First time poster! Posted this on Spiceworks and was given advice from Scott Alan Miller to come here looking for some additional help

Local Elastix/FreePBX install. Currently the PBX connects to a WRT54GL running Tomato which then goes out to the WAN. Local IP was set to 192.168.0.0 network, switched to 192.168.3.0 and now I cannot call out/call in. Local phones CAN call each other. SIP Trunk Provider is Voxox and authentication is done through WAN IP, which has not changed. I've changed the 192.168.0.0 network to the 192.168.3.0 everywhere that I can think of - any device on the network can see the web, and I can ping my SIP provided IP, but still no communication.

Any ideas?

Reviving my thread -

Weirdest thing. We had a power failure/surge over the weekend which knocked down all my servers, switching, etc.

My phones have been working FLAWLESSLY since I last posted. But now, after getting everything back up and running, the SAME issue is back.. but the rules that fixed it before are still in place! Outgoing calling works (though it seems highly delayed now.. like 5-8 seconds after dialing a number for it to start ringing) but I CANNOT call in UNLESS I call out first, which fixes it for 2-3 minutes.

Once again, it's gotta be a firewall issue. This is stupid.

I've checked the settings on both the WRT54GL and the FreePBX firewall. The WRT54GL only had some port forwarding rules for the 192.168.0.0 FreePBX host but those have been updated to match the new 192.168.3.0 address. I had an issue with the local GUI connections through the FreePBX firewall but someone assisted me in getting that switched to the 192.168.3.0 network so that should be good I hope.

I'm looking at the logs and I'm getting a "network is unreachable" error for my SIP provider IP when attempting to dial out and also "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0"

When I call out, I get a few seconds of silence and then eventually an "all circuits are busy" message

Also if you post the solution on Spiceworks as well I'll give you a best answer since you were responding on both

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)

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

Just noticed a line on Asterisk that says -

Got SIP Response 486 "Busy Here" back from <trunk IP>

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.

So I messed with my SIP trunk settings and inbound calling changed from dead silence to a busy signal so it's definitely getting through the firewall.

@jaredbusch said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

https://www.voip-info.org/asterisk-sip-qualify/

Interesting that this was working before without requiring this

Wow, that trunk is fucked up if you did not have those set...
I am surprised shit ever worked.

This is a typical SIP trunk setup.

username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

I am sure you have mentioned it in one post or another, but what version of what are you on?

It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

Haha yeah definitely NOT going to try to migrate.. any suggestions on what I should move to? Was thinking maybe latest actual FreePBX or something like 3CX

You are on Asterisk now, so stay on it.

Move to FreePBX 14.

I mean if you want to learn more, you could try Wazo or some other Asterisk distro.

But that is for people that want to be PBX people.

Yeah I mainly just want something simple and stable at this point

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

https://www.voip-info.org/asterisk-sip-qualify/

Interesting that this was working before without requiring this

Wow, that trunk is fucked up if you did not have those set...
I am surprised shit ever worked.

This is a typical SIP trunk setup.

username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

I am sure you have mentioned it in one post or another, but what version of what are you on?

It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

Haha yeah definitely NOT going to try to migrate.. any suggestions on what I should move to? Was thinking maybe latest actual FreePBX or something like 3CX

You are on Asterisk now, so stay on it.

Move to FreePBX 14.

Sounds like a plan! Thanks

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

https://www.voip-info.org/asterisk-sip-qualify/

Interesting that this was working before without requiring this

Wow, that trunk is fucked up if you did not have those set...
I am surprised shit ever worked.

This is a typical SIP trunk setup.

username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

I am sure you have mentioned it in one post or another, but what version of what are you on?

It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

Yup time to move. Do not bother trying to migrate. I tried Sangoma's migration script twice on two different Elastix servers. It sucked both times. Both times I rebuilt and manually migrated.

Haha yeah definitely NOT going to try to migrate.. any suggestions on what I should move to? Was thinking maybe latest actual FreePBX or something like 3CX

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

https://www.voip-info.org/asterisk-sip-qualify/

Interesting that this was working before without requiring this

Wow, that trunk is fucked up if you did not have those set...
I am surprised shit ever worked.

This is a typical SIP trunk setup.

username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

I am sure you have mentioned it in one post or another, but what version of what are you on?

It's not actually FreePBX, I mislabeled this and my previous thread originally. It's Elastix 2.6.18. We had a conversation awhile back about it as I didn't realize Elastix used a FreePBX GUI and thought I was running FreePBX the entire time

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

https://www.voip-info.org/asterisk-sip-qualify/

Interesting that this was working before without requiring this

Wow, that trunk is fucked up if you did not have those set...
I am surprised shit ever worked.

This is a typical SIP trunk setup.

username=TRUNKUSERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=TRUNKPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=TRUNK.IP.ADD.RESS
fromuser=TRUNKUSERNAME
context=from-trunk
canreinvite=nonat
disallow=all
allow=ulaw

Yeah no clue... now that it's working I'm going to start looking at my options to upgrade this entire system. New PBX, new trunk provider, etc. I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.

Well it appears to be working now... the busy signal lead me to think I could possibly tweak the trunk settings further on the PBX to get it all working. Switchd nat=no to nat=yes (Not sure why that didn't matter last time) and I also added qualify=yes. Looks like this explains why -

https://www.voip-info.org/asterisk-sip-qualify/

Interesting that this was working before without requiring this

So I messed with my SIP trunk settings and inbound calling changed from dead silence to a busy signal so it's definitely getting through the firewall.

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

There are only two occasions when you want to port forward the traffic for your voice over IP.

Condition one if you have external phones.

Condition to is if your sip trunk provider does not use registration but instead uses IP validation. This is a rare case normally.

My SIP provider does actually use IP validation instead of registration.

I put in a support ticket with Sophos and they went through all of the rules/logs and confirmed that traffic is getting through both ways on the firewall. When inbound calling it is in fact being forwarded to the PBX, but the PBX is not responding back, which is why I'm getting dead silence on my inbound calls.

Actually no that’s not what was happening before. Before when the inbound calls were not working nothing hit the PBX

And it still fails to show anything on the PBX when it comes up blank.. but Sophos shows it as being pushed to the PBX with no drops! How the heck am I supposed to troubleshoot that?

@jaredbusch said in FreePBX inbound call issue:

There are only two occasions when you want to port forward the traffic for your voice over IP.

Condition one if you have external phones.

Condition to is if your sip trunk provider does not use registration but instead uses IP validation. This is a rare case normally.

My SIP provider does actually use IP validation instead of registration.

I put in a support ticket with Sophos and they went through all of the rules/logs and confirmed that traffic is getting through both ways on the firewall. When inbound calling it is in fact being forwarded to the PBX, but the PBX is not responding back, which is why I'm getting dead silence on my inbound calls.

I have been watching the logs for the last day or so as I've been testing. I've noticed on the Sophos that when the inbound calls don't work I get a hit on the firewall logs for my DNAT rule for my VOIP Provider > External WAN on port 5060

When inbound calls DO work, I get a hit for my DNAT rule, same IPs, but the port always shows as one of the RTP ports. So either way the calls ARE hitting at least the WAN interface and I'm getting a different response on the firewall depending on whether it works or not.