Configure your Linksys VoIP ATA the right way!
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.)
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: ( 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.
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: 126.96.36.199
Secondary DNS: 188.8.131.52
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.
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.