OKAY. So.. we are back to the original problem but I have fully migrated the PBX to the newer infrastructure PBX > Sophos UTM > Fiber vs PBX > Linksys router > Comcast Business Modem
I can call out, I have audio, but I can only call in for a short duration AFTER an outbound call is made.
So made some progress.. If I call OUT now to my cell my cell will see it. There is no ring from the internal phone and once connected there is zero audio. Still cannot call IN.
I will say.. I like the Sophos box, but I hate the GUI and how "proprietary" everything feels. It's difficult to navigate and determine how to properly open ports, etc. for this to work. Still no outbound or inbound calling working.
@scottalanmiller said in FreePBX inbound call issue:
@samsmart84 said in FreePBX inbound call issue:
Alright I'm tired of dinking with this old equipment. My primary network is running on a Sophos SG 230 connected to fiber. No more Linksys, no more Comcast! If I'm going to fix this I might as well do it right.
We have great luck with Ubiquiti gear. Fraction of the cost of Sophos and I dare say, better quality.
I love Ubiquiti stuff. I built out our WiFi and our IP camera system using all Ubiquiti gear. Awesome quality and reliability for the cost.
Alright I'm tired of dinking with this old equipment. My primary network is running on a Sophos SG 230 connected to fiber. No more Linksys, no more Comcast! If I'm going to fix this I might as well do it right.
@scottalanmiller said in FreePBX inbound call issue:
@samsmart84 said in FreePBX inbound call issue:
Just noticed a line on Asterisk that says -
Got SIP Response 486 "Busy Here" back from <trunk IP>
Does that come up reliably when an incoming call has the problem that you mention?
That's the first one I've seen go through.. underneath that it has a few lines stating everyone is busy/congested at this time, dial failed due to trunk reporting busy, etc.
Update: Looks like that was an outgoing call. Incoming calls are going through right now as people keep calling out, which fixes the issue for a bit as stated before.
Just noticed a line on Asterisk that says -
Got SIP Response 486 "Busy Here" back from <trunk IP>
Not my doing, on my list to upgrade/replace.. 
Looks like when the call goes to a busy signal that it's indeed not hitting the PBX as I get zero incoming call info in Asterisk
I don't see any SIP and/or ALG type settings on the router. It's a Linksys WRT54G running Tomato 1.28
Yeah the router has been rebooted 2-3 times today. Trunk provider suggested that too - I've done reboots on it, the PBX and the gateway.
I did the "sip show peers" command and I only see the new IP in the list. Watched a call out and a call in (that worked right after) and everything looked in order as far I could tell.
For the firewall side, the PBX is just running through an old Linksys router/Cable modem setup. I've checked through the router and modem and can't find any references to the trunk IP, just the routes from our static IP to the internal PBX IP. What could I be missing there?
Our provider uses our IP for authentication so no registration settings are configuring on the PBX side. They updated their IP configuration and sent us an email saying we needed to update it on our end. Once that was done, I let them know that the IP had been changed so they could remove the old IP from our config on their end. I've talked with them a little this morning and they have told me that everything is set correctly on their end.
Hi all,
First off, I'm not super good with PBX stuff. We're using an older system which is soon to be upgraded. However, last week our trunk provider informed us that we needed to swap the IP for our inbound/outbound config. I did the swap (as far as I can tell the only place this IP is listed is under our PEER section for trunks and no other config settings were changed - just the new IP), called out and in and all seemed well. Here we are a few days later and my OUTBOUND calling works 100% of the time, however my INBOUND now works maybe 20% of the time - randomly. The rest of the time I get a busy signal when calling in. Trunk provider is literally zero help so... what could be causing this?
Update: Another oddity.. if I call OUT to someone, it appears to fix the ability to call IN for a short duration (5-10 mins)
Cool I'll give that a read. Thanks
Must have been a different one
Makes sense to not recommend Elastix if the support isn't there. Is that from a lack of effort from the team behind Elastix or just lots of mistakes/problems in comparison to FreePBX? We're fairly small, only around 50 phones. This system was put together by local paid consultant years ago, before my employment here began. It's probably due for an update/overhaul as I think it was a cheap toss-together in the first place. I'll add switching over to full blown FreePBX as a potential option.
Learn something new everyday. It always confused me why people referenced one or the other when it seemed like mine was both. Now I understand. That'll help me moving forward if/when I have to ask questions about the system. Thanks for the info.
Okay so I'm running JUST Elastix even though it utilizes a web app labeled FreePBX? That makes a lot more sense now.
Also if you post the solution on Spiceworks as well I'll give you a best answer since you were responding on both
I actually didn't fully realize that Elastix and FreePBX were completely separate since changing some setting in one seemed to do it for both, while not others and since FreePBX was accessible directly from Elastix. Is there a reason why both are present?