The Michigan Telephone blog

In yesterday’s installment of this review of the Atcom AG-188N (sold in North America by CIGear), I had mentioned that you could download the documentation from the manufacturer’s web site in .PDF format. Since then I have come to realize that the .doc files on the CD are actually a bit more complete – some options are covered there that are not covered in the PDF version. Perhaps someone at Atcom should make the effort to generate new online PDF files from the current documentation.

Also, today I found out that because I didn’t have the latest firmware, there are more CODEC options on the Audio Settings page than what I had pictured – you can now select up to five CODECs, in order of preference. I’ve uploaded an additional screenshot to yesterday’s article, so you can see what’s changed.

Yesterday, we pretty much covered IAX settings that would allow you to receive incoming calls on the AG-188N, but if you’d like the correct time to show on your CallerID display, you probably want to click on the Time Config selection in the left-hand menu, to access this page:

Atcom AG-188N Time configuration screen

This one is pretty self-explanatory, but note one thing: The time server must support Simple Network Time Protocol a.k.a. SNTP. Some time servers only support NTP. When I tried to use the time server on the Asterisk system, which does NTP only, I got no time information with my incoming calls.  However, when I went to the time server pool at ntp.org (specifically us.pool.ntp.org) it seemed to work fine, so apparently their time servers “speak” SNTP.  us.pool.ntp.org is a good time server choice if you’re in the U.S.A, while in Canada you’d probably want ca.pool.ntp.org, and if you’re someplace else you can start here and select your continent, then drill down until you find the time server for your country.

Then select your time zone, check “Daylight” if your area observes Daylight Savings Time, and check “select sntp” (if you don’t check that box, it expects that you will be entering the time manually using the lower section!), and Apply. The default timeout of 60 seconds is fine; that’s just how long the device will wait for the time server to respond before giving up.  There’s no real reason to use the Manual Timeset section, unless perhaps for some reason you are on a system that has no connection to the Internet (I suppose such “closed” systems must exist somewhere).

You may be wondering if the Daylight time rules are configurable.  Not from the Web interface, sorry.  Newer Linksys devices, in contrast, let you specify a daylight time change rule (which is a string format that’s clear as mud for many users, but at least they make it possible to install a rule for your area from the Web interface). I had originally thought there was no way to change the rule in the AG-188N, but then I discovered that if you click on “Backup Config” in the left-hand menu, you can save your configuration settings in a plain text file, and in that file there is a section in there that looks like this:

DayLight Shift Min :60
DayLight Start Mon :3
DayLight Start Week:5
DayLight Start Wday:0
DayLight Start Hour:2
DayLight Start Min :0
DayLight End Mon   :10
DayLight End Week  :5
DayLight End Wday  :0
DayLight End Hour  :2
DayLight End Min   :0

If you were to edit those values (using a text editor that does not change the line endings, please!) and then use “Web Update” to reload the configuration, you could change the Daylight time rules. While this is a bit of a pain, it’s probably actually easier to understand than the way Linksys does it.  If you need to know when Daylight time starts and ends in your part of the world you can consult Wikipedia, but in most of the United States and Canada, DST starts on the second Sunday of March and ends on the first Sunday of November. Wikipedia says that they don’t observe DST in China, so I have no idea whose DST rule is loaded into the AG-188N, but it sure doesn’t look like the one for the U.S.A. and Canada to me.  I’m guessing that the following values should be changed as follows (assuming that they are using “1″ based numbering on week and month):

DayLight Start Week:2
DayLight End Mon   :11
DayLight End Week  :1

If I’m wrong, I’ll bet I’m still closer than the original rule! Anyway, if you do feel brave enough to edit the configuration backup file, after you carefully make the edits, rename the backup file extension from .txt to .cfg before using “Web Update” to load it back in. I actually did it and it took the changes (confirmed by saving another backup), but I’m not suggesting anyone should do it – if you want to try it, you assume the risk that something may go wrong, not me!

Okay, with that out of the way, let’s move on to making outgoing calls. Right out of the box it’s possible to dial some numbers, but if you have ever configured a VoIP adapter in the past, you know there are tweaks that need to be made to accommodate local dialing patterns. For one thing, it you were to use the AG-188N to try dialing a feature code that starts with a “star” character (*), such as *43 for the echo test on a FreePBX box, you’d note that the moment you hit the * key it goes to a busy signal. One reason for that is that by default, the star code indicates that you want to access the PSTN line connected to the PSTN port on the device, instead of the IAX and/or SIP account(s) configured on the AG-188N. This allows one analog phone to use either IAX/SIP or the PSTN. But if you aren’t using the PSTN port, and you are using an account on an Asterisk server, you probably want to be able to dial “star” feature codes without the AG-188N intercepting them. In order to accomplish that, we must select “Dial-Peer” in the left-hand menu – yes, I know it looks like it’s not a link, but it is — trust me. Click on it, and you will see this:

Atcom AG-188N Dial-Peer configuration screen

By default, there is only one entry here, for *T — you should delete this rule entirely.  If you still want to be able to access a phone line plugged into the PSTN port on the device, you could create a new rule, such as **T (which would require two depressions of the * key to select the PSTN line) and use the same parameters as the original *T (in case you are wondering, no you can’t simply modify the existing rule, because when you click on “Modify” you can modify every field except the Number). By the way, the T with no digit after it is the same as T0 (T followed by a zero), which means that there is no delay once the pattern is matched.

You can think of the Dial-Peer page as a variation on speed dialing, and a place to add digits to, or strip digits from a dialed number.  According to the AG-188N manual, here’s what can be accomplished on this page:

• Replace, delete or add a prefix to the dialed number
• Make a direct IP to IP call
• Place the call to a different SIP server after dialing a prefix
• Make PSTN calls using the Lifeline function (the rule I just had you delete or modify above)

If you need to do any of those things, make sure you read the section of the manual dealing with the Dial-Peer configuration page. The specific case of adding a prefix (1 + area code) to a seven digit local number is covered in Part 7 of this series, along with some additional information about dialed number modification using the Dial-Peer page.

Now at this point you might pick up the phone and hit the star key, only to discover that you’re still getting a busy signal.  What’s going on? Well, it turns out that there’s one more place where an adjustment needs to be made. In the left-hand menu, select “Digital Map” – that should bring up this page:

Atcom AG-188N Digital Map configuration screen

“Digital Map” is a fancy way of saying “Dial Plan” – it’s the dial rules used to determine when a number that has been dialed is complete, and should be sent to the switch. Yes, they used the word “Prefix” in this section – I attribute it to someone picking the wrong English word when translating from Chinese.  Either “Pattern” or “Rule” would have probably been a more appropriate choice than “Prefix.”

You’ll note that there’s a rule for a single star (in Texas they’d call it a lone star). But, we don’t want the AG-188N to think that dialing is complete after we’ve only pressed the star key once, so using the dropdown and “Delete” key on the bottom line of the page, I deleted that rule. At that point, I had no dial rules at all.  I could dial any number, but there would be a five second delay after the last digit dialed before the call went through. Obviously, we don’t want a post-dial delay on calls if that can be avoided, so we should add some rules, or prefixes as they are called here. But first, we’d better know what we can place in the rules:

• The digits 0-9 and the * character
• A range of digits in square brackets [ ], for example it could be a range such as [1-4], or use comma to separate individual numbers, such as [1,3,5], or use a list such as [234].
• x  is a “wildcard” character represents any one digit between 0~9
• Tn if used, must be the last thing on a line, and represents the amount of time we will wait after a pattern is matched to see if the user will depress any additional digits. n can be any number of seconds in the range 0-9. If the n digit is omitted and T used by itself, it means the same as T0, which means there would be no wait at all after the last digit is dialed. However, it appears that either T0 or a T by itself may be redundant in this screen – when a pattern is fully matched it is assumed to be complete, and there would be no further delay unless Tn is used, when n is some non-zero value.

Now, for those used to constructing Dial Plans on a Linksys or Sipura device, the principle here is pretty much the same.  Just keep in mind that instead of using a bar character | to separate individual dial plan rules, here you add each separately, and it occupies its own line.  Also, instead of using the S (seconds) to indicate a wait delay, here we use T (time in seconds).

The manual gives some usage examples:

• [1-8]xxx  All number from 1000 to 89999 will be sent immediately
• 9xxxxxxx 8 digits numbers beginning with 9 will be sent immediately
• 911   The number 911 will be sent immediately
• 99xT4  3 digits numbers that begin with 99 will be sent after four seconds

So, let’s take a Dial Plan string that we might find on a Linksys or Sipura device in the USA or Canada, where local extensions are assigned numbers in the range 1000 through 1999 and it is desired to have seven digit dialing of calls with one’s own area code, and eleven digit dialing to all other area codes:

([3-9]11|1[01]xx|[2-9]xxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|*xxS0|[*x].S4)

Here’s how each rule would be translated to a “Prefix” on the AG-188N:

• [3-9]11 — [3-9]11 or [3-9]11T or [3-9]11T0
• 1[01]xx — 1[01]xx or 1[01]xxT or 1[01]xxT0
• [2-9]xxxxxxS0 — [2-9]xxxxxx or [2-9]xxxxxxT or [2-9]xxxxxxT0
• 1[2-9]xx[2-9]xxxxxxS0 —1[2-9]xx[2-9]xxxxxx or 1[2-9]xx[2-9]xxxxxxT or 1[2-9]xx[2-9]xxxxxxT0
• *xxS0 — *xx or *xxT or *xxT0
• [*x].S4 — [*0-9].T4

As you can see, the most common patterns translate very easily.  But some things are questionable.  Note that last example – whereas a Linksys device allows [*x] to represent “any digit or the star key”, the Ag-188N seems to require the slightly more specific [*0-9].  But also, that rule contains a period (.) period character, which on a Linksys/Sipura device means “one or more of the preceeding character.”  Here it means any length pattern starting with * or 0-9, with any number of digits (possibly mixed with star key presses) following.  This type of rule (or something very similar) is commonly used as a “default” rule, to set a four second timeout for anything that doesn’t fit one of our other rules, such as international numbers. The problem here is that the AG-188N documentation makes absolutely no mention of the period character as having any significance in these settings, yet when I use it as I would on a Linksys/Sipura device, it appears to work in exactly the same manner.  Something they forgot to document, perhaps?

Of course, you could do away with that last pattern entirely and simply set a default timeout, as shown in the image above, but keep in mind, that controls your maximum interdigit delay during all normal dialing (though NOT the length of the initial dial tone).  If you pause to look at a number while dialing, you probably don’t want the device to time out on you too soon.  You might prefer to use a long default “Time Out” (something like 25 seconds), but then specify a shorter timeout for international calls, so instead of that final catch-all rule, you might use something a bit more specific, such as 011xxxxx.T4 (which would mean that once you’ve dialed 011 plus five additional digits, your interdigit timeout is reduced to four seconds.

Or, you could simply use a long “Time Out”, but then always hit the pound key # at the end of dialing any number that doesn’t match one of your patterns. That would work as long as the box for ‘End with “#”‘ is checked – when you check that box, it means that the # key signifies the end of the dialed number.

So, here’s an example of what this page might look like after you have converted the Linksys/Sipura Dial Plan string:

Atcom AG-188N Digital Map sample configuration

I know that there are a few things that can be done in a Linksys/Sipura Dial Plan that can’t be done on this page, but often those things can be done, just in a different way.  For example, if you need to do number translations (where you dial one number but the adapter sends something else), you can do that on the aforementioned Dial-Peer configuration page.  And if you want to set up “hotline” service, where the adapter connects to a particular number as soon as someone picks up the phone, that’s also possible, but you’d have to do it from the Call Service configuration page, which is worthy of note because of the capabilities it offers:

Atcom AG-188N Call Service configuration screen

On this page you can set up the following:

• Hotline: If you enter a number here, the  AG-188N will immediately dial that number after the phone is taken off-hook.
• Call Forward: You can call forward on busy, no answer, or always.  In the case of “no answer”, be sure to set the timeout in the “No Answer Time(seconds)” field.
• No Disturb: Usually known as “Do Not Disturb”, when enabled this option will refuse all incoming calls.
• Ban Outgoing: Enable this to ban outgoing calls (useful if, for some reason, you want a phone to receive incoming calls only).
• Enable Call Transfer: “If A is the AG-188N user, and B calls and talking with A through VoIP. A can press the Hook-Flash to hold the call with B, and then press * and then enter C’s number. B will be transferred to C and can talk with C.”  Note that Asterisk and FreePBX users might prefer to use Asterisk’s call transfer facilities.
• Enable Three Way Call: According to the manual, only the SIP protocol supports this function — but don’t necessarily believe what the manual says (see below). “Assume A is the AG-188N user, and B calls and talking with A through VoIP. A can press Hook-Flash to hold the call with B, then enter C’s number to talk with C, and then press Hook-Flash again switch back to user B,  then A can press * to make 3-way conference calls.”
• Enable Call Waiting: Enable/disable Call Waiting
• Accept Any Call: If this option is disabled, the AG-188N will refuse the incoming call when the called number is different from AG-188N’s phone number.
• No Answer Time: no answer call forward time setting.
• P2P IP Prefix: Configure point to point calling. For example, if you want to call IP 192.168.1.119 Just define 192.168.1. here and you can dial #119 to make the point to point call with the IP phone of 192.168.1.119
• Use Record Server: Configure the support of server recording, transferring all the recordings to the server.
• Remote Record No: Configure the recording number – if you upload your recording to this account, you can login to this account to get your voice recording. I assume the idea here is that you have some internal number you can call that will simply begin recording everything it hears – this is certainly possible to set up on an Asterisk server.
• Black List: incoming calls with these phone numbers will be refused.
• Limit List: outgoing calls with these phone numbers will be refused. Although the documentation doesn’t say so, I tried using a pattern here (29x to ban calls to all extension 290-299) and it did work. I assume that’s true of the Black List also, which makes both of these settings much more useful.

Note than many of the above features can also be accomplished via FreePBX or Asterisk, using feature codes, but many folks might appreciate the ability to set certain features from a web page rather than having to remember and dial a particular feature code.  The only thing I changed on this page was that I enabled Call Transfer, Call Waiting, and Three Way Call (I left Accept Any Call checked as well).

The one thing that initially seemed to be a bit of a disappointment here was the inability to do three-way calling using IAX, according to the manual.  I had actually tried to use the feature anyway, failed, and had written a couple of paragraphs lamenting this limitation and suggesting a really crude workaround in Asterisk.  But then I took another look at the above description again, and realized that I had done it wrong. The way it works is, you have to call both parties (call the first, flash, and call the second – or – receive incoming call, flash and call second party) and then, when you are in a condition where you can flash back and forth between the two parties, press the * key — that last minor but crucial step is what makes all the difference.  I have no idea why the manual would say that only SIP supports this function, but I’m very happy to report that it’s just plain wrong on that point, at least according to my testing.

Call transfer is another function that doesn’t work exactly as it might on some other adapters — for example,  on a Linksys/Sipura, you simply flash and call the number you want to transfer the call to, then hang up (either while it is ringing, or after the recipient of the transferred call answers and you’ve had a chance to talk to them before handing over the call). With the AG-188N it’s strictly “blind” transfer — once you flash and then dial * plus the extension number (don’t forget the leading *), the call is immediately transferred.  Of course, FreePBX and Asterisk support both “blind” and “attended” transfers internally, so this probably isn’t a big deal if you’re using an Asterisk server (which you would be if you’re using IAX protocol). And I would guess that 95%+ of all transfers are “blind” transfers anyway, judging by calls where I’ve been the one transferred.

Do remember, if you have made any changes after reading this article, to save your configuration, so you don’t lose all your changes when you reboot.

In the next installment, I want to cover setting up a SIP account.  I’m expecting it will be pretty straightforward, but I’ll let you know once I actually do it.  Overall, I’m still very impressed with this unit.

And now a small bonus, for those that might want to know such things.  First of all, here’s a list of things you can do from the telephone handset, just by dialing certain codes:

• #****# — reboot gateway
• #*000# — clear settings (don’t do this unless you really need to!)
• #*100# — set the IP type to static ip
• #*101# — set IP type to DHCP
• #*102# — set IP type to PPPoE
• #*111# — say unit’s ip address (on WAN port)
• #*222# — say phone number

The settings below need a reboot to take effect:

• #*103# — change to bridge mode
• #*104# — change to router mode
• #*50192.168.1.117# — set WAN port IP address
• #*51192.168.1.1# — set default gateway IP
• #*52202.112.10.37# — set dns server
• #*53255.255.255.0# — set netmask, use 255.255.255.0 if none given

In all of the above, the trailing # is optional, but speeds things up. In the last four examples, I would ASSUME that the star key * substitutes for the dot in the dotted IP addresses.

Also, this unit does have telnet access, or as the manual says, “User may use telnet command to access and manage gateway.” However, as I prefer to work within the GUI, I’ll stick with it. But I mention this because I know there’s a certain class of folks that love to get into the internals of a piece of equipment, and for those folks telnet access might be quite useful.

Disclosure: CIGear provided me with an Atcom AG-188N for review purposes, and allowed me to keep it after I was finished writing this series, and for that I am most grateful.

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Articles in the series: Review of Atcom AG-188N IAX+SIP ATA (VoIP adapter)

Part 1 – The unboxing
Part 2 – Initial setup using IAX
Part 3 – Setting the time and configuring outbound dialing
Part 4 – Setting up SIP, and securing the adapter
Part 5 – Networking and Internal Router
Part 6 – Final Thoughts and Summary Review
Part 7 – Addendum

In yesterday’s installment I explained why, in some cases, IAX is the superior protocol to use for VoIP calls.  Of course, having a superior protocol doesn’t count for much if the calls don’t sound good.  So, my first goal with the AG-188N (sold in North America by CIGear) was to try and get it up and running to the point that I could test the actual audio quality.

First, a clarification from yesterday’s post – I’ve been told that the unit I received had been previously used on a demo and therefore had been repackaged. Normally the power supply comes in a small white box, not wrapped in bubble wrap. Since I had specifically asked for a unit for review purposes, it doesn’t bother me at all that I received a unit that is new for all intents and purposes, but can’t be sold as new – that’s the perfect unit to send to a reviewer.

After unboxing the unit, I hooked it up to my local network, plugged a phone into the phone port, and connected the power supply and plugged it in. I picked up the phone and heard a distinctly foreign sounding dial tone (I later figured out it was a Chinese dial tone, probably since the unit is manufactured in China). I knew there had to be a web interface, but had no idea what IP address had been assigned to the unit. This is the point at which I might normally have referred to the manual, but there was no printed manual supplied. The only manual was on a small “mini CD” that came with the unit. Unfortunately, my primary computer is a Mac Mini, which has a front-loading DVD drive (where you shove the CD or DVD into a slot, and the Mac Mini sort of sucks it in). I had no idea if that drive could handle a “mini” CD, or if attempting to insert it would jam up the Mac Mini’s drive. Since I didn’t want to chance it, I took the expedient solution of firing up my old Windows box just long enough to read the CD and copy it to a shared drive, where I could peruse it on my Mac.

Mac Mini + Mini Disk? Nah, not gonna try it!

Unfortunately that hit a snag as well – the files on the CD are in .doc (Microsoft Word) format, not the more widely accepted PDF format. I never use .doc format – if I am writing a letter or something I use .rtf (Rich Text Format) which is far more universal. Nevertheless, to its credit, QuickLook on the Mac could display the .doc files (I might have installed a plugin to allow that at some point in the past), albeit in very tiny text. I later discovered that I could download the documentation in PDF format at the manufacturer’s web site, although it actually appears that the documentation on the disk is more complete in some sections.

The documentation revealed that in order to discover which IP address the device was occupying, I needed to pick up the phone and dial #*111#

After doing that, a very North American-sounding male voice read the IP address to me.  I entered that address into my web browser, and was greeted by a rather plain but functional Logon page:

Atcom AG-188N Logon screen

Back to the .doc file, which revealed that the default logons are “admin/admin for administrator and guest/guest for user.” So I logged in as an administrator, and got this screen:

Atcom AG-188N first screen after logon

Note that there is a left-hand sidebar and a main window. In the upcoming screenshots I’ll only be showing the main window, so let’s get a closer look at that sidebar now:

Atcom AG-188N sidebar

As you can see, this unit has many pages of options, but fortunately we don’t need to bother with most of them just to get the device working well enough to make a test call.

The first thing you will probably want to check is the WAN config page. By default, it looked like this (I have deliberately blurred portions of a few items that probably shouldn’t be shown):

Atcom AG-188N WAN configuration screen

For the majority of users, the main thing to check here is that it is using the DHCP server on your network to get its address – with the “DHCP” button selected and “Obtain DNS server automatically” checked, you can probably ignore the other text boxes. One thing the documentation is strangely silent on is the use of the “Mac Authenticating Code” field – I would ASSUME that’s so you can “clone” a Mac address, if you happen to be so unlucky as to be served by one of those providers that still actually cares what your router’s MAC address is, and you plan to plug the AG-188N directly into your cable or DSL modem rather than a router.

After that I decided to skip ahead a bit and get the IAX setup working. I clicked on “IAX2 Config” and the following screen came up:

Atcom AG-188N IAX2 configuration screen

This is super easy to set up. You only need edit the following fields for use with Asterisk or FreePBX:

• IAX Server Addr – this is the address of your server, such as 192.168.0.100 or myserver.dyndns.com
• IAX Server port – leave at 4569
• Account Name – just use your extension number here
• Account Password – the same as the Asterisk “secret” for your extension
• Phone Number – your extension number (again)
• Local Port – leave at 4569
• Voice mail, Echo Test fields – just blank out all four (you dial codes like *43 or *98 to reach these in FreePBX)
• Refresh Time – all the documentation says about this is that it is the “IAX refresh time” – I assume it’s how often the device re-registers with the server. I had originally thought 60 might be a bit too short of an interval, but when I tried a higher value I discovered that calls to the unit failed. So, my advice is, leave it at 60!
• Enable Register – Always check this box, to enable IAX registration, if you plan to use IAX.
• Enable G.729 – G.729 is a codec that saves bandwidth at the expense of audio fidelity.  Unless you are sure you need it and the server supports it, leave it unchecked.
• IAX(Default protocol) – I checked this box, because I wanted to test IAX and don’t have SIP configured yet in any case.  By default, the AG-188N assumes you want SIP to be the default protocol; this changes that assumption

After you make the above changes, be sure to click the “Apply” button at the bottom.  That saves your changes, as long as the unit is not rebooted or powered down. So now I must bring up this important point:

IMPORTANT: After you make changes on any page, you must press the Apply button at the bottom of the page.  But then, after you have made all the changes you plan to make, in order to save them pernamently you MUST click on “Save Config” in the left-hand menu, after which you will see:

IMPORTANT! You MUST click this SAVE button to save your changes!

CLICK THE SAVE BUTTON!!! If you forget this step, you get to enter all your changes again after the next reboot of the unit!

At this point, I tried making a test call to the unit.  The phone rang (a short ring by North American standards) but it was perfectly usable, although to my ear the receiver volume was low (people on the other end could hear me just fine, but I had a bit of trouble hearing them).  But I quickly found the fix for those things.  Click on “Audio Settings” in the left-hand menu and you will see this screen (if you have the older firmware — the newer firmware has additional CODEC options, as explained below):

Atcom AG-188N Audio configuration screen

As you can see, unless you happen to live in China there are probably a few things you’ll want to change there! So I started by changing these:

• CODEC – This is the codec you “prefer” to use – in the U.S.A. and Canada you’ll almost certainly want to use g711Ulaw64k (note Ulaw, not Alaw) unless you have a severe limitation on available bandwidth. Your other choices are g711Alaw64k, g729, g726-32k, iLBC, and (if you have the newer firmware — see below) None. Also, in the newer firmware, there are five CODEC settings, allowing you to select up to five CODECs in order of preference.
• Signal Standard – This controls things like Dial Tone and Ring Length.  Your choices are CHINA, USA, JAPAN, UK, ISRAEL, BRASIL (their spelling), TOPCOM, and AUSTRALIA.  I’m not sure what country TOPCOM is, but if you live there, you’re supported!   (Apparently, Topcom is a European communications company that must have their own signal standard.)
• Input Volume – 0 seemed to work fine for me, but if your callers complain they can’t hear you, try bumping this a number or two.
• Output Volume – I found 3 or 4 to be a comfortable setting.  Note that your preference here may in part depend on the phone you are using, but if you are having trouble hearing the people you are talking to (or if they come through too loud), this is the setting to tweak.

I didn’t touch any of the other settings on this page.  Remember to click “Apply”, and then don’t forget to save your configuration! EDIT: After originally posting this article, I was informed that there are now additional CODEC settings on the page that are in the newest firmware, so I went and grabbed that firmware from Atcom’s web site, uploaded it, and this is what the page looks like after configuration and with the additional CODEC settings:

Atcom AG-188N Audio configuration screen - updated firmware

Now I could receive incoming calls, and I could place some outgoing calls.  I say “some” because I still needed to make some adjustments for dialing out (those will be covered next).

So, how did it work? Wonderfully! Calls are at least as clear as any other adapter I’ve used.  I have a friend who calls me frequently, and he just happened to call while I was setting this up, so when I got it configured I transferred his call to this adapter and he said that I sounded better than on the original connection (which was another phone that’s connected to a Linksys PAP2).  On my end, once I had turned up the volume a bit as shown above, he sounded great as well – certainly as good as any other VoIP or landline connection I’ve ever used, and perhaps even a bit better. My friend also said that he didn’t notice the little bit of echo “tail” that he occasionally hears from my end when we use the other phone and adapter – granted, that may be a configuration issue on the PAP2, but it’s worth noting that this unit seems to be performing better in that regard right out of the box. I would certainly have no reservations about recommending this unit to someone based on audio quality.

So you may be asking, have I found anything not to like so far?  Well, only one very minor thing, maybe.  The “wall wart” power supply that comes with the unit has the following specifications:

• Input:  100-240V~50-60 Hz 0.18A
• Output: 12 volts – 0.5 A
• Has FCC and UL certifications

It’s a rather small unit, typical of a “wall wart” except for one thing – when connected to the AG-188N and plugged in, it gets a little bit warmer than I’d expect – sometimes.  Yesterday, it seemed to be running a little warm, not “burn your hand” warm or anything like that, but definitely the warmest of the dozen or so “wall warts” in my immediate vicinity.  I then unplugged it overnight, and after reconnecting it this morning (and leaving it plugged into the wall but not connected to the adapter for an hour or so, during which time it ran cool as a cucumber), today it seems to be only about as warm as some of the others, maybe even not quite as warm as a couple of them.  So what changed – my sense of touch, or did the actual amount of heat produced decrease?  Is the AG-188N drawing less current today? Did the fact that I plugged it into a different outlet on my UPS make some difference? Was I just oversensitive to temperature last night? I dunno, and it may remain one of the mysteries of life.  Anyway, it’s not something I’m really concerned about.

Next up, we take on outbound dial plans, and maybe more… stay tuned!

Disclosure: CIGear provided me with an Atcom AG-188N for review purposes, and allowed me to keep it after I was finished writing this series, and for that I am most grateful.

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Articles in the series: Review of Atcom AG-188N IAX+SIP ATA (VoIP adapter)

Part 1 – The unboxing
Part 2 – Initial setup using IAX
Part 3 – Setting the time and configuring outbound dialing
Part 4 – Setting up SIP, and securing the adapter
Part 5 – Networking and Internal Router
Part 6 – Final Thoughts and Summary Review
Part 7 – Addendum

One of the problems often encountered with VoIP is the use of the SIP protocol, specifically when used between a VoIP adapter such as the venerable Linksys PAP2 (or something similar) and an Asterisk server. Everything works great until you try to deploy an adapter at a remote location and the user has some kind of funky NAT firewall that just doesn’t play well with SIP. If you have any doubts that this is a real issue, Google the phrase “one-way audio.”

The crux of the problem is that SIP uses a wide number of ports, and unless you get everything set up exactly so at both ends, you may find that although the VoiP adapter will register with the server, when you actually attempt to place a call it will go through but neither party will be able to hear the other, or more commonly, only one party will be able to hear the other.

It’s an especially frustrating problem for VoIP providers, because a certain number of customers — we don’t know how many, because it’s not the sort of statistic providers like to report — cancel their VoIP service because they just can’t get it working in the first place, and the one-way audio probably certainly must account for a not-insignificant percentage of those cancellations.

There is another protocol that Asterisk supports, called IAX (actually it’s now up to IAX2, but we’ll just call it IAX to keep things simple). It works quite well in “difficult” situations because it only uses ONE port for all information. Therefore, if you can see the device register on your Asterisk system, you know that you’re also going to get audio. Assuming you don’t have problems with excessive packet loss or something of that nature, an IAX connection will almost always work, even if the end user is behind some sort of funky router (or even a chain of routers). The main reason that IAX hasn’t been more widely used is that there has been a real lack of decent ATA’s that support IAX (also, I’ve heard some comments that it doesn’t “scale” well in very large installations, but unless you plan on competing with one of the large commercial VoIP providers, I wouldn’t worry about that). For a while, Digium (the company behind Asterisk) made a unit called the IAXy, but like much of the hardware that Digium sells it was priced just a tad on the high side, and everyone was looking for a less expensive solution, so SIP-based ATA’s have become the norm.

But that doesn’t mean that the need for a decent IAX-based adapter has gone away. Just about every Asterisk system administrator has run into one or two “tough cases” where SIP just wasn’t cutting it, and if they did manage to resolve the issue they probably acquired a few grey hairs in the process. If you are the administrator of an Asterisk system, perhaps you’d just like to be able to pre-configure an ATA before sending it to someone that’s doing some off-site work for you, and know that when they receive it they can plug it into whatever crappy old router they happen to have and it’s probably going to work — right out of the box, with no frustrating hours of “… try changing this setting …” as you burn up cell phone minutes.

Recently I became aware of the fact that there’s an adapter available that can communicate using both IAX and SIP, and it even has both a WAN and a LAN port so that if the end-user doesn’t even have a router, they can plug this unit into the cable or DSL modem, and then plug their computer into the LAN port on the adapter and both will work. I made arrangements to obtain a unit for review purposes, and in the coming days I plan to tell you more about this device. It’s called the Atcom AG-188N and it’s sold in North America by CIGear. Just to give you a small preview, here are a few “unboxing” pictures.  The small box below came inside a larger carton with all the requisite packing peanuts, but I’m not a “pro” at making these photos (a couple came out way too blurry to use) so I decided to just show you the interesting part:

Good things come in small packages

Out of the carton

Contents of the box, without the bubble wrap

Closeup of the AG-188N, sorry it's a bit blurry

The back side, showing the connections, from CIGear's web site.

EDIT: Since I first posted this review I’ve since found out that the particular unit I received had been previously used on a demo and therefore had been repackaged.  Normally the power supply comes in a small white box, not wrapped in bubble wrap. Since I had specifically asked for a unit for review purposes, it doesn’t bother me at all that I received a unit that is new for all intents and purposes, but can’t be sold as new – that’s the perfect unit to send to a reviewer.

Note that the yellow ruler doesn’t come with the unit, it’s just a 7″ ruler I had lying around that I put in the pictures to give you an idea of relative sizes. Actually, if you set this unit next to a Linksys PAP2 you’d be very hard-pressed to tell any difference in the size. One thing notably missing from the contents of the box: A network cable. If you were going to buy these in quantity to send out to users, you’d probably also want to obtain some some network cables in bulk, so you could throw a cable in with each shipment, because for every guy like me who’s somehow obtained a small surplus of them, there will always be the folks that only have ONE network cable to their name.

One thing you may notice in the last image above is that there is a phone port and a PSTN (telephone company line) port. Yes, that does mean that this is a single line unit.  But, don’t get the idea that because there is a PSTN port, that this is the full equivalent of something like a Sipura SPA-3000 or Linksys SPA-3102, because it isn’t — the PSTN port is a “passthru” port, allowing you to send some calls to the PSTN and others to your VoIP provider based on the pattern dialed, and letting you receive calls from both the PSTN and your VoIP provider without the need use two separate telephones. The difference between this and one of the aforementioned Linksys/Sipura boxes is that you can configure the Linksys/Sipura’s PSTN port to be a trunk on the Asterisk server (essentially an FXO port), and that particular functionality is not included with this unit.

EDIT:  The above paragraph was edited slightly after receiving some clarification from CIGear — they also offered this additional information: With one phone hooked to the ‘phone’ jack, you can dial a star code to access the PSTN line instead of the SIP/IAX configured on the ATA. So, one analog phone enables you to use either SIP/IAX or the PSTN. However, people using the ATA with Asterisk have a problem when they want to dial Asterisk * codes like *97 for voicemail (as the * takes you to PSTN). The solution is to delete the “lifeline *T” option from “Dial-Peer” menu (this gets a bit into configuration, which will be discused much more thoroughly in upcoming posts).

Of course, you don’t HAVE to connect a PSTN line to this unit, and you don’t HAVE to connect a computer to the LAN port — I have used it successfully tonight without doing either.  Ah, but I’m getting ahead of myself a bit – more on that in the next installment, after I’ve had some time to spend working with the unit and reading the documentation (yes, I did get it working mostly without referring to the documentation, so for those who hate wading through documentation, fear not — it’s really not difficult to set up)!

Disclosure: CIGear provided me with an Atcom AG-188N for review purposes, and allowed me to keep it after I was finished writing this series, and for that I am most grateful.

Next installment

Articles in the series: Review of Atcom AG-188N IAX+SIP ATA (VoIP adapter)

Part 1 – The unboxing
Part 2 – Initial setup using IAX
Part 3 – Setting the time and configuring outbound dialing
Part 4 – Setting up SIP, and securing the adapter
Part 5 – Networking and Internal Router
Part 6 – Final Thoughts and Summary Review
Part 7 – Addendum