I have setup a no-ip host name to my home dynamic ip.

I have spend multiple hours of combination of setting and understandings.

I have hard time having messages of asterisk because of not knowing linux.

All or most combination of setting on inbound routes and trunks incoming detail where tryed.

I think I have something with host=IP of my ATA, because of stats the show online when is seems to talk to the IP of ATA.

I know I dial with STUN, my home server and have a call with audio no problem.

I set my DID to call [email protected] works fine.

I have tryed custom extension.

So now relying on freePBX to point my DID on there directly is not working.


I’m lost now.

I was testing sip extension on the local network.
it registers fine.

I have switched to another router (nat/on/off not sure) and extension is registering fine from wan side.

this router goes into another one wich is internet connected.

from outside this second router it does not register.

I am playing with routers and seeing their limits.

I am sure nat is partly in cause.

I am now fixing my network before anything else.

NAT and STUN when their specific addressed and ports
regarding RTP
when opened only in firewall authorized both ways in duplex
from wan to lan AND from lan to wan
is correct and I have audio without STUN.
For all that I can say I have solved a STUN problem need for a direct DID calling my home pbx. (or home ATA system module direct)

port forwarding - virtual servers - firewall are all conflicting.

One microstep at a time I was able to register locally on the pbx network then was able from wan side, then from router to router then from the wan connected to the internet…

Anyways, its fun all that. I’m happy to have learned all that systemic things.

Moreover when testing or else switching configuration changes fastly yields
problems, because delay needs to be thought of to let the system accustom, even more the changes may not actualize, its probably like when networks are configured, the computer needs boot up or the router need a minute (…)

I have read your message twice and don’t understand it.

Asterisk support for STUN and TURN is minimal at best.

All you had to do was use a dynamic DNS service then put the FQDN in the externhost field in Sip Settings module. Lower your RTP range to something lower than default (say 10,000-10,400) then forward UDP 5060 and 10,000-10,400 to the service inside your network. Make sure all SIP ALG’s and transforms are turned off on your router and you would be good to go.



I said only that I had network serious issue that was solved microsteps at a time.

All the audio goes on the RTP port; not much channels/port need to be open for duplex conversation unless, one needs to double the RTP port range for three-way calling?! Is that it. Or else what clearly does lowering the RTP ports range/numbers? Making those port on unusual numbers must avoid conflicting ports?

Does nat count as ALG? I am ok with nat.

The other problem I am facing is my voip/internet real provider:
their sip ata phone spa2102 sometimes loses inbound call; its the sip messages the problems. Now I have at home their ATA + my ATA + a system of one and more spa1001 reserved on my network. And I thought and saw that maybe sip ports maybe confilcting: now I have their on ? sip, my ATA on another sip port, and all others will be different sip.

I am waiting now to see is there is more conflicts.

I have setup my trunk.
for incoming call.

user context = TEST
host=ip of my ata spa 1001
port=port of my ata spa 1001 5062
user=extension of ATA&freePBX
username=extension of ATA&freePBX

yield when calling SIP/[email protected]:5061 the number you have dialed is not in service.

I truly know that context is good, user and or username is good, host ip of ATA good, codec are excellent and well alined, nat good everywhere (but did no configutration on asterisk server only thru freepbx).

Call is going into asterisk, I see from the CLI.

Its not the distro the problem.

I have not writed down my setup of last setup. I thought I was copnfident enought to set it up again.

note: call are going out very well.


If someone can tell me how to setup “sip debug on” and whatever messages or verbose output on the freepbx ASTERISK CLI webpage I could supply you the output.

There is many sip anonymous sip setting switches.

I have re-visited the SETTING TAB of Freepbx webpage,
under general setting saw security settings and
Allow Anonymous Inbound SIP Calls? was set to no.
It has been allowed everywhere else.
Now, Allow Anonymous Inbound SIP Calls? is set to yes.
Number was ringing from inside out both ends and awnsered,
had duplex audio.

Now I am back.

Still very happy with FreePBX.

Thank you for your , passion?