annnd it’s gone again
That sounds like a NAT session failing. Look back through the archive for “30 second” and you’ll see lots of posts about things you can do to keep the port open and alive.
Isn’t that on a call though? or just it happen for extensions being registered too?
These are the NAT settings i have.
I have remote extensions at multiple sites > They hit the remote sites firewall (all traffic allowed from phone IPs to PBX external IP > Hits the USG Pro and allows all traffic through on those ports > hits the FreePBX firewall
This session through the firewall has to stay open for the phone to stay registered.
Being unreachable is almost always some kind of network error, so if it isn’t your NAT session, it’s something else in the network jacking you up.
Still having issues with the registration becoming unreachable from our school sites. But i have setup a softphone that works fine so i’m guessing it’s something on the schools firewall which i’ll have to investigate Monday.
My next issue is i can’t seem to get the PJSIP trunks registered, we use Gamma who are IP auth based so in my mind it should be as simple as;
Trunk Name: School Name
Outbound Caller ID: The DDI we use for that trunk
Max Channels: 3
SIP Server: 109...31 - double checked that this matches with what was provided by Gamma
SIP Port: 5060 - confirmed with Gamma this is correct.
Looking at the logs when making a call i get:
[2020-02-09 11:44:55] VERBOSE[C-00000009] pbx.c: Executing [s@macro-dialout-trunk:34] NoOp("SIP/1056-00000009", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3") in new stack [2020-02-09 11:44:55] VERBOSE[C-00000009] pbx.c: Executing [s@macro-dialout-trunk:35] GotoIf("SIP/1056-00000009", "0?continue,1:s-CHANUNAVAIL,1") in new stack
I am on Gamma (through Daisy) and our Trunk is configured as such.
User Context: (This is the primary telephone number they gave you, even if you ported another number over).
fromdomain=(Your External IP)
Trunk Name: Gamma (Primary number) Out
I don’t know if Gamma provided you with a different IP Address but thats what I was given for the primary.
However my MAX channels is set to 10 which has just been enough for us. Remember that’s inbound and outbound concurrent so any more will literally drop as if your number is disconnected (costs more through gamma to setup a divert).
Unfortunately I don’t have a UniFi gateway in front of my server so can’t help you there.
My trunks used to disconnect after a minute of no outbound calls… eventually found an issue with my Draytek router needing a restart for firewalls and DMZ rules to properly take effect.
Just checking your external IP in Asterisk SIP settings is correctly set?
Have you tried with CHAN_SIP and then come in from Gamma on e164format@YourIP:5160 for the DDI and built trunks for all possible IPs they might come in on?
This is now solved, Gamma gave me the incorrect IP details which would certainly stop it from registering!
Now to solve a lack of audio on calls… i’m nearly there!
[ ] 426 INVITE email@example.com firstname.lastname@example.org 8 REMOTE_EXT_IP:26571 Internal_PBX_IP:5160 IN CALL
This is from a remote extension, i’m pretty sure the destination shouldn’t be the internal IP of the PBX and should be the external one. I’ve looked in all settings etc and it has the external IP configured.
Adding the below. as i’m unsure if this looks correct or not.
2012 INVITE email@example.com firstname.lastname@example.org 10 192.168.1.1:36112 192.168.1.241:5160 COMPLETED [ ] 2013 INVITE 1056@pbx_external_ip:5160 email@example.com:5060 6 192.168.1.241:5160 remote_site_external_ip:26571 IN CALL
Do you have the RTP port range forwarded?
Also, did you specify the WAN address as well as the local LAN subnet in the Asterisk Settings?
If you have not found a solution yet… then I would suggest the following to start… A. enable the FreePBX firewall and enable SIP and PJSIP(if you use it) on the firewall. B. Enable Intrusion Prevention, you have to have the SYSAdmin module, well worth the price. C. Then at your unifi create a DMZ rule to allow unrestricted traffic. C. this is where it get tricky… We use the Device + Extension setup and with this setup when you create a device we have to go to the advance properties and enable NAT (auto is recommended). If you are using the Extension setup method then where ever you configure your phone setup you want to enable “NAT auto” for the device. Now I recently did a new client PBX and site deployment with 40 phones and 1 site worked fine but the others had weird issues.; could not register, call drops after 30 seconds. Since we do not control their network the simply solution was to enable the OpenVPN and configure the phones with it. Now alot of new phones support vpn connections from the phone, so you would have to explore this. It was our way of overcoming the clients network issues.
Happy to pitch in more if you have questions.
I have a few Freepbx systems set up on UniFi networks. If you are still having issues with this post back where you are with your network setup at this point and I will be glad to try and help you.
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