Configure your Linksys VoIP ATA the right way!

March 20, 2009
ATAs made by Linksys (formerly Sipura) are arguably the most popular ATAs amongst consumers and small businesses, because of their wide array of configuration options. However, their default settings are not appropriate for users in Canada and the USA. Let's talk about some settings you can use to ensure that your VoIP equipment properly matches your region. We apologize in advance for Mango's verbosity but truly feel that the information in this article is very important. If you're in a hurry, read the bold parts, and the last three paragraphs labeled important note. Please note: Though the Linksys adapters have historically been an industry standard, they are built on old technology. If you are considering buying a VoIP ATA, we recommend the new OBi ATAs. The OBi ATAs were built by the same engineers that built the PAP/SPA devices, and are sure to make an already great VoIP experience even better. We do not recommend the new SPA112 and SPA122. Though the SPA112 and SPA122 are sold by Cisco as successors to the PAP2T and SPA2102, the new devices are not built on the same technology as the old devices and have received poor reception in the community. Last update: August 6, 2012. If your ATA has -NA after its model number or it has been permanently unlocked or "NAized", you might want to upgrade its firmware - we noticed that upgrading to the latest version of firmware dramatically reduced echo. (You can find the latest version of the firmware at Also, you can reset your ATA to its factory settings to have a clean slate to start from. To do this, connect a phone to your ATA and dial ****73738#. It's okay if you don't hear a dial tone. Then, dial 1 to confirm. Most of these settings may only be set using the Advanced Administrator login. To access this, navigate to http://[ATA_IP_address]/admin/advanced. Not sure what your device's IP address is? Pick up your phone and dial ****110# and a friendly voice will tell you. (But don't do that after 10PM or you'll wake him up.)

Let's start our configurations with the System tab.

We suggest you set an Admin Passwd to protect you from an unauthorized user accessing your device. It's best practice to keep your VoIP hardware behind a firewall unless you have a really good reason not to, so using a password isn't really necessary, but it's useful to prevent "Oops!" situations. Why not set an NTP server so that the date and time that appears on your Caller ID is always correct? Additionally, some users have reported that this is required in order for the device to automatically adjust for Daylight Saving Time. You may do this on the System tab. Try and

Let's move on to the SIP tab.

The default SIP timers are too aggressive. You should set SIP T1 to 1 to mitigate a problem that causes the ATA to fail to register. For the most popular codecs, G.711 and G.729, the optimal RTP Packet Size setting is 0.02. The default will likely cause very choppy voice with G.729 and slightly choppy voice with G.711.

Next, we move to the Regional tab

Let's configure the ATA to properly match our region. You have just 10 seconds to begin dialing after lifting your handset. Why not increase this to 30 seconds? Set the following: Dial Tone: 350@-19,440@-19;30(*/0/1+2) Second Dial Tone: 420@-19,520@-19;30(*/0/1+2) Outside Dial Tone: 420@-19;30(*/0/1) MWI Dial Tone: 350@-19,440@-19;2(.1/.1/1+2);30(*/0/1+2) Cfwd Dial Tone: 350@-19,440@-19;2(.2/.2/1+2);30(*/0/1+2) The ATA we received shipped without a North American ring. We were able to achieve a "normal-sounding" ring by setting the Ring Waveform to Sinusoid and the Ring Frequency to 25. If you use an answering machine, instead of voicemail provided by your VoIP provider, you should set Reorder Delay to 15. You may want to set the CPC delay to 10 and the CPC duration to 0.5. With the default settings, our phones had to be on the hook for an inordinate amount of time before the device would actually end the call. Because of the new North American Daylight Saving Time rules, ATAs by default calculate DST incorrectly. Also on the Regional tab, set your Daylight Saving Time Rule to start=3/8/7/2:00;end=11/1/7/2:00;save=1 and your time zone appropriately for your region. (Trivia: 3/8/7/2:00 translates literally to "The Sunday that is on or after March 8th at 2AM." The second parameter is commonly misunderstood as the week, however this is not correct.)

Line tab

As we planned to place our device behind a router, we turned on NAT Mapping and NAT Keep Alive. You should also set Register Expires to 300 to avoid "phone doesn't ring" issues. Among other things, this will let your VoIP provider know within five minutes when your ISP changes your IP address. We mentioned DNS failover before. On that topic, if your provider supports DNS SRV, you should use it. This allows the provider to specify the priority and weight of multiple SIP switches. If one is down or otherwise unreachable, your device will immediately try the next one in the list. If DNS SRV is supported, you should set Use DNS SRV to Yes. However, if your provider does not publish DNS SRV records, this will make the device not work. You can just try it; if it doesn't work, change it back. We configured the Preferred Codec to be G.711u because we had the bandwidth available and were very pleased with its quality. (Trivia: Though G.711 is a 64Kbit codec, it actually uses about 90Kbit/sec due to overhead.) We tested a few codecs and have samples available comparing G.711 vs. G.729 and also VoIP sound quality vs. an Analog Phone. Sometimes, one party or the other will hear sounds as if a digit on the phone had been pressed. This is often caused by "talk-off". Here is how you may solve it: To solve you hearing sounds as if the other party had pressed a digit, try to set DTMF Process INFO and DTMF Process AVT to No. If you require remote access to a physical answering machine, be sure to test this after making these changes. To solve the other party hearing sounds as if you had pressed a digit, set DTMF Tx Method to InBand. You may also need to set this on your VoIP provider's control panel. The Dial Plan that ships with the ATA isn't particularly useful. This causes many a forum post that goes something like, "whenever I make a call, it takes ten seconds to start ringing!" Our favourite dial plan is: ( [23456789]11 | *xxx. | <:1>[2-9]xx[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011xxxxxxx. | [#*x][*x][*x][*x][*x][*x][*x][*x][*x][*x][*x][*x]. ) More dial plans are described on the Linksys Dial Plan page. You should change the SIP Port from its default of 5060. It is our opinion that obscurity is an important part of security, and if your equipment isn't behind a firewall, using a random port number will drastically reduce the number of brute force attacks you receive. Additionally, the SIP Port on Line 1 and Line 2 should be different. Sometimes it will work if they are both the same but there are certain situations when it will not, so we just change the Line 2 SIP Port to something unique as a matter of course.

Other stuff

Important note #1: Did you buy your ATA on eBay? Are you having problems? Uh-oh. eBay is notorious for fake ATAs. Check this thread at DSLReports for tips on how to spot a counterfeit Linksys ATA. Important note #2: This isn't strictly related to Linksys ATAs, however it is no less important: if your router has options for SIP ALG or SIP Helper you may wish to disable them, if things are not working properly. These are infamous for having poor implementations and often cause more problems than they solve. Important note #3 to users of Tomato firmware: since many users of Linksys ATAs have a router with Tomato firmware, we feel it prudent to mention a bug that prevents VoIP devices from working properly upon reboot of the router. To work around the bug, simply enter Conntrack/Netfilter and set Unreplied UDP Timeout to 10. Note that some versions of Tomato have a SIP Helper. You MUST disable it on the Conntrack/Netfilter page. (See note #2 above.) Important note #4 to users of 2Wire routers: we have had reports of some 2Wire routers that fail to respond to DNS requests when a computer is not in use. To work around this, navigate to the System tab in your ATA and set the following: Primary DNS: Secondary DNS: DNS Server Order: Manual, DHCP Important note #5 if you use VoIP ATAs with an old analog PBX, such as a Norstar or Meridian: if incoming calls disconnect when you attempt to answer them, it is possible that the disconnect supervision setting on your PBX is too sensitive. Increase it. Happy VoIPing!
If you are having technical problems with your VoIP equipment, we recommend you post in the VoIP Tech Chat Forum. Due to the volume of questions we receive, we cannot answer them all. However, Mango loves to read comments about this or any article. He is also an active participant in the forum, and will be delighted to help you there if he can.
June 25, 2009
Awesome. this is exactly what i was looking for. For a long time i couldn't figure out why the ring was not the right cadence. Your suggestion of changing from Trapezoid to Sinusoid did the trick.
September 20, 2009
Hi Mango, I actually have an older SPA-2000 with firmware revision 2.0.9(d). Just curious if you knew if the PAP is essentially the same device?
September 20, 2009
According to Wikipedia (, the SPA-2000 is the same as the PAP2, not the PAP2T.

m. :)
October 11, 2009
I have the PAP2T-NA and To get inbound calls to ring to your PAP2T instead of giving a busy signal you should check you main DID settings on your account. By default it is set to IP-PBX but should be changed to ATA Device.
October 11, 2009
Good tip except it's in the Account Settings or Manage Sub Account area, not DID settings ;)
Corky Romana
December 2, 2009
Mango, your recommendations for setting up the PAP2T really helped me a lot. I just switched from and am happy so far.

Thanks for sharing your wisdom, great work.
December 6, 2009
I am very glad to hear that! Thank you for reading :)
December 24, 2009
Hi Great resource just starting out with a PAP2T was having a bit of trouble with dial plans (well still am exploring) got calls in out with name and number. Going to try a few tweaks from here then leave it alone So I can make all my xmas calls. I am using and may try use FPL if I can figure it out.

Anyway Happy Xmas to all
January 5, 2010
Hi...I just migrated from Vonage to but there is one feature I loved with Vonage that doesn't seem to have. That feature is what Vonage called simultaneous ring. More than just forwarding, it was a simultaneous ring to any other number(s) I wanted. It worked great. Is there a way to program another number to be simultaneously dialed in the PAP2? Thanks.
January 5, 2010 does indeed have simultaneous ring - they call it "Ring Groups". It's in the "DID Numbers" menu.

If you want to ring some other phone such as a cell phone you must first set up a Call Forwarding entry to the cell phone. Note that you will be billed for both legs of the call (unless you have a flat rate DID plan in which case you will only be billed for one leg.)
March 19, 2010
This review is a goldmine of information! I have made many of the changes..some I didn't since I'm too new.
Thanks SO much...
Rainy Roamer
March 20, 2010
Where is that "donate" button anyway?!
August 3, 2010
Very useful information. Thanks kindly for providing it. Really helped me get the most from my Sapura/Linksys/Cisco SPA 2102.
August 9, 2010
Good information and very useful ! I have a 514 DID with and I'll try to forward it right away and see how it goes..

Audrey Gozon
August 17, 2010
I am a linksys Technician and I say this article is very informative.
August 17, 2010
That's great to know! Thank you very much for writing in.
Robert March
September 5, 2010
Thank you for taking the time to post this very useful information for us.

As an aside, I use and Acanac for VOIP. beats Acanac on voice quality, service, and price.
Chris Thomas
October 12, 2010
Thank you for taking time to provide such useful info for beginners like me - and so clearly explained.
November 28, 2010
Hey, very informative. I am a VOIP provider and have just gone through a lot of pain trying to work out why one my clients is taking so long to connect a call and why they cant tell the difference between an engaged tone and a number mis-dialed/doesnt exist tone. They are using Linksys SPA921´s on default settings. Thanks a lot, this info. potentially saved me a client.
December 16, 2010
My answering machine wasn't working for 2 years. I didn't bother to look into why. I have Linksys PAT2T.
Read your post. Changed 'Reorder Delay' to 15. Voila!
Answering machine works! (also changed Voltage to 90; Ring waveform to Sinusoid)
December 16, 2010
We're very happy to hear it! Thanks for writing in :)
January 9, 2011
Thanks Budd ! this post helped me a lot with my PAP2T to get rid off those choppy voice.
January 9, 2011
Oh wow. Thanks! I'm a total beginner and this is *exactly* what I needed. Phone's working much better now.
March 9, 2011
Thanks man, I was looking everywhere for the correct Daylight Saving Time Rule!
March 16, 2011
Thanks very much! In particular the info explaining the variables in the daylight savings time string start=3/8/7/2:00;end=11/1/7/2:00;save=1 solved my problem with DST.

For Canadian users, ntp servers listed here Enter the NTP server names on the WAN page.
March 23, 2011
The trapezoid to sinusoid did the trick. However, I experimented with the voltage. I lowered it to 70 after reading in another forum that the sipura 2102 could overload when trying to output near 90 volts. i dealt with the default 85v and warbling ring for 2 years. Once at 70v, the trapezoid waveform worked too.
April 14, 2011
Thanks for the detailed configuration options I spent 2 days on line with trying to fix the it takes ten seconds to start ringing thing with the end result being it then taking 12-14 seconds to start ringing. Then I stumbled upon you post and made the changes you suggested and all is fine now I get a ring in 2 seconds great work thanks again.
April 15, 2011
I'd like to echo this one: "first thing we would recommend you do is upgrade the firmware of your ATA." My recently purchased PAP2T ATA came with firmware 3.15. It didn't even have the 5.1.6 version, which dates from 2007. I thought I would leave well enough alone and not try to fix what wasn't broken, but within a day I was experience periods when I couldn't call in to my number. Upgrading to 5.1.6 cleared this up.
May 7, 2011
Awesome... Just new to and just got PAP2T, working well, and even better with these settings... Thanks!
October 6, 2011
Hi Mango,

Thanks for the very informative blog. Keep up the good work!

Chris Thornton
November 23, 2011
Excellent information! We are deploying SPA8000s to business customers to allow them to go VOIP without changing their familiar phone system (and offering fully managed network / IT service to boot!). Your site has been extremely helpful in solving some of the nitty issue that have come up in dealing with the key systems / PBXs. Please keep up the good work!!

December 4, 2011
This is by far the best voip instruction online. Thanks so much for fixing everything that my provider had no idea about.
December 16, 2011
I'm glad to help! Maybe you could mention my blog to your provider ;)
December 28, 2011
Thanks so much , as a newbie to all of this, it was EXACTLY what I needed!
wanted to thank you for your help
Rémy Roy
January 23, 2012
Thank you so much. You are currently the only proper guide I found to configure my PAP2T on th web for North America and I search for many hours.

February 9, 2012
Great information! Setup VOIP on PAP2T works right away! Thanks!
Matthew Walters
February 28, 2012
What a lucky man I am, stumbling upon this post!

Setting SIP T1 to 1 (previously had it at .5) solves my problem of the second SIP service on line two not ringing during incoming calls.

I use the excellent NodePhone service. Unfortunately the PAP2T, is not supported by Internode here in Australia. My PAP2T is still running software version 3.1.15(LS)

Internode's advice for the Sipura 3000 plus your T1 suggestion, makes the PAP2T work very gracefully. No delay to start ringing, good volumes, good clarity.

Works behind a Billion VGP 7402 modem/router, and also an Optus Cable Modem
March 2, 2012
AWESOME!!! Thanks man! This is the ultimate guide I saw on Linksys ATAs... It took me minutes to configure this in right way. I was putting pressure on because of impossibility to place a call etc. now it seems to work fine.

March 4, 2012
Thanks, very helpful
Alexandre Trepanier
April 11, 2012
GREAT !!!!
May 18, 2012
What an awesome article, this really made a difference on our PAP2T/Callcentric. Thanks!!
June 14, 2012
A bit late on the uptake, but wow - this is one of the most complete and most accurate config guides I've seen for PAP2 - and I do this for a living... Nice! :)
August 6, 2012

many THX for this instruction!
as the web interface of the PAP2T really IS complex, it is really good to have a nice guide through the most important customisations!

keep up the cool work, many thx from germany!
Steve St-Pierre
December 14, 2012
I just wanted to Thank you so much for writing this tutorial, it help me in great way.

Thanks again for sharing your experience and knowledge with the community.

Keep up the great work !!!!
January 6, 2013
I reference this page every time I reset my ATA or when configuring one for a friend. As part of my new year's resolution, I'm making an effort to leave my thanks on your helpful article. THANKS MANGO!
January 25, 2013
Thank you so much!
Without you, no chance for me.
James Reed
August 31, 2013
Thank you for taking the time to figure this out and share it with all of us. I just switched my VoIP provider and found this page highly recommended. You just made my new VoIP service feel truly shiny and new. Thank you.
September 1, 2013
You are very welcome. :)
September 27, 2013
Thank you! Your instructions are great also after all this time!
Great Mango!
November 25, 2013
WOW, thank you so much for this article! I always had problem and line dropping with my VoIP and since 2-3 weeks my phone kept bugging "never ringing, always occupied, etc" over 5 time a day! I was about to throw my PAP2T in the garbage but this whole guide gave it a brand new life and saved me a new unit at 50$!

I'm curious though to know how come the "newer" SPA 112 is actualy worse than this discountinued PAP2T O_o
June 29, 2014
Like PAP2.
Like this post
June 29, 2014
Now there's a ringing endorsement!
November 28, 2014
For those who would like a vintage dialtone like in the old day , you can try this one wich will remember some Northern Crossbar dialtone : 520@-3,410@-3;20(*/0/1+2)

i will soon post a ringback vintage tone as well !
hope you enjoy !
August 21, 2016
Many thanks for your public spirited article on configuring the PAP2 (I have the PAP2T) for North America. Have found it invaluable when origionally getting my system working a few years ago in Toronto, then again recently when I relocated to British Columbia. In both instances the wiki didn't quite do it, but your advice worked like a charm! Many people would have kept such information to themselves, but you have shared it making hundreds happy!
G. Rosch
March 11, 2017
Extremely useful text. I was looking all over for this information. Very helpful to the still growing PAP2T community worldwide.
Pascal Gienger
October 24, 2017
Like to beamed back into the past :)
The Cisco SPA112 and 122 got much improvements, with firmware 1.4.1SR1 (August 2017) they run rock solid and you can get them for under $15 in may places.
October 24, 2017
The newer SPA ATAs may be solid, but Obihai ATAs are solid too, and are significantly more powerful in terms of call routing features and protocols supported. The extra cost of an OBi is negligible when you consider the savings of using VoIP.