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Jan 30 2011

Review of the Obihai OBi110 VoIP device, Part 4: How to use the OBi110 as an FXO port with Asterisk and FreePBX

Important!

This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging.  It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog. In order to comply with Federal Trade Commission regulations, I am disclosing that he received a free product sample of the item under review prior to writing the review, and that any links to Amazon.com in this article are affiliate links, and if you make a purchase through one of those links I will receive a small commission on the sale.

This article was originally written in January, 2011. Some edits have been made in an attempt to bring it up to date, however it was a first attempt at doing this and does not necessarily show the current best method. It is reposted here because it was part of the original review series.

This article sort of assumes that you have read the earlier parts of this series, including Review of the Obihai OBi110 VoIP device, Part 1: Use your phone with Google Voice for free incoming and outgoing calls, Review of the Obihai OBi110 VoIP device, Part 2: The OBiTALK portal, documentation, and using the device as an FXO port with Asterisk, and Review of the Obihai OBi110 VoIP device, Part 3: 911 on the cheap?

I’m sure some of you have been waiting for this, but I just want to say up front that whenever you use any kind of PSTN interface with Asterisk, be it an Obihai OBi110, a Sipura SPA-3000, a Linksys/Cisco SPA-3102, a USB device, or a card that is inserted into your server computer, you’re probably going to face a few challenges.  Some of these devices offer more challenges than others!  Getting the OBi110 Line port to work as an Asterisk trunk definitely wasn’t the hardest thing I’ve ever done, but it wasn’t something I figured out in 15 minutes, either.  And just because I finally got it working doesn’t necessarily mean that if you follow my “recipe”, yours will work.   As I pointed out in Part 2, I gave up my landline many moons ago, so the only way I have to test this is by connecting it to the FXS (phone) port of another VoIP adapter, which is acting as a stand-in for a telephone company central office.  However, that connection is via a six-foot long phone cord.  Things may be different if you are connected to a real PSTN line over miles of old copper (or worse).  I’m only saying this in case you try to emulate what I’ve done and it doesn’t work as you expect — there’s a possibility that could happen and if it does, you have my sympathy, but I probably won’t be able to offer any concrete assistance.

That said, I figure any instructions are better than none, so here’s what I’ve done.  Some of the information here was obtained from this thread on the OBiTALK forum.

As always, this article is intended for experimental purposes only — if it works for you, great, but I make no guarantees. Don’t use this in a production environment until you have satisfied yourself that it works the way you need it to. If you use my instructions and have lost income due to missed calls, or if someone manages to hack into your device and runs up your phone bill, you agree that by using these instructions all the liability for that rests on you and nobody else! If you’re the type of person who wants to sue someone else when things go wrong, then you absolutely may not use these instructions. We’re all experimenters here, and nobody’s offering insurance that nothing will go wrong.

All of the images below can be viewed in a larger format by clicking on them.  If a page is shown here, any settings on it that are not specifically mentioned or shown should be set to their default values (except on the first screen shown below — on that page just leave any settings I don’t specifically mention alone).  These instructions sort of assume you’ll be connecting to a PSTN line with a ten-digit phone number, which for convenience is used in several places.  If that’s not the case for you, you’ll have to make a mental substitution.

This article was originally written before Obihai introduced the OBi Expert Configuration feature in the OBiTALK portal.  You can (and should) make the following changes there if you normally use the OBiTALK portal to configure your device.  Note, however, that you will still need to enter the device’s local web portal long enough to change the IP address to a static one — that is one of the few things you cannot do from the OBi Expert Configuration section of the OBiTALK portal.

If you have not done so already, you’ll want to make sure the device has a static IP address on your network.  I show some typical values in the example below, but be sure to use values that are sane for your network.  This must be done from the OBi110’s internal web interface. You can get the address by picking up the phone connected to the phone PORT and dialing * * * 1. When you go to the device in your web browser, the user name will be “admin” and the password will be whatever you chose when you set up the device using the OBiTALK portal (you can yiew it there if you have forgotten what it is). Navigate to System Management, Networks settings to set up your static IP address, and after making the changes be sure to click Submit at the bottom of the page (and don’t forget to click Submit on any page where you make changes).

Where to set a static IP address and other network settings

The rest of this configuration can and should be done using the OBi Expert Configuration feature in the OBiTALK portal, even though the screenshots below show manual configuration – again, that’s only because the OBi Expert Configuration feature had not yet been rolled out when this article was originally written.

I’m going to assume that you want to keep using your phone port with Google Voice and that it’s Service Provider A, so we’ll use Service Provider B to set up our Asterisk trunk information. So go to Service Providers, ITSP Profile B, SIP.  In the Proxy Server field, put the address of your Asterisk server.  Then scroll down (quite a way down) and find the X_SpoofCallerID field and make sure that is checked (otherwise you won’t get Caller ID passed through on incoming calls). Also, for security reasons, you may want to put the address of your Asterisk server in the X_AccessList field — I will draw your attention to this (slightly edited) note in the original thread on the OBiTALK forum:

NOTE: [The OBi110] does not challenge inbound INVITE [from the Asterisk server]. However you can setup a list of trusted IP addresses in the X_AccessList parameter (under ITSP Profile – SIP) to limit who can send SIP messages to the OBi [Service Provider]. Usually the gateway (OBi) and Asterisk machines are in the same subnet; normally not a big issue.

But it could be an issue if the OBi110 is not behind a firewall, and therefore could be accessed from the wide open Internet — in that case, anyone could make calls on your PSTN line unless you use the X_AccessList parameter to limit access. So to be safe, I strongly suggest putting the IP address of your Asterisk server in the X_AccessList parameter.

OBi110 ITSP Profile B SIP settings

Next, go to Voice Services, SP2 Service.  There are several settings on this page.  Make sure that Enable, X_RegisterEnable, and X_KeepAliveEnable are all checked.  Set X_RingProfile to A (the default) but set X_ServProvProfile and X_CodecProfile to B.  X_InboundCallRoute must be set to LI (very important!).  Then further below, in the SIP Credentials section, put the 10 digit number of your PSTN line in the AuthUserName, a good strong password in the AuthPassword field, and a URI string composed of the 10 digit number of your PSTN line, the @ symbol, and the address of your server (but preferably not in dotted IP format — even if you use the dotted IP address elsewhere, here you should use the address that is used to reach your server from the Internet.  Believe it or not, getting the URI right can make all the difference between it working and not working).

OBi110 Service Provider 2 Service settings

Finally, go to Physical Interface, Line Port and make two changes.  For the InboundCallRoute put SP2 and then in parenthesis the 10 digit number of your PSTN line.  The RingDelay value is how long it will wait to try to obtain incoming Caller ID data – the default is 5500 (5.5 seconds) and as you may know, I hate to make callers wait unnecessarily, so I pared off a couple of seconds and set it to 3500 – that works for me, but you may need to experiment to find what works for you.

OBi110 Line Port settings

While you are in the Line Port settings, scroll down to the PSTN Disconnect Detection settings.  This section control what conditions on your PSTN line will cause the OBi110 to think that the call has ended and hang up the call.  Most lines will present at least one of the triggers shown – a CPC signal (drop in voltage), a polarity reversal on the line, or a disconnect tone (what people often call a “fast busy” signal, though you can change the tone that’s used for detection in the “Disconnect Tone Pattern” setting).  But as a fallback, in case none of those triggers are present, the OBi110 will consider 60 seconds of silence from the far end as a disconnect.  The problem with that is that if you call into a business and they put you on hold, and they don’t provide music or “comfort tones” or some other indication that you’re still on hold, the Obi110 will disconnect you after 60 seconds.  So unless your PSTN line provides none of the other indications, I would suggest either unchecking the “DetectFarEndLongSilence” checkboxes, or setting the SilenceTimeThreshold to a much higher value than 60.  In this example I have set SilenceTimeThreshold to 1800 (30 minutes) – I figure that if we’ve received nothing but silence from the distant end for that long, it’s time to give up!  Please keep in mind that if the non-PSTN side of the call disconnects it will also end the call and cause the OBi110 to disconnect the PSTN line, so it would be a very rare situation where this timeout would ever be invoked, if you set it to a reasonably long value.

PSTN Disconnect Detection section of OBi110 Line Port settings

After you’ve saved all those, you may want to hold off on rebooting the adapter until you have set up your FreePBX trunk.  And here’s a slightly modified screenshot of the FreePBX trunk page (the USER details field was enlarged so you could see everything):

FreePBX trunk setting for use with OBi110

Note this is from FreePBX 2.7 – later versions will look a little different but the settings should be the same.  They are:

Trunk Description: Whatever you like — I used OBi110.

Maximum Channels: I now recommend that you leave this blank. The Obihai device notifies Asterisk when it cannot complete a call.

Dial Rules: 1+NXXNXXXXXX (in FreePBX 2.8 or later, put “1” in the first field and “NXXNXXXXXX” in the third field, without the quotation marks, of course).

Trunk Name: OBi110

PEER Details:

type=peer
host=192.168.0.222 (use the Static IP address of your OBi110)
port=5061
disallow=all
allow=ulaw
dtmfmode=rfc2833

USER Context: the 10 digit number of your PSTN line

USER Details:

type=friend
secret=the AuthPassword from your OBi110
host=dynamic  (really! This is important!)
context=from-trunk-sip-OBi110
canreinvite=no
nat=yes  (yes even if device and server are on same LAN!*)
port=5061
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.255.0  (use sane values for your network)

*Never use nat=never — it won’t work. nat=no might work if the device and server are on the same LAN, but nat=yes will most likely work in any situation, and there doesn’t appear to be any downside to setting it that way.

After you’ve saved the above settings and have done an “orange bar reload”, then you can reboot your OBi110.  Hopefully it will register with Asterisk and you’ll be all set.

Now, am I saying the above settings are perfect?  Absolutely not! I tried a whole bunch of different things and when it finally started working I pretty much stopped tinkering, so there may be an unnecessary setting or two. In particular, note that there is a way to do the trunk configuration without using the USER Details section at all, as explained in the comments on this article. This article was a first attempt to get this working.

If you run into an issue where, when the device’s Line port is already in use on one call and another call tries to use it, the subsequent call doesn’t properly “fall through” to another unused trunk, you could try setting Maximum Channels to 1. But if you do that, you should also set the context line to read context=from-trunk-sip-OBi110, so that both incoming and outgoing calls count toward the Maximum Channels count.  That’s so that when the PSTN line is in use, another available trunk will be used if anyone else attempts to make an outgoing call (assuming that another trunk is available). This should not be an issue with recent versions of the OBi110 firmware and therefore you should not make this change unless you actually experience this issue, because doing so could cause some unintended and unwanted side effects.

Example of FreePBX Maximum Channels tooltip

The format from-trunk-sip-trunk_name is a FreePBX thing. If you mouse over Maximum Channels, the tool tip that pops up will show you the context you would need to use to have both inbound and outbound calls counted against the maximum channels count.

In very limited testing it seems to work great — the sound quality seems excellent — but as I say, it might be a bit different when you have it connected to some old decrepit copper pair from the local telephone company that runs for several miles.

If you use Asterisk but NOT FreePBX, you might also want take a look at this article: Configuring the OBi110 to replace a linksys spa-3102 as a google voice and pstn gateway using asterisk 1.4

The Obihai OBi110 can be purchased at Amazon.com and at some other online retailers.

For another variation on the subject of this article, see How to use the Obihai OBi100 or OBi110 VoIP device as a gateway between Asterisk/FreePBX and Google Voice and/or the OBiTALK network, or preferably the newer revision, How to use the Obihai OBi100, OBi110, OBi200, or OBi202 VoIP device as a gateway between Asterisk/FreePBX and Google Voice and/or the OBiTALK network (UPDATED).

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  1. Bill

    OK, here’s the million-dollar question for me. Can it act as a Google Voice VoIP gateway, without connecting it up to another ATA to do an analog bridge? In other words, it connects to GV, and it connects to Asterisk, so that I can remove the GV configuration from Asterisk and have the Obihai do all the talking to and from GV?

    I see a lot of Asterisk people going wild for this device but other than the GV connectivity I am not seeing the benefits over any other low-cost ATA. What am I missing?

    If it can do what I asked in the first paragraph, that’s gold. But I haven’t seen that described anywhere.

    1. Admin

      (EDIT: The answer to Bill’s question is “yes”, and the instructions are in How to use the Obihai OBi100 or OBi110 VoIP device as a gateway between Asterisk/FreePBX and Google Voice and/or the OBiTALK network. But I hadn’t yet written that article when I wrote this reply and the next one.)

      Bill, if you’re going to run Asterisk anyway then I personally think you’re better off using Asterisk’s built-in Google Voice support, assuming you are running Asterisk 1.8.2 or later. If you are running an earlier version of Asterisk and don’t want to upgrade, then in that case an OBi110 might well give superior results.

      Now, your question is whether you can send a call to the OBi110 and route it to Google Voice rather than the Line port. I am 99.9% sure that you can (and if anyone could figure out how, I’m sure it would be you!). My first guess is that if you had Google Voice enabled as Service Provider 1, and you used the instructions in this article to use Service Provider 2 as a trunk to Asterisk, then you could simply replace the references to “LI” with “SP1″ (or just “SP”, since 1 is the default if you don’t specify), and of course you wouldn’t need to mess with the Line port settings. You might have to tweak a couple other configuration settings but after seeing some of the things you’ve done recently, I have no doubt whatsoever that you could make it work. In fact, you might even be able to do it with the OBi100 when it comes out, since you wouldn’t actually need the Line port.

      This unit has a LOT of flexibility in how you can route things – I don’t even pretend to understand it all. But let me quote a couple portions of an e-mail I received from Sherman Scholten at Obihai:

      … with the OBi110 (and OBi100), you can do an in-and-out call on the same SP (SP1, SP2). This is really nice for ‘hair pin” calls given we have the trusted PSTN numbers that can route directly to AA in the InboundCallRoute. This is not so useful with GV for two reasons. The first being they only support one call. The second being they can do this “in the cloud” already. However it is very useful for customers of ITSPs like Callcentric, Voip.ms, VoicePulse, etc.

      … [Using the OBiAPP] we can now do a single-stage call via an OBi device acting as a gateway. The ICR of the OBi (OBiTALK Voice Service) will have a rule that can route a call from a trusted OBiTALK endpoint directly to another service (SP1, SP2, Li1). The OBiAPP in the http://www.obihai.com web site has this now, and we will release the iPhone and Android version soon (with announcement).

      {290654321>(xx.):sp2} is in the OBiTALK voice service InboundCallRoute

      So, if you use the new app release, you can dial the OBi110 (200123456) from this OBi soft client 290654321 and one stage dial a PSTN number to SP2 (GV).
      From the OBi soft phone you dial one of the two following ways.

      20012345614155551212
      Or (if the gateway OBi110 has a speed dial of 8 ) 8141555551212

      Unfortunately I can’t use the OBiAPP because the current version is Windows only, and I don’t have an iPhone or Android phone so I won’t be able to run those versions either. I’ve been gently nudging them to at least make the current OBiAPP version compatible with WINE and/or Codeweavers Crossover so that we Mac/Linux users could run it (right now the installer throws up an error message under Crosssover). But I mention all that just to note that I don’t think I’ve really even scratched the surface of the hidden capabilities inside this unit. If someone really understood it (the way the designers undoubtedly do, but probably few others), I suspect you could do a lot of useful tricks with it. Once people smarter than I get their hands on these, I think you’re going to see a lot of useful applications.

      Remember that when Asterisk first came out, most people didn’t “get it” at first, and had a “so what’s it good for?” type reaction. I suspect there’s going to be a similar curve with this unit – once users start to build a knowledge base, people will have a much better understanding of what this unit is capable of.

    2. Admin

      Bill, in giving it a bit more thought I realized it would be incredibly easy to do what you want.

      Assuming that you have set up a Google Voice account on Service Provider 1 as described in Part 1, and that you have set up an Asterisk trunk on Service Provider 2 as described in this article, you need only make TWO changes:

      Go to Voice Services, SP1 Service, and change the value of X_InboundCallRoute from ph to SP2
      Go to Voice Services, SP2 Service, and change the value of X_InboundCallRoute from LI to SP1

      That’s it! When you make an outbound call from Asterisk through your OBi110 (or probably the OBi100, when it comes out) trunk it will loop around and go out to your Google Voice account. Of course the Caller ID will be that of the Google Voice account.

      When a call comes in from Google Voice, it will loop around and go to your Asterisk trunk, and from there to wherever you send incoming calls in your Inbound route. I actually tested this setup briefly and it worked great!

      Note that either of these could operate independently of the other. And also note that X_InboundCallRoute can hold a lot more complex routing instructions than just a simple reroute. It would be entirely possible, for example, to have Asterisk send certain calls out with a prefix (such as **8) to say that those calls should be sent to the Line port (LI) rather than SP1, and then intercept the **8 code in X_InboundCallRoute and direct them to the Line port. Or you could even set it up on your Asterisk server so that if someone dials 00 plus and OBiTALK 9 digit number, your Outbound Route would send that to the OBi110 trunk, which would pass it through to the OBi110, and in the X_InboundCallRoute you could strip off the 00 and then send the call via the OBiTALK network.

      You’d want to spend some time in the “OBi Call Routing and Digit Map” section of the OBi Device Administration Guide to figure out how to do these kinds of tricks, but they are entirely possible.

  2. Bill

    Thanks for the information. I’m tempted to get the 100 model when it appears, just to experiment with some of this off-box call routing.

  3. Admin

    I’ll just point out one other thing: I’ve published a couple of additional articles, the titles of which should be self-explanatory:

    How to use the Obihai OBi100 or OBi110 VoIP device as a gateway between Asterisk/FreePBX and Google Voice and/or the OBiTALK network

    How to use the Obihai OBi100 or OBi110 VoIP device as a gateway between Asterisk/FreePBX and Google Voice and/or the OBiTALK network — Part 2: Using the Phone port as an Asterisk extension

    However, it occurs to me that there’s no technical reason you couldn’t do a full “trifecta” with the OBi110 — that is, use it as a gateway between Asterisk AND the OBiTalk Network, a Google Voice connection, AND a PSTN-type line — and still use the phone port as an extension off the Asterisk server. You’d have to reconcile the three articles and use parts of each (for example, from this article you’d use the Line port settings) and you’d have to figure out how to map your outgoing calls, depending on whether you wanted them to go to PSTN line or to the Google Voice connection (or to the OBiTALK network). There are a number of ways that the latter could be done, but it’s certainly possible once you understand how all this works.

  4. Ray S

    I have the pure dsl product from Centurytel in Missouri (I can only dial 911, all incoming calls are free). i had planned on using the call back feature from Google Voice to make free calls, however, it is impossible to verify the number when GV calls to verify the number, their system will not accept the verification codes, I entered, (believe me I’ve tried many methods). I was able to get GV to forward calls to my almost naked incoming dsl line by spoofing that number to set up a temporary forwarding number (but no call backs from GV to temp numbers).
    How do these restraints mesh with the obi100 or obi110 devices?
    I use a Skypeout number on a linksys iPhone (no longer available) and the voip is awful with that device, Skype on my computer is pretty good. Is the voice quality on the Obi boxes closer to Skype on the computer or Skype on the linksys iPhone (under water bad)?
    Would the OBi110 be a better choice if I could plug the incoming landline into the phone jack on the device and be able to receive the incoming landline calls through that jack?
    Thanks,
    Ray S

    1. Admin

      Ray S – without commenting on how well any VoIP device will work with Centurytel DSL (because I honestly don’t know), I will tell you that an OBi110 should do what you need it to do. You can directly access Google Voice, so incoming and outgoing calls should be no problem (you don’t even need the Centurytel line for those), and there’s no verification call needed (assuming that you already have an active Google Voice account, which I’m assuming you do). If someone dials your Centurytel number (and you have the incoming line connected to the LINE jack on the device, NOT THE PHONE JACK), it should ring your phone that’s connected to the PHONE jack. And if you dial 911, by default that will go to the LINE jack connection because Google Voice doesn’t support 911, so they route 911 calls to the LINE jack by default. So in short, once you get your Google Voice account configured, I am fairly certain it will work exactly the way you need it to, unless I’m not completely understanding your questions here.

  5. Ray S

    Thank you, I believe you understood my convoluted question nicely, and I applaud you for that. Your explanation helps and I’m going to order the OBi110.
    Thanks again for a well written explanation setting up the many options, I love digging into these things, there is a lot to learn.
    Ray

  6. RobF

    After reading this I’m thinking my approach is the right one but I’m not quite able to get things working properly all around.

    I’m doing things a bit differently using SP1 in the obi for my FXS registration since I’m managing GV from my asterisk server. I have the FXS registering as an extension and that appears to be working as I can call it from a softphone on my pc.

    My goal is that asterisk manages the trunks and the extensions so I want to set the obi to register the FXO as a trunk and the FXS as an extension.

    Part of that appears to be working as I can dial other extensions from the FXS endpoint and asterisk shows the sip peer as registered. As I said, I can also dial that extension from a pc based softphone.

    If I dial my POTS line from a cell phone then my FXS port rings as I have set an inbound route for that extension. So inbound seems to working properly too.

    The problem occurs when I try to dial out. Asterisk doesn’t appear to match the outbound route… or so it seems and it ends up at the IVR saying “Please wait while we find someone to take your call” and then ext 701 rings…

    So dialing out the of the POTS trunk is failing at this point.
    I only have 2 extensions configured and whichever I try to dial out with the pots trunk the other one ends up ringing after the “Please wait while…” message.

    Strange, but I’m still digging.

    Many Thanks.
    Rob.

    1. Admin

      Rob, my initial impression is that it sounds like you are going into a queue, or perhaps a ring group. Do you have any queues or ring groups set up? Could there be a conflict between the number you are trying to call and the number associated with a queue or ring group?

  7. RobF

    double checked my settings and restarted after tweaking things like disabling unused trunks as the CLI showed it was always trying to use GV trunk even though my route for pots is first.

    When I try to dial out at this point I get “all circuits are busy now”
    I have max channels set to 1 and the context to “from-trunk-sip-trunk_name” as you have indicated in the article.

    inbound seems to work fine… haven’t done any quality tests yet.
    pots line works great if I move back to my production box with a TDM400 and dahdi.

    when

  8. Martin

    You can set this up without using the peer settings on FreePBX. Replace the items below in {} with the thing described inside the {}.

    Trunk Name: TENDIGIT##

    NOTE: Using the 10 digit number you specified in the Obi as the trunk name is critical! It won’t work unless you make it match!

    Peer Details:
    username={USERNAME}
    type=friend
    secret={PASSWORD}
    qualify=yes
    port=5061
    nat=yes
    incominglimit=1
    host=dynamic
    dtmfmode=rfc2833
    disallow=all
    context=from-trunk
    canreinvite=no
    allow=ulaw

    If you use these settings, the USER details are not necessary.

    P.S. I’d love to take credit for this, but I stole it from the SPA3102 configuration I found at the Freepbx site.

  9. Martin

    P.S. It turns out you can leave out the Port=5061 as well.

  10. Adrian

    I just saw the link to my blog from your site. Thanks!!!!

  11. Martin

    A couple more important points.

    If you use your ten digit phone # as the trunk and user name, it appears to interfere with inbound routing of calls that come in using that ten digit # as the caller ID. So, it’s better to use a word, such as “OBITRUNK1″ as the Trunk Name, Username, etc. in the FreePBX Route and in the Obi configuration settings. The only place you should use your ten digit # in the OBI settings is in the dial plan area and the only place you should your use ten digit # in FreePBX is in the inbound route DID area.

    Second, change the context to “from-trunk-sip-OBITRUNK1″ so that inbound calls count against the trunk maximum.

  12. RobF

    I have things working properly for the most part. Call routing is all working perfectly. My POTS, GV number and OBiTalk numbers route in to asterisk and land in the right place. Outbound is working great too.. Many thanks for the valuable posts here that allowed me to get that working.

    The one case that is NOT handled properly is when a call comes into the OBI110 from the POTS line WITHOUT any callerid data. The obi routes as expected to asterisk which is correct. The problem is that Asterisk puts the trunk name in the callerid field before routing back to the obi…. I am assuming because its blank. As stated it properly routes the call back to the obi and to the Phone port there. The problem is that it sets the callerid to the trunk name … Whether I had the trunk set to one of my DIDs on the obi (GV or LINE) or OBITRUNK1 asterisk would “fill in” those values as the callerid when sending back to my PH extension on the obi. So the phone shows a callerid of either my GV number or my POTS line number or the actual Trunk Name set in the trunk.(depending on how I set the obi to login in the incoming context)
    If callerid is provided on the incoming POTS call then that is sent to Asterisk and its handled properly either by blacklist, incoming routes etc and if forwarded to my phone extension on the obi then the callerid is displayed as expected.

    Has anybody else seen this?
    I would prefer that asterisk catch this and allow me to route based on no callerid. Not sure what inbound route I would need but presumably that is where I would deal with this corner case.

    Many Thanks,
    Rob.

  13. Admin

    RobF, are you saying that if the trunk name is OBi110 that Asterisk puts that in the Caller ID NUMBER field before routing it back to the OBi? I guess I’m not quite certain of what’s happening here, but here are a few thoughts.

    First, inbound routes can route calls on both the DID and the Caller ID. The mouseover for the Caller ID Number field states:

    Define the Caller ID Number to be matched on incoming calls.

    Leave this field blank to match any or no CID info. In addition to standard dial sequences, you can also put Private, Blocked, Unknown, Restricted, Anonymous and Unavailable in order to catch these special cases if the Telco transmits them.

    So if you are getting any of those “special” strings you could match on them. Unfortunately, as shown above, Inbound routes treat “any” or “no” CID info exactly the same way.

    Now suppose you added a context in extensions_custom.conf (this is UNTESTED so I’m not sure if it would work):


    [from-trunk-obi]
    exten => _X!,1,GotoIf($["${CALLERID(num)}" != ""]?hascontent)
    exten => _X!,n,Set(CALLERID(num)=UNKNOWN)
    exten => _X!,n(hascontent),Goto(from-trunk,${EXTEN},1)

    Then if you changed your trunk context statement to point to from-trunk-obi, it should go to that context, where if there is a “nothing” sent as a Caller ID number, it will change it to UNKNOWN, which is one of the special strings that you could match on.

    Might be worth a try, anyway.

  14. robfreyder

    Thanks for the prompt reply.. I’ll try your suggestion.

    I wasn’t matching on CID just the DID which is working fine in terms of routing the call.( I’m matching ANY CID)
    I’ll test and let you know what I find.

    many thanks.
    Rob.

  15. Martin

    Okay, so I found that with my prior configuration, I was occasionally getting a situation where the OBI 110 would reboot when I tried to make an outgoing call. I’ve tinkered some more and I think I have it working. Here’s what I’ve got.

    Trunk Name: OBITRUNK1 (NOTE: This must match AuthUserName and words before @ in URI in OBI device). DON’T USE TEN DIGIT # here, or inbound calls from that number to other trunks will fail.

    Maximum Channels: 1

    Peer Details:

    username=OBITRUNK1
    type=friend
    secret={PASSWORD}
    qualify=yes
    host=dynamic
    dtmfmode=rfc2833
    disallow=all
    canreinvite=no
    allow=ulaw

  16. Phil

    I’m probably repeating a question, sorry, but I read a little on this box at the Amazon site and it appears that you are required to enter your Gmail address AND password into OBi’s website.

    Am I correct and does anyone have an issue with that. I don’t want anyone knowing my email password.

    Thanks, these are excellent reviews.

    Phil

    1. Admin

      If you enter your password into the OBiTalk portal, it only uses it to send to your Obi device and then does not save it anywhere on the site (unlike your other settings). If you’re really paranoid about it, you could just insert a dummy password (so it will go ahead and configure the device with the proper settings), then directly access the device’s built-in web portal and enter the correct password and save it.

      But, that aside, I recommend you get a different Gmail account (that you do not use with normal e-mail) for use with Google Voice. Otherwise you run the risk of not getting calls while you are logged into Gmail. If you are careful not to log into Gmail at times when you might be expecting calls AND you always log out when finished reading your mail, or if you use an e-mail client (such as Thunderbird) to get your e-mail via POP or IMAP, then it’s not as much of an issue.

  17. HoangNguyen

    Hi,

    I just bought obi110 last week. I followed instruction from youtube. It is easy to setup but i didn’t see more information. I did the search and found this website. I have some questions and I hope you guys can answer.
    If GV is ended, can you use NetTalk or Magic Jack service to put in Obi110?
    Do you know any other VOIPs (like NetTalk and Magic Jack) which I can use on Obi110?

    1. Admin

      No, the OBi110 won’t work with the two specific services you mentioned (they both use their own devices). It will work with the following providers:

      Broadvoice
      Callcentric
      CallWithUs
      Engin
      Google Voice
      InPhonex
      Sipgate
      Vitelity
      VoicePulse
      voip.ms

      The OBiTALK portal provides easy configuration for all of the above. If you wish to use the device with a service not shown here, you can probably set it up using the Generic Service Provider option, assuming the service you are using allows you to “bring your own device”. For example, a guy in this thread says he has used an Obi110 with Voipo, Viatalk and Voip.ms (presumably not all at the same time). So there will be plenty of other ways you can use your OBi110 if Google Voice service comes to an end (which I consider rather unlikely anytime soon, barring some sort of natural disaster or other unanticipated calamity).

  18. HoangNguyen

    Hi michigantelephone,

    I didn’t expect you replied so soon. Thank you very much. I try to find a cheap VOIP service to replace GV when it is ended. Majic Jack and NetTalk seem to be the cheapest one which can call Canada for free.

    I setup the my Obi110 as a gateway to dial out to Canada. I added my parents’ number to “trusted phone #” and they can use this box to dial out to Canada. I use my Iphone, which has Obion apps installed and had its softphone Obi number, to dial to my Obi110 number but AA of Obi110 doesn’t pick up; however, I did add my Iphone softphone Obi number to trusted phone # in my Obi110 box. How do I configure the Obi110 so that AA can pick up the incoming numbers which they are regular phone numbers (Land Line or cellular phone) and Obi #? I know I can configure the Obi110 from its IP address. I can switch AA to ObiTalk but if I do, the AA of the box doesn’t pick up on regular phone #. Please advise. I am new to VOIP technology. I apologize if my words may confuse you. Thanks again.

  19. Admin

    NOTICE: All comments above this one were imported from the original Michigan Telephone Blog and may or may not be relevant to the edited article above.

  1. Transition from Comcast Digital Voice to Google Voice | The Blog of Peter Filias

    [...] my research, I found this great review over at michigantelephoneblog.com: Part 1, Part 2, Part 3, Part 4. Part 3 covers how to get 911 service, which you don’t get with Google Voice. That’ll [...]

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