Configure your Linksys VoIP ATA the right way!

March 20th, 2009 Leave a comment Go to comments

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 Cisco.com.) 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 1.pool.ntp.org and 2.pool.ntp.org.

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: 8.8.4.4
Secondary DNS: 208.67.222.222
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.

 
  1. Anonymous
    June 25th, 2009 at 13:50 | #1

    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.
     

  2. Anonymous
    September 20th, 2009 at 05:49 | #2

    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? 
     

  3. September 20th, 2009 at 08:11 | #3

    According to Wikipedia (http://en.wikipedia.org/wiki/Linksys_PAP2), the SPA-2000 is the same as the PAP2, not the PAP2T.

    m.  ๐Ÿ™‚
     

  4. foobysmacker
    October 11th, 2009 at 09:31 | #4

    I have the PAP2T-NA and voip.ms.  To get inbound calls to ring to your PAP2T instead of giving a busy signal you should check you main DID settings on your voip.ms account.  By default it is set to IP-PBX but should be changed to ATA Device.
     

  5. October 11th, 2009 at 10:09 | #5

    Good tip except it's in the Account Settings or Manage Sub Account area, not DID settings ๐Ÿ˜‰
     

  6. Corky Romana
    December 2nd, 2009 at 01:22 | #6

    Mango, your recommendations for setting up the PAP2T really helped me a lot.  I just switched from voip.ms and am happy so far. 

    Thanks for sharing your wisdom, great work.
     

  7. December 6th, 2009 at 18:43 | #7

    I am very glad to hear that!  Thank you for reading ๐Ÿ™‚
     

  8. Chris
    December 24th, 2009 at 21:55 | #8

    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 voip.ms and may try use FPL if I can figure it out.

    Anyway Happy Xmas to all
     

  9. January 5th, 2010 at 17:35 | #9

    Hi...I just migrated from Vonage to Voip.ms but there is one feature I loved with Vonage that viop.ms 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.
     

  10. January 5th, 2010 at 17:45 | #10

    VoIP.ms 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.)
     

  11. jackie999
    March 19th, 2010 at 12:34 | #11

    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...
     

  12. March 20th, 2010 at 18:23 | #12

    Where is that "donate" button anyway?!
     

  13. Protos
    August 3rd, 2010 at 12:15 | #13

    Very useful information. Thanks kindly for providing it. Really helped me get the most from my Sapura/Linksys/Cisco SPA 2102.
     

  14. Ian
    August 9th, 2010 at 19:22 | #14

    Good information and very useful ! I have a 514 DID with myowntelco.net and I'll try to forward it right away and see how it goes..

    Cheers
     

  15. Audrey Gozon
    August 17th, 2010 at 13:48 | #15

    I am a linksys Technician and I say this article is very informative.
     

  16. August 17th, 2010 at 13:54 | #16

    That's great to know!  Thank you very much for writing in.
     

  17. Robert March
    September 5th, 2010 at 00:06 | #17

    Thank you for taking the time to post this very useful information for us. 

    As an aside, I use voip.ms and Acanac for VOIP.  voip.ms beats Acanac on voice quality, service, and price.
     

  18. Chris Thomas
    October 12th, 2010 at 05:16 | #18

    Thank you for taking time to provide such useful info for beginners like me - and so clearly explained.
     

  19. Lee
    November 28th, 2010 at 00:58 | #19

    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 SPA921ยดs on default settings. Thanks a lot, this info. potentially saved me a client.
     

  20. Rick
    December 16th, 2010 at 19:09 | #20

    Hi,
    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)
    Thanks!
     

  21. December 16th, 2010 at 19:13 | #21

    We're very happy to hear it!  Thanks for writing in ๐Ÿ™‚
     

  22. George
    January 9th, 2011 at 09:13 | #22

    Thanks Budd ! this post helped me a lot with my PAP2T to get rid off those choppy voice.
     

  23. Wendy
    January 9th, 2011 at 15:13 | #23

    Oh wow. Thanks! I'm a total beginner and this is *exactly* what I needed. Phone's working much better now.
     

  24. helpdeskdan
    March 9th, 2011 at 17:24 | #24

    Thanks man, I was looking everywhere for the correct Daylight Saving Time Rule!
     

  25. Tony
    March 16th, 2011 at 10:23 | #25

    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
    http://www.pool.ntp.org/zone/ca Enter the NTP server names on the WAN page.
     

  26. Mike
    March 23rd, 2011 at 12:20 | #26

    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.
     

  27. D
    April 14th, 2011 at 00:55 | #27

    Thanks for the detailed configuration options I spent 2 days on line with voip.ms 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.
     

  28. Alphonse
    April 15th, 2011 at 07:41 | #28

    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.
     

  29. Leo
    May 7th, 2011 at 05:08 | #29

    Awesome... Just new to voip.ms and just got PAP2T, working well, and even better with these settings... Thanks!
     

  30. Martin
    October 6th, 2011 at 00:39 | #30

    Hi Mango,

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

    Martin
     

  31. Chris Thornton
    November 23rd, 2011 at 08:22 | #31

    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!!

    Cheers!
     

  32. Steve
    December 4th, 2011 at 16:26 | #32

    This is by far the best voip instruction online. Thanks so much for fixing everything that my provider had no idea about.
     

  33. December 16th, 2011 at 18:46 | #33

    I'm glad to help!  Maybe you could mention my blog to your provider ๐Ÿ˜‰
     

  34. Darlene
    December 28th, 2011 at 07:08 | #34

    Thanks so much , as a newbie to all of this, it was EXACTLY what I needed!
    wanted to thank you for your help
     

  35. January 23rd, 2012 at 07:36 | #35

    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.

    Thanks.
     

  36. Richard
    February 9th, 2012 at 09:40 | #36

    Great information! Setup VOIP on PAP2T works right away! Thanks!
     

  37. Matthew Walters
    February 28th, 2012 at 05:29 | #37

    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
     

  38. March 2nd, 2012 at 07:36 | #38

    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 voip.ms because of impossibility to place a call etc. now it seems to work fine.

    THANKS
     

  39. David
    March 4th, 2012 at 07:22 | #39

    Thanks, very helpful
     

  40. Alexandre Trepanier
    April 11th, 2012 at 18:05 | #40

    GREAT !!!!
     

  41. Darren
    May 18th, 2012 at 05:09 | #41

    What an awesome article, this really made a difference on our PAP2T/Callcentric.  Thanks!!
     

  42. June 14th, 2012 at 19:50 | #42

    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! ๐Ÿ™‚
     

  43. patrick
    August 6th, 2012 at 11:28 | #43

    Hi,

    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!
     

  44. Steve St-Pierre
    December 14th, 2012 at 09:51 | #44

    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 !!!!
     

  45. James
    January 6th, 2013 at 22:49 | #45

    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!
     

  46. rattini
    January 25th, 2013 at 13:20 | #46

    Thank you so much!
    Without you, no chance for me.
     

  47. James Reed
    August 31st, 2013 at 22:56 | #47

    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.
     

  48. September 1st, 2013 at 05:40 | #48

    You are very welcome.  ๐Ÿ™‚
     

  49. ItaMangM
    September 27th, 2013 at 15:52 | #49

    Thank you! Your instructions are great also after all this time!
    Great Mango!
     

  50. Alex
    November 25th, 2013 at 21:26 | #50

    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
     

  51. Anonymous
    June 29th, 2014 at 11:01 | #51

    Like PAP2.
    Like this post
     

  52. June 29th, 2014 at 11:03 | #52

    Now there's a ringing endorsement!
     

  53. Telecom-Wizard
    November 28th, 2014 at 00:11 | #53

    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 !
     

  54. John
    August 21st, 2016 at 15:49 | #54

    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 Voip.ms 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!
     

  55. G. Rosch
    March 11th, 2017 at 06:43 | #55

    Extremely useful text. I was looking all over for this information. Very helpful to the still growing PAP2T community worldwide.
     

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