Note: This review was originally posted on the Michigan Telephone, VoIP and Broadband 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.
Previous Installment | 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
