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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.
Last update: June 18, 2011.
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.
If your Linksys or Sipura ATA is locked to your former VoIP provider, why not have your ATA
permanently unlocked so that it may be safely factory reset, and may be used with any provider you
like? Contact DogFace05 for more information.
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. You should also set Reg Retry Long Intvl to 600 so that your
device recovers quickly if it does fail.
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 "normalsounding" ring by setting the Ring Waveform to Sinusoid. You may also need to set the Ring
Voltage to 90. If your phone gives a half-ring for incoming calls and then disconnects, instead set
Ring Voltage to 70. (Thanks Stewart!)
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.)
In some situations, the default gain settings are inappropriate. This can cause audio to be distorted, it
can cause echo, and of course cause audio to be too quiet. If this is the case, you can adjust FXS Port
Input Gain and/or FXS Port Output Gain, one at a time, in increments of three. However, you
should only adjust these if you are experiencing a problem. Note: Input Gain = how you sound to the
other party. Output Gain = how the other party sounds to you.

Line tab
As we planned to place our device behind a router, we turned on NAT Mapping and NAT Keep

Alive.
If you hear a reorder (fast busy) tone occasionally when you attempt to make calls, it is likely because
your VoIP provider does not respond to the ATA's SIP NOTIFY message. To work around this, set
NAT Keep Alive Msg to KeepAlive. However, you should only do this if you are experiencing the
problem.
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
The technique for setting up Visual Call Waiting is somewhat involved. Here's how to do it: On the
Line tab, be sure that Call Waiting Serv is set to "yes". Next, go to the Regional tab. You need to set
up four activation codes. If they're already set up, then that's fine, just make a note of them. If there is
no code listed, make one up (that is not already in use on that page) and type it in. The four features

that require activation codes are: CW Act Code, CW Deact Code, CWCID Act Code, and finally
CWCID Deact Code. Note that these codes must all be different. We used *56, *57, *58, and *59. It
doesn't matter what you use as long as you remember it, and as long as the code is not already in use
for some other feature. Save your changes and wait for the device to restart. Pick up your telephone
and dial your Call Waiting Activation Code. Wait for the dial tone and then dial your Call Waiting
Caller ID Activation Code. Visual Call Waiting is now ready for use.
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, SIP Helper, Stateful Packet Inspection, and/or SPI you should
disable them. Otherwise, your VoIP equipment will end up in a lake soon. These are infamous for
causing more problems than they solve (if they solve any at all).
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.
Happy VoIPing!
Comments (38) Leave a comment
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
July 2nd, 2009 at 08:24 | #2
Good write-up Mango. I understand you are using voip.ms. Do you know how the "Callback"
feature works? When reading the instructions on voip.ms, it indicates that the DID needs to be
forwarded to the Callback number. This makes no sense, because if I forward all calls to the
callback number, they will get a busy signal and then my cell phone will ring giving me a dial
tone when I answer? That's how I understood it to work from the instructions. Do you use this
feature?
Thanks.

3.
Mango
July 2nd, 2009 at 09:45 | #3

Hey
I don't use the callback feature. I think it's intended for people who have a cell phone. (I don't.)
They could route a DID to the callback, call the DID with their cell phone, have it call their cell
phone back, and then make their call. This way, they could make calls at VoIP.ms rates rather
than cell phone rates. It would be a good solution for people who have unlimited incoming
plans.
The downside is that you need to order another DID to use this feature. You can't use your
primary DID because, as you mentioned, nobody would be able to call you. This feature could
use some work. It would be more useful if you could route calls to the callback feature using
Caller ID filtering or an IVR - this way you would only need one DID. Additionally, this
feature will not work with pay phones because you do not know in advance the number of the
pay phone that you will use. It would be very convenient if VoIP.ms had a toll-free access
number that one could dial from a pay phone, enter their account number and a password, and
be presented with a dial tone. Even if we had to pay for both origination and termination, it
would certainly be cheaper than the $1 that Telus charges for a calling card call.
Hope this helps!
m.

4.
Mail
August 2nd, 2009 at 18:01 | #4
Hey Mango ...... please give us more details on "Calls can be made to another VoIP device
without either device even having a VoIP provider, both on your LAN and through the
internet...absolutely free."
Thank you

5.
Mango
August 2nd, 2009 at 18:06 | #5
Sure It's actually very easy. What you need to do is set the options for (Line tab) Make Call
Without Reg and Ans Call Without Reg to Yes. Then, you can set the adapters up to call each
other by adding the following to your speed dial: <User ID@IP address:SIP Port>
If the adapters are each behind separate routers, you would need to mess around with port
forwarding to get it to work properly; I suspect you would need to forward the port specified in
(Line tab) SIP Port on the ATA, and also the port range specified in (SIP tab) RTP Port Min
through RTP Port Max.
m.

6.
priller

August 3rd, 2009 at 06:12 | #6

Here are some references that validate your observations:
The PAP2 and the PAP2T do NOT support the T.38 fax protocol.
http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a35ffb.shtml
The PAP2 or PAP2T does not support usage of simultaneous G.729 codec.
http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a35d91.shtml

7.
Anonymous
September 20th, 2009 at 05:49 | #7
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?
Also, I've configured the device as per your instructions and am also using VOIP.MS. I am
impressed with the quality of service, but, one very small issue. My Panasoic cordless phone
seems to generate just a hint of white noise when I am on a call. It's only perceptible to me...
Have you noticed this kind of issue with a cordless phone too?
When I make a call on my corded phone, the connection is definitely clearer... So I'm thinking
the issue is more on the cordless phone side.
Lastly, have you tried any cordless IP phones?
Cheers, bp

8.
Mango
September 20th, 2009 at 08:11 | #8
According to Wikipedia (http://en.wikipedia.org/wiki/Linksys_PAP2), the SPA-2000 is the
same as the PAP2, not the PAP2T.
I do not use any sort of cordless phones. But, here are some suggestions about the issue:
- If you use wireless internet, try moving the cordless phone's base unit as well as the phone
itself as far away as possible (like in a different room) from the wireless router and see if that
changes anything.
- Be sure that the setting (on the Line tab) for "Silence Supp Enable" is turned off.
- Try changing the cable that goes to your cordless phone's base station. This issue can happen
with very long or poor quality cables.
- The other thing you could try is using the second line port of the SPA-2000. You mentioned

the issue does not occur with a corded phone, so the problem is probably not the SPA-2000 but
it is possible.
Unfortunately, some cordless phones do have a slight noise in the background. But, if you do
solve it, I would be interested in hearing how you did it!

9.
foobysmacker
October 11th, 2009 at 09:31 | #9
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.

10.
Mango
October 11th, 2009 at 10:09 | #10
Good tip except it's in the Account Settings or Manage Sub Account area, not DID settings

11.
Corky Romana
December 2nd, 2009 at 01:22 | #11
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.

12.
Mango
December 6th, 2009 at 18:43 | #12
I am very glad to hear that! Thank you for reading

13.
Chris
December 24th, 2009 at 21:55 | #13
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

14.
Robert
January 5th, 2010 at 17:35 | #14
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.

15.
Mango
January 5th, 2010 at 17:45 | #15
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.)

16.
jackie999
March 19th, 2010 at 12:34 | #16
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...

17.
Christopher
March 20th, 2010 at 07:59 | #17
The callback feature with Voip.ms has recently been mad much more useful. I setup a SIP DID
(0.25/min). Then setup a CallerID filtering rule to send any calls from my cellphone to an IVR.
But this filtering rule is defined to ONLY apply to my main DID. This IVR has a number of
entries (press 1 to ring all lines, press 102 to call extension 2, press 103 to call extension 3...
[lots you can do], but then press 5 for callback. The callback actually directs the call to my SIP
DID. Now I've defined another filtering rule for calls from my cell phone that applies only to
calls received on my SIP DID. This filtering rule routes the call to the callback function. I have
a callback rule that rings my cellphone.
So if I call home and want to speak with someone, I use the IVR to direct me to an appropriate
extension (or to ring all lines). If I want callback, I press 5, get busytone and then hangup. Soon,

my cell will ring and give me dialtone - after which I can dial any number and get billed by
voip.ms for LD.
I was at the train station last night to pick someone up. While waiting, I called my daughter (in
Canada) using this scheme. Since I have free cell calling after 6pm, a 30 minute call like this
cost me only 15 cents. SWEET!

18.
Christopher
March 20th, 2010 at 08:10 | #18
Sorry above when I said SIP DID 25 cents /min - I meant 25 cents/month! Also to clarify, the
announcement that says press 5 for callback - does not actually use the callback feature, but
routes the call to my SIP DID. The filtering rules on the SIP DID then send the call to the actual
callback function. Convoluted - but it works! I also us my SIP DID for a bunch of other stuff so no extra cost for me. (Check out http://www.ipkall.com - can route to your SIP DID - and for
those of us in Canada gives a way to get Google voice working)

19.
Rainy Roamer
March 20th, 2010 at 18:23 | #19
Where is that "donate" button anyway?!

20.
Protos
August 3rd, 2010 at 12:15 | #20
Very useful information. Thanks kindly for providing it. Really helped me get the most from my
Sapura/Linksys/Cisco SPA 2102.

21.
Ian
August 9th, 2010 at 19:22 | #21
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

22.
Audrey Gozon
August 17th, 2010 at 13:48 | #22

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

23.
Mango
August 17th, 2010 at 13:54 | #23
That's great to know! Thank you very much for writing in.

24.
Robert March
September 5th, 2010 at 00:06 | #24
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.

25.
Chris Thomas
October 12th, 2010 at 05:16 | #25
Thank you for taking time to provide such useful info for beginners like me - and so clearly
explained.

26.
Lee
November 28th, 2010 at 00:58 | #26
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 SPA921s on default settings. Thanks a lot, this info. potentially saved me a client.

27.
Rick
December 16th, 2010 at 19:09 | #27
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!

28.
Mango
December 16th, 2010 at 19:13 | #28
We're very happy to hear it! Thanks for writing in

29.
George
January 9th, 2011 at 09:13 | #29
Thanks Budd ! this post helped me a lot with my PAP2T to get rid off those choppy voice.

30.
Wendy
January 9th, 2011 at 15:13 | #30
Oh wow. Thanks! I'm a total beginner and this is *exactly* what I needed. Phone's working
much better now.

31.
helpdeskdan
March 9th, 2011 at 17:24 | #31
Thanks man, I was looking everywhere for the correct Daylight Saving Time Rule!

32.
Tony
March 16th, 2011 at 10:23 | #32
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.

33.
Mike
March 23rd, 2011 at 12:20 | #33
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.

34.
D
April 14th, 2011 at 00:55 | #34
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.

35.
Alphonse
April 15th, 2011 at 07:41 | #35
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.

36.
Leo
May 7th, 2011 at 05:08 | #36
Awesome... Just new to voip.ms and just got PAP2T, working well, and even better with these
settings... Thanks!

37.
Scott Jordan
June 25th, 2011 at 19:38 | #37
Mango, thanks for your informative posts. I have recently started a blog to memorialize cool
tricks that friends have expressed interest in, http://www.unvexed.blogspot.com --and I've
linked to this post (and to your review of the PAP2T-NA and siblings) so others can gain the
benefit of your expertise. Again, many thanks for your helpful and well-written posts.

38.
Mike
June 28th, 2011 at 20:03 | #38
Another security setting to consider is "Restrict Source IP" on the Line tab. Set this to "yes" if
you do not intend to make/receive direct IP calls. This (after SIP registration) will ignore calls
from everything except the IP address you are registered to.

I got recommended this setting by Cisco: they correctly concluded that my ATA was crashing
due to brute force user/password guessing attempts overloading it. (I changed the SIP port from
5060 to a different port too, as Mango recommends).
Also (if you have a choice) the choice of a non-numeric User ID and a strong Password will
make brute force user id / password guessing harder.

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