
5060 from Flowroute servers only
10000-20000 RTP from any location

pfSense Firewall Optimization Options set to “Conservative”

I’d have to get into asterisk and run “core restart now” or reboot the whole freepbx server to resolve incoming call failure.

If anyone has any suggestions please let me know and thank you! I may have missed something big.


#2

This should be very easy to troubleshoot.

  1. On Flowroute, if you are using registration, I recommend that you switch to IP authentication and static routing for inbound. This avoids the possibility of ‘lost registration’ and is also somewhat more secure. Also, specify the numeric IP address of your server in the route, avoiding the possibility of a DNS issue. See
    https://support.flowroute.com/customer/en/portal/articles/1851381-statically-route-your-phone-number-to-a-host-system-for-inbound-calls
  2. On pfSense, if you have not dedicated a public IP address for your PBX, confirm that UDP packets from the Flowroute servers and with destination port 5060 are actively forwarded to the PBX. See Option B step 1 in
    https://www.outsideopen.com/pfsense-asterisk/
    but note that this example opens the ports to the world – specify the Flowroute server addresses as Source.
  3. If your PBX is configured with a software firewall, confirm that your iptables rules will accept UDP packets from Flowroute destined for port 5060.

If you still have trouble, I assume that for testing you can cause this at will by disconnecting the modem for a few minutes. Of course, do this before or after business hours and be sure that Flowroute is set to failover to your mobile or other suitable destination. On a failing test call, does anything appear in the Asterisk log? If not, is Asterisk otherwise functional (internal calls are ok)? Does a tcpdump capture on the PBX show the incoming INVITE? If not, does a packet capture on the pfSense WAN interface show the incoming INVITE? If not, do you have a ‘gateway’ from your ISP (anything other than a dumb bridged modem) that might be blocking it? Is the PBX on a VM with other than unrestricted bridged networking?

Is the trunk pjsip or chan_sip?


(system) closed #3

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