Linksys PAP2T-NA Configuration

If anyone is familiar with Linksys PAP2T-NA configuration problems please help.

I read all the information provided on the website regarding the configuration of these ATAs. The PAPT2 is configured as recommended and it does receive calls but it will not dial. The CDR report shows “FAILED”.

I have determined that the problem is with the dial pattern. The PAPT2 is configured with the standard dial pattern:


and the FreePBS has the following outbound dial patterns:

There is a disconnect between the two dial patterns and nothing or something very wrong is sent to the SIP.

Any ideas or suggestion would be greatly appreciated.


dosn’t match




as a note, 011. can be an incredible expensive mistake. you should probably be more selective :slight_smile:

(And please explain why you think “xxxxxxxxxxxx.” is “standard” :slight_smile: )

Although dicko is of course correct, that’s not your primary trouble. You need to decide what format(s) of North American numbers you will use and make your devices, FreePBX configuration and provider configuration consistent with them.

As your system is presently set up, if you dial a 7-digit number, the PAP2T will accept it, FreePBX will accept it, no modification will be done and it will be sent to your provider as 7 digits. Most providers won’t accept that format at all; a few only after configuring your default area code on their portal.

If you dial area code + number (10 digits), even though that’s not in your PAP2T dial plan, after a long delay it will still send it out and FreePBX will accept it. However, by default, most providers will either not accept it, or interpret it as an international number (treating the first digits as the country code). Many providers have a setting on their web portal to allow this format for North America calling.

If you dial 1 + area code + number (11 digits), the PAP2T will accept it but FreePBX won’t. Those calls will probably get a “your call cannot be completed as dialed” message from FreePBX and not appear in the CDRs at all.

IMO, you should add a 1NXXNXXXXXX pattern to your outbound route and try some test calls by dialing 11 digits. If they still don’t work, look at the FreePBX logs and those at your provider to see what’s wrong. Once that’s working correctly, you can set up FreePBX to properly rewrite the shorter formats that you want to use.

[quote=“dicko, post:3, topic:22874, full:true”]
(And please explain why you think “xxxxxxxxxxxx.” is “standard” )
[/quote]Correct or not, that really is in the Linksys default. It allows sending out (after a delay) international numbers with 00, 011, or no prefix, so it’s compatible with many setups.

Thank you very much this is very helpful.

Ok so:

  1. I used the word “standard” in error, I should have written “Linksys default”.
  2. The 011 dial pattern in my FreePBX doesn’t really do anything because international calling is blocked by the SIP trunk provider and it takes my calling them to turn it on or off however … for now I will remove it anyway.

Since I don’t really understand the syntax should my Lincsys dial pattern look like this?:


I did try the FreePBX recommended (911S2|[2-9]xxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|*67S0|[*x][*x].S4), which did not work. I also tried to have no dial pattern thinking that the FreePBX will do the translation and that did not work. Perhaps somebody who is using PAP2Ts successfully could just send me the dial pattern he/she is using.


The ATA admin manual is at . Starting on page 56, you’ll find more than you’ll ever want to know about dial plans.

If you use the FreePBX recommended dial plan and add a 1NXXNXXXXXX pattern to your outbound route, you should be able to call numbers in North America by dialing 1 + area code + number (11 digits total). If it doesn’t work, something is wrong other than the dial plan; you need to look at some logs or otherwise find out where the trouble lies.

Once that’s working, you need to plan what formats of numbers you want the system to accept. For example, if you want to be able to dial 7 digits for a local call, the area code must be supplied somehow. You can do that in the PAP2T dial plan, you can do it with the FreePBX outbound route or trunk settings, or you may be able to do it at your provider. But, you need to plan and do it in a consistent way.

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Ok, I understand but, what is the relationship between the dial pattern in the freePBX and the dial pattern in the ATA? Do they have to match and why do both have a dial pattern? It seems to me that one is enough either in the telephone instrument or its ATA (if it’s an analog device) or in the PBX … why both?


There are two dial plans because the ATA collects the digits and sends them to the PBX. If you want to wait until the timeout or press # at the end of dialing you can have X. as you dial plan and call it a day. That is not very elegant.

Most users match the internal extensions, then the emergency codes, then feature codes, then local short dialing and LD dialing.

The outbound route dial patterns (not dial plan) select what outbound route a PSTN call will take. In that plan if a user dials 7 digits then you need to add the area code in the prefix of that pattern (and a 1 if you carrier needs it). Any outbound digit modification take place in the outbound routes (DNMOD in Lucent 5ESS just to make Dicko feel at home).

All my examples are based on North American Dial Plans. The same concept applies for Euro and LATAM etc.

Thanks, now I understand. I my opinion it’s far to complicated to accomplish what I see as a simple task but I am sure it carries a huge legacy and old technology compatabilities with it. Anyway, I will give it a try. In the mean time I installed a Grandstream HT502 which just works. All I had to do is give it a proxy, extension and password and … bingo … it just works. Unfortunately their dial plan character command set is different so I can’t just copy it to the Lincsys ATA. And I am stuck with 4 unlocked Linksys ATAs.

Oh well … eBay should solve my problem if I can’t get them to wok right? :smile:

Try setting the PAP2T dial plan to ([*x][*x].) and test. There will be a delay after dialing, but it should work. If not, there is something wrong with the config, other than the dial plan.

That is correct. And I am sure that the grandstream has a delay after you press the last digit. Try dialing an internal extension and see how long it takes to start ringing. I guess everyone’s definition of working is different. The grandstream “Handy Tones” are also complete turds. They run hot, don’t drive long cable lengths and sound like crap.

What task are your referring to? Setting up dialing plans? Your comments make it seem you don’t understand the purpose of the phones local dial plan.

If you use either of the Endpoint managers, this task is made much simpler. Every business/feature phone in the business (not just SIP phones) has a dial plan, it’s part of setting up a phone system.

Lastly if you have more than a handful of phones to provision, programming them manually is very difficult and hard to manage.

Well, I tried ([*x][*x].) and it doesn’t work. When dialing out, there is a delay and I get a busy signal and CDR log shows FAILED. At at this point I give up … not enough life left to debug these “turds”. These days I don’t even buy green bananas!
But, I very much appreciate your help and I certainly learn quite a bit.


Sorry to hear you give up so easy. Steward gave you so wrong info. Should have caught that. Try just (xxxxxxxxxxx.)

Dial a # at the end of the dial string just to speed things up.

It’s just a matter of getting the dial strings to match what you want to send.

What did you dial in the Handy Tone when it worked?

Lastly I did not call the Linksys turds, I reserve that distinction for Granstream products.

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Sorry … NO … it doesn’t work, same result, (xxxxxxxxxxx.) made no difference.
I know you did not call Linksys turds … I DID and I DO.
Anyway, if anybody wants to exchange my 4 turds for 4 Grandstream turds, I am interested. Actually I would gladly give my 4 Linksys ATAs for two Grandstream HT502.


Perhaps the obvious question has not been asked or answered, can you call from another extension the the exact same dialstring and get an answer?

Let me try to clear up a few things. I believe that the suggested dialplan would allow dialing FreePBX feature codes that begin with ‘*’. It also has a shorter delay for numbers with fewer than 10 digits. (Dialplan x matches any digit but does not match a *.)

Now, if you dial a number on a PAP2T that does not match the dial plan, the device will play a reorder tone (default sounds like a busy signal, but two beeps per second) and will not send the call to Asterisk at all, so nothing will be logged in the CDR.

If you dial a number that matches too soon (not all the digits get sent), then if FreePBX has no route to the incorrect (short) number, FreePBX will by default play “Your call cannot be completed as dialed” and log nothing to the CDR.

A ‘failed’ log generally indicates that the provider rejected the call. The OP can see in the CDR if the call was sent to the proper channel and if the correct number was sent. Using sip debug, it’s easy to see what status the provider sent and one can usually tell why this occurred.

IMHO a PAP2T is more reliable and more flexible than the Grandstream equivalents and it’s worth a little time to see what’s wrong. Possibly, there are modified settings left over from a previous use of this device; you may want to try a factory reset and re-enter the proxy, username and password.

Yes I agree. I am my company has deployed 1000 PAP2 variants going back to their original name Supera.

How about logging in to the CLI on Asterisk, turn verbosity up (core set verbose 16) and watching for the incoming digits. If nothing show on Asterisk you have a peering issue.

You should also take the time to disable all the vertical feature activation codes and everything under the provisioning tab.

Don’t be so quick to dismiss the most popular ATA in existence.

I also concur, you can buy them in bulk from China for 12 bucks a piece, individually on ebay for < $25 postage paid, the DOA percentage has been far less than 1%, as Scott says, they or their antecedents have been around since Noah and in general their dialstring syntax has become almost an industry standard for ATA’s, they are predictable and reliable, ideal for fax machines anywhere in your deployment. Keep away from Grandstream like the plague :slight_smile:


(Get the 3000 variants with an FXO if your remote sites have Alarm systems. line grabbing or otherwise, which should always have a real landline, they failsafe to a wire path between FXO and FXS and that landline can also act as your backdoor modem access into a remotely managed system)