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3web/Distributel VOIP Behind Router: So close, help please!

>I can't receive incoming calls!
Cause: Your ATA don't know its WAN address.

This can be via either a properly configured NAT setting (requires correct setting on VoIP provider, but does not need a STUN server) or a talking to a 3rd party STUN server.

Check this step. "Inputed STUN settings and changed the NAT settings" on your list.

reply to Twitch_City
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reply to Twitch_City
what NAT settings did you enable exactly?

reply to Twitch_City
You're close indeed.

Do you have a static IP address?
In that case you could do this:
admin --> voice --> SIP

Substitute VIA Addr: YES
EXT IP:

In any way, also in the same section:
NAT Keep Alive Intvl: 60

In the router port forward 5060 UDP to the internal address of your ATA or any other address you might have used (you can check that on the Line1 page: SIP port and EXT SIP port.

My VOIP has been running error-free for months now, but I had the same issue starting out. Some people say I've doubled the effort (since the ATA is supposed to keep it's port open in the router through NAT stay-alive), but I like to not take chances. Got one Acanac VOIP line and one ThinkTel business VOIP line.

Whatever you do, don't use the ATA as a router, use it for SIP only.

reply to zorxd
This is probably where I went wrong. I will take screenshots of the admin pages as soon as I get home (~ an hour).

I was following this tutorial, which basically says "check all the NAT boxes to Yes on the SIP page and the Line 1 page" and enter a STUN server on the STUN server line.

Ott_Cable: Thanks for the link. If I am not able to figure this out soon, I will make a post over there! (I put it here because I think that a substantial portion of the problem is the 3web provision script and their requirements, rather than generic VOIP requirements, but I could definitely be wrong!)

reply to Jurjen

reply to Jurjen
I made those changes - and provided my external IP address on that line (it changes every few months but rarely enough that I don't mind reupdating it when it does so).

However, same issue: green light on Line 1, dial-tone + external calls are OK; incoming calls "number is not registered".

I'm going to make screen shots of the SPA2101 configuration settings now.

reply to Twitch_City
Does your ATA web status page or your VoIP provider customer portal shows that it is successfully registered with the SIP server? On my voip.ms account status page, it would show my accounts registered to the servers.

If it is not registered, check server IP address, user ID, password etc for typos.

reply to Twitch_City

Settings on the Time Capsule

Router (SPA ···Side.pdf Router settings on the SPA2102

VOIP (SPA 21···ings.pdf Voip settings on the SPA2101

Shouldn't you get voicemail if your ATA isn't connected? Sounds like a problem with your number, not with your device configuration.

That is a good point. 3web said they were successful in porting the number, so I have no idea why it is not going straight to voice mail instead of this "Number not registered" garbage.

I am able to activate the voice mail system on my handset, which is connected to a mailbox somewhere...

EDIT: Oh, and obviously - when I make an outgoing call to our cellphone, the number appears correctly.

>when I make an outgoing call to our cellphone, the number appears correctly.

That has more to do with whatever CID# you like to be inputted into your ATA and not whether your number is ported.

As a matter of fact, I had VoIP for outgoing only long distance calls for a few months without having a DID# (or in your case a ported number).

So, are you suggesting I should call 3web and try to figure out the status of the ported number?

I was really hoping there was just a few configuration mistakes in the above files, and all the problems were on my end!

edit: Another interesting observation, if I call the temporary number 3web assigned me (which will be cancelled at the end of the month), I get a "the person at ______, is on the phone" message. If I plug my phone into Line 2 (which I guess should be set-up for that temporary number), I get no dial-tone. So, for some reason, the temporary number is more "in the system" then my real number.

edit2: Changed the NAT settings for Line 2 to mimic Line 1, and now when I call the number I get the answering machine - as expected.

edit3: Further testing shows that the temporary line works! I have received and made calls from it. The only issue I found was that dialing a number took the phone ~25 seconds between dialing and the phone beginning to ring, which is kind of a weird pause.

distributel and acanac are the same company

Acanac has a lot of issues similar to this with their VoIP. There is nothing you can do, except bug tech support so that they fix it for you.
The sad part is when they tell you to check your configuration...

Haha, naw - the worst part is the being on hold. 17 minutes and counting so far.

Just to ease my worries, does everything else in my configuration look OK? I'm happy that I have a proof of concept (that Line 2 does work as it should), but I would be relieved to have a seal of approval from someone who knows what they are doing more than I do!

It is okay, but I don't even need port forward, STUN servers, DMZ.

port forward: security issue, static IP address so have to remember to change it when you change the network. Not my cup of tea.
STUN: one more things to go wrong. When it works, it works.
DMZ: security issues.

reply to Twitch_City
I also would discourage putting the ATA in the DMZ. Just get him out of there! Just only have 5060 forwarded. Only connect the WAN to the rest of the network and open up the administration on WAN side.

This is a configuration that works for Acanac (so very close to what would work for Distributel/3Web, I presume):

Login as admin / advanced, we'll start with ROUTER settings.

Connection type: static IP
Static IP: 192.168.1.251 (remember/note down this address for future login use)
Netmask: 255.255.255.0
Gateway: 192.168.1.1 (assuming this is the address of your router)

Hostname: you may choose (in my case: LinksysATA)
DNS servers: choose your favorite ones. I like to combine several DNS servers over the world, just in case censorship might start.
DNS server order: manual
NTP servers: 0.ca.pool.ntp.org & 1.ca.pool.ntp.org
Enable WAN Web Server: YES (port: 80)

You may do a save in the mean while.

Go to VOICE settings (still admin/advanced)

Tab SIP:
On the bottom (NAT Support Parameters section)
Substitute VIA Addr: YES
All other option to: NO
EXT IP: your external IP address (if you have a static one, otherwise leave empty)
NAT keep alive interval: 60

Tab REGIONAL:
Go, again, to the bottom (Misc section)
If you set the date and time (pay attention to the syntax!), save immediately.
Time zone: you'll know where you are.
DST rule: start=3/8/7/02:0:0;end=11/1/7/02:0:0;save=1
the rest you can leave on default

Tab LINE 1:
enable: YES

NAT settings:
- NAT Mapping Enable: YES
- NAT keep alive: YES

SIP settings:
- Restrict Source IP: YES
- the rest remains default

Proxy and Registration:
Proxy: sub.domain.ca (see your welcome email)
Register: YES
Register expires: 1800
Proxy Fallback Intvl: 3600
Voice Mail Server: same as proxy

Subscriber info:
fill in your telephone number (just numbers, no spaces or special characters) in: display name, user ID and Auth ID
Use Auth ID: YES
Password: the device password from the welcome email

Dial plan:
Dial plan: (*123|911S0|S0|0|011xxxxxxxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0)

reply to Twitch_City
Mine relies on the NAT transversal setting on the server message to figure out the WAN address. No STUN, no DMZ, no port forwarding nor having to set WAN IP manually. voip.ms doesn't even have a STUN server. I don't think they recommend one either as you have no idea when a 3rd party serve might go down (either failure or cost reasons or unmaintained) in the future. If your provider has one for the subscribers, that's probably okay, but still one extra things that can fail.

It took me about 6-7 hours to figure out on my own what needed to be done from reading 200+pages of docs and tracing logfile. My ATA was too old so no sample configurations from them.

I think you have those option selected, but also enabled STUN.

I think you should get thing working and worry about it after the baby. You'll have plenty of time when you can't sleep at night.