Posts made by SamSmart84

@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 I just don't trust this setup and I'd feel better having a system in place that I put in vs. an outdated one that I inherited.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.

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

Okay.. so now I'm more confused than I've ever been. I deleted my DNAT and my outbound rule for my SIP provider and now it works. Flawlessly. WHAT!?

I never understood what your router was doing in the first place. So any number of things might be the issue.

Well false alarm.. it's not actually working

Firewall Rule:
Internal Network > SIP > AnyIPv4

DNAT:
SIP Trunk > SIP > Public IP > Internal SIP Server

Okay.. so now I'm more confused than I've ever been. I deleted my DNAT and my outbound rule for my SIP provider and now it works. Flawlessly. WHAT!? How do the calls know where to go!?!?! Did my Sophos UTM get superpowers after a power spike?

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.

@scottalanmiller said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@triple9 said in FreePBX inbound call issue:

@samsmart84 Maybe this Sophos KB article will help?

Success! So basically some trial and error with the settings.. I had to turn OFF the Sophos VOIP options as that was actually BLOCKING everything (Go figure).

That’s actually expected. That’s SIP-ALG. Always have to disable that. It’s basically just SIP blocking.

Nice of them to include that as a SIP "feature"

@triple9 said in FreePBX inbound call issue:

@samsmart84 Maybe this Sophos KB article will help?

Success! So basically some trial and error with the settings.. I had to turn OFF the Sophos VOIP options as that was actually BLOCKING everything (Go figure). I had made the DNAT rule listed in that link before but didn't have any success - the auto firewall rules with it didn't work evidently. Running the DNAT mixed with creating manual firewall rules for outbound calling and it all seems to be working. I've been testing for about an hour - when I turn the DNAT off it stops working and when it's on it works just fine so I think we've got the right combo at this point.

Thanks for all your help everyone! That was a pain in the butt to track down.

@jaredbusch said in FreePBX inbound call issue:

I am trying to think where else to look, but not getting anywhere.

Yeah it's just so odd.. I can't wrap my head around why the SIP trunk IP changing would break inbound. On the old router config there was literally no reference to the actual SIP trunk IP or anything else I could see that would be affecting it

@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 okay, so this confirms shit is just fucked up.

Ok. The hard part here is making sure you have no outbound calls.

Then when you make an inbound call monitor the CLI

asterisk -rvvvvv

And then call in.

I get nothing in the CLI for inbound calling when it fails. When I call out first then call in, stuff shows up obviously but not seeing anything that would indicate an issue (as far as I can tell)

What version of FreePBX?

Old. 2.8.1.5

umm, ok.. /me racks brain cells

So first, shut off iptables and see if the problem goes away.

service iptables stop

Your system if not direct ont he internet right? So this should be safe enough for a short term test.

No it is not direct so should be safe. Issue remains after shutting off iptables

I really need to just trash the whole thing and start with a setup that's actually modern/supported

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 said in FreePBX inbound call issue:

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 okay, so this confirms shit is just fucked up.

Ok. The hard part here is making sure you have no outbound calls.

Then when you make an inbound call monitor the CLI

asterisk -rvvvvv

And then call in.

I get nothing in the CLI for inbound calling when it fails. When I call out first then call in, stuff shows up obviously but not seeing anything that would indicate an issue (as far as I can tell)

What version of FreePBX?

Old. 2.8.1.5

@jaredbusch said in FreePBX inbound call issue:

@samsmart84 okay, so this confirms shit is just fucked up.

Ok. The hard part here is making sure you have no outbound calls.

Then when you make an inbound call monitor the CLI

asterisk -rvvvvv

And then call in.

I get nothing in the CLI for inbound calling when it fails. When I call out first then call in, stuff shows up obviously but not seeing anything that would indicate an issue (as far as I can tell)