A company that we deal with has been using VoIP for ages and loves it, so we decided it wouldn't do for us not to have the same shiny toys.  The thought of saving $100 a month made things even better.

Looking for Mango's recommended PAP2T settings?  Scroll down.

For those of you who are just joining us, the PAP2T is known as an Analog Telephone Adapter or ATA.  It allows you to use any standard telephone or PBX and route your calls over the Internet using VoIP.  VoIP is typically less expensive than a typical phone line because a VoIP provider does not have to deal with costs of infrastructure such as running phone lines to customers.

The provider that was recommended to us was DigitalVoice.ca.  (Edit: Due to subsequent service quality issues, we do not recommend this provider any longer.  We still recommend the PAP2T very highly.) We bought a PAP2T to test them out and were pleasantly surprised. We had a horrific experience with a different provider called Primus a few years ago, and we expected quality loss at the very least.  Call quality with PAP2T using the G.711 codec was actually better than an analog phone line!

The reason we like this device is that it can be configured to do practically everything but card tricks.  For example, the Linksys/Cisco dial plans are the best we've seen on any ATA.

Nifty Things to do with a PAP2T

We've discovered a few nifty things to do with this adapter that we haven't seen widely mentioned, so we decided to mention them:

You can call forward specific numbers to specific locations.  We have made a list of telemarketers' numbers and forwarded them to a special extension we set up with our PBX that plays "not in service" tones.

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.  (I foresee future geeks asking girls out like "Call me sometime!  My number is fabio@72.52.202.7:5060!") You could even configure this to support station-to-station calling for a small office or an IP-based intercom over any distance.

You can set up special rings for specific callers, so that you can tell who's calling from across the room.

The device can be configured to match the regional settings of many different countries.  The documentation for some of these settings is lacking, but plenty of good examples may be found online.

The PAP2T will talk to a syslog server and send it all kinds of information for ease of troubleshooting.  Additionally, one could use this to do fancy things such as search for a customer's record based on their phone number, and have it appear on an agent's screen.

Settings We Changed from the Default and Think You Should, Too.

The first thing we would recommend you do is upgrade the firmware of your PAP2T.  We noticed that upgrading to the latest version of firmware dramatically reduced echo.  If you do not have echo problems, you can skip this step if you like.

Most of these settings may only be set using the Advanced Administrator login.  To access this, navigate to http://[PAP2T_IP_address]/admin/advanced.

Let's start with the System tab.  Why not set an NTP server so that the date and time is always correct?  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.  You may wish to set Reg Retry Long Intvl to 120 so that your device recovers quickly if it fails registration.

The default RTP Packet Size is 0.03.  For the most popular codecs, G.711 and G.729, the optimal setting is 0.02.  (0.01 results in even less latency at the expense of using extra bandwidth, IF your provider supports it.) The default setting of 0.03 will likely cause very choppy voice with G.729 and slightly choppy voice with G.711.

On the Provisioning tab, we set Provision Enable to No as we wanted to manage the settings of our VoIP device ourselves, rather than having our provider do it.

Next, we move to the Regional tab to configure the PAP2T to properly match our region.  You may want to set the Dial Tone to 350@-19,440@-19;20(*/0/1+2) so that the user has 20 seconds with which to begin dialing instead of the default of 10.

We decided to change the Reorder Tone because unless one is listening closely and knows the difference (two beeps per second for reorder and one per second for busy) the Reorder Tone sounds a lot like the Busy Tone.  We set our Reorder Tone to 480@-19,620@-19;10(.125/.125/1+2) which plays four beeps per second and is easier to identify.  Obviously this is not a true reorder tone but for our purposes it is useful.  The reorder tone is the tone that will play when you dial an invalid number or your VoIP provider is unreachable.

The PAP2T we received shipped without a North American ring.  We were able to achieve a "normal-sounding" ring by setting the Ring1 Cadence to 60(2/4), the Ring Waveform to Sinusoid, and the Ring Frequency to 20.  We're told that these settings are standard Bellcore settings, though we've also had reports of specific telephones that respond better to a Trapezoid Ring Waveform.  Try Sinuoid first, and if your phone doesn't ring properly or you have Caller ID issues, try Trapezoid instead.  You may also need to set the Ring Voltage to 90.

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 it would actually end the call.

Because of the new North American Daylight Saving Time rules, PAP2Ts 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.)

The default gain on the PAP2T is typically too high and can cause echo.  We adjusted the FXS Port Input Gain and FXS Port Output Gain, one at a time, in increments of three.  We found that -1 and -11 eliminates echo entirely and produces appropriate conversation volume.  However, we've had reports from people who have used these settings and found the volume too quiet.  You may need to play with these until they suit you.

Let's move on to the Line tab.  As we planned to place our device behind a router, we turned on NAT Mapping and NAT Keep Alive.

We like to configure the Proxy to the IP address of our VoIP provider's server instead of the hostname.  This is because there is a little-known DNS bug with Sipura / Linksys / Cisco devices.  If the hostname is used, occasionally a DNS lookup will fail for no reason and the user will hear a reorder tone.  It may do this several times in a row, and then return to normal.  The only way we know of to avoid this issue is to use the IP address instead of the hostname.  There are some situations where this solution does not work however.  If your VoIP provider has multiple IPs for the hostname they provide to you, you should use the hostname.  To find out if this is the case, use nslookup.  (Example with multiple IPs: nslookup callcentric.com   Example with single IP: nslookup sip.us3.voip.ms)

Edited to add: We have recently been testing Linksys devices with DNS SRV records.  We were delighted to discover that we could not make the Linksys DNS Bug appear when DNS SRV records were in use.  We expect to post more about this later, however in the mean time you should use DNS SRV records if your provider supports it.  To use DNS SRV, turn on Use DNS SRV and DNS SRV Auto Prefix.  To find out whether or not your provider supports DNS SRV, use nslookup.  Callcentric supports DNS SRV and to see a sample of what a proper DNS SRV record looks like, use this command: nslookup -type=srv _sip._udp.callcentric.com

Those of you not using Windows machines can use dig instead of nslookup.

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 sooner when your ISP changes your IP address.  You can also set Proxy Fallback Intvl to 300.  If your VoIP provider's server becomes unreachable and your PAP2T fails over to a backup server, it will retry the primary server again in 300 seconds.

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.

The dial plan that ships with the PAP2T 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 described on the Linksys/Cisco Dial Plan Tips and Tricks page.

Other Stuff

We called our internet provider one day to have them make a change on our account.  We used the VoIP line, not thinking that in order to make the change, they would need to reboot our modem.  Which they did.  Which caused every device on the floor to lose its internet connection.  But when the modem came back up, our call was still connected, and we finished the conversation.  Colour us impressed.

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.

So far, we've only found a few features that this device lacks, although one really can't complain when considering the price of the PAP2T.  It can't sustain more than one G.729 conversation at one time.  But, since we use better quality G.711, (and recommend you do too) that isn't a problem for us.  The PAP2T also does not support T.38 for faxing, though faxing may be done over G.711 if one's internet connection is very stable.  Note that there are a great deal of references online that say that the PAP2T can in fact sustain two G.729 conversations, and supports T.38.  We have no idea where this information came from; we only know that both of ours definitely cannot.

The other feature that this device lacks is the ability to use a backup set of SIP credentials.  It would be great to be able to automatically fail over to another provider if one was unreachable.  We can fail over to another server but we do not know of any way of using a different username/password for it.

This is our favourite ATA.  We like it even better than other Linksys models that have routers included.  We find the performance of these routers to be sub-par.  Besides, an ATA such as the PAP2T should work perfectly fine behind a (decent) router without even any port forwarding, as long as your provider is NAT-aware.  The only other ATA we might consider is the SPA2102 if you require T.38 support.

Happy VoIPing!