Configuring the OBiHai OBi110 to replace a linksys spa-3102 as a google voice and pstn gateway using asterisk 1.4

If you were looking for how to configure the SPA-3102 with Asterisk 1.4 click here.

All the rage in the VOIP world is the Obihai obi110 voice gateway device, and while due largely to its native integration of google voice, I believe it’s because it does quite a bit for a very reasonable cost (Under $50 from here!). And while just good news for asterisk 1.8 users, this is great news for asterisk 1.4 and 1.6 users! The Obi110 is essentially a user friendly linksys spa-3102 with a slightly watered down interface, with the addition of native google voice support (and setup wizards!). It is somewhat less capable in functionality (no router feature, or independent trunking capabilities with asterisk, etc.) but for my purposes, I decided to give it a try to see if I could replace my spa-3102 and found out that it could quite easily.

The obi110

I have a simple asterisk 1.4 based setup with several extensions located around the world. The main goal is for my family to connect to my system using their iphones, computers, voip devices and make calls to the US and also to receive calls with local DID numbers to their extensions. My previous setup with the linksys spa-3102 allowed them to use my ooma device to call out as an outgoing trunk, however I wanted to see if I could add two trunks to my setup and give my users the flexibility of outgoing call redundancy if they so desired (choose to call out through GV or my Ooma).

Below is my guide for setting this up.

-Network connectivity to the obi110 and ability to administer its local web admin page and setup through
-Physically connected pstn service (ooma, telco) to the obi110 local Line port (Note: if line port is disconnected, the obi110 will report the lines as busy even if just calling out through GV when the SP2 local dialplan shown below is specified).
-Newest version of firmware on the obi110 (as of 03/21/11 version 1.1.0 (Build: 1892))

I found that it is much simpler to initially set up the obi110 through the portal and then to disable auto-provisioning and make customized changes to the web admin afterwards.

Google Voice Trunk setup – portal
1. Goto obi110 device portal after you have successfully registered your hardware to your user account on
2. Configure Voice services for Service Provider 1 and select Google voice
3. Enter in your google voice email and password.
4. Make sure to put a checkmark to make this provider the default Line to call out.
5. Click submit and wait for your ob110 to reboot

Asterisk Trunk setup – portal
1. While still logged into the Obitalk portal, configure the Voice service for Service Provider 2 as a generic service provider (this will essentially act as an extension to asterisk)
2. Enter in your asterisk server information
3. Click submit and wait for the obi110 to reboot.

Asterisk Service provider 2 Trunk

Configure Dialplan on Obi110 device and disable Auto Provisioning
Thats all there is to setting up obi110 trunks! Couldnt be easier right? Now for the small dialplan customizations to make the incoming and outgoing calls route properly.

1. Log into your obi110 device through its web admin (enter in the devices IP address in a web browser. default user/pass is admin/admin)
2. Disable Auto-Provisioning by expanding the System Management Tree on the left pane, then clicking on the Auto Provisioning link and set the Auto Provisioning Method to Disabled. Click submit and reboot (if not disabled, your changes will get overwritten).
3. Expand the Voice Services Tree to the left and click on the SP1 Service link
4. Under X_InboundCallRoute enter in SP2 and click on submit (this routes all calls coming in from GV to your asterisk extension).

Service Provider 1 Local Dialplan

5. Under the Voice Services Tree to the left, click on the SP2 Service link
6. Under X_InboundCallRoute enter in {>(xxx xxx xxxx):sp1},{>(1xxx xxx xxxx):li1}  and click on submit

Service Provider 2 Local Dialplan

The above line in Step 6 routes any 10 digit calls out through GV (configured as SP1), and any 11 digit calls starting with 1 out through the local line port…In my case, through my ooma device). The reason you will need to do this is because unlike the spa-3102, there is no seperate interfacing for asterisk to trunk to the different Service providers which the obi110 can be configured. Instead, asterisk needs to send the dialed number in a way where the obi110 can then determine to which interface it will send its calls. In this case, I will have asterisk format the numbers to be only 10 digits if a user wants to call out through GV, and append a 1 to the 10 digit number if the user wants to call out through my ooma. Once asterisk does this and sends the number to the obi110, the obi110 will then direct the call accordingly depending on if there is a 1 in front of the number – a little hurdle, but easy enough to work around.

Double checking your Asterisk configuration settings
1. Under the Service Providers Tree to the left, click on the ITSP Profile B link
2. Make sure the settings are correct for the SIP and RTP sections. Most likely you will have to modify your RTP LocalPortMin and LocalPortMax settings to match your current asterisk RTP port range.

Thats all there is to configuring your local OBi110 device. The next steps are just to configure asterisk with the user you specified in the service provider 2 settings, and to create a dialplan which handles the numbers so that obi110 can route the calls appropriately.

Below is my user configuration for asterisk 1.4 so that the Obi110 can register to it. ‘gvtrunk’ is my username and ‘password’ is the password which I specified in my Service Provider 2 setup on the portal. Place this in your sip.conf or users.conf file.

username = gvtrunk
fromuser = gvtrunk
secret = password
type = friend
disallow = all
allow = ulaw
allow = alaw
allow = gsm
context = gv-in
host = dynamic

Next we have to edit our extensions.conf file and to tell asterisk what to do with our calls. Below, any call the user makes whether be it 1XXX-XXX-XXXX or XXX-XXX-XXXX will be routed out to the obi110 and then through google voice. This is because I have asterisk removing the 1 digit the user has dialed so that the obi110 dialplan will route the 10 digit number to GV.

In this dialplan I also have any number with a 9 dialed before it sent out to the obi110 as a 1XXX-XXX-XXXX number. If the user has dialed a 9XXX-XXX-XXXX number without the 1, asterisk will append the 1 in front of the number to make it an 11 digit number so that the obi110 can route the call through the local line port (and out through my ooma).

_XXXX is needed because my internal extension numbers are 4 digits long. I need this so that any numbers longer than 4 digits go out appropriately and are not seen as internal extensions.

;make outgoing calls to gvtrunk with 10 digit dialing
exten => _XXXX.,1,Dial(SIP/${EXTEN}@gvtrunk,20)
exten => _XXXX.,n,Hangup()
exten => _XXXX.,n,Congestion
;make outgoing calls to gvtrunk with 11 digit dialing
exten => _1XXXX.,1,Dial(SIP/${EXTEN:1}@gvtrunk,20)
exten => _1XXXX.,n,Hangup()
exten => _1XXXX.,n,Congestion
;make outgoing calls to gvtrunk through ooma with 10 digit dialing
exten => _9XXXX.,1,Set(CHEXTEN=${EXTEN:1})
exten => _9XXXX.,n,Dial(SIP/1${CHEXTEN}@gvtrunk,20)
exten => _9XXXX.,n,Hangup()
exten => _9XXXX.,n,Congestion
;make outgoing calls to gvtrunk through ooma with 11 digit dialing
exten => _91XXXX.,1,Dial(SIP/${EXTEN:1}@gvtrunk,20)
exten => _91XXXX.,n,Hangup()
exten => _91XXXX.,n,Congestion

How to handle inbound calls from GV with Asterisk
The last thing we need to do is tell asterisk how to handle inbound calls from Google Voice. In my user setup above, I specified gv-in as the user context. This just tells asterisk where to go and how to handle the incoming calls from the obi110 device. Place this in your extensions.conf file and modify it accordingly.

exten = s,1,answer
exten = s,n,wait(1)
exten = s,n,Dial(SIP/4000,20)
exten = s,n,wait(1)
exten = s,n,Voicemail(4000,u)
exten = s,n,Hangup

Above, I have all calls from GV calling my extension 4000. Just change it to your desired extension.

Thats all there is to it. It may look complicated, but the dirty work is done for you…Enjoy!

Disclosure of Material Connection: Some of the links in the post above are "affiliate links." This means if you click on the link and purchase the item, I will receive an affiliate commission. Regardless, I only recommend products or services I use personally and believe will add value to my readers. I am disclosing this in accordance with the Federal Trade Commission's 16 CFR, Part 255: "Guides Concerning the Use of Endorsements and Testimonials in Advertising."

Post to Twitter

9 thoughts on “Configuring the OBiHai OBi110 to replace a linksys spa-3102 as a google voice and pstn gateway using asterisk 1.4

  1. Michigan Telephone

    I’m glad to see someone has covered this subject for those who use Asterisk but not FreePBX. For those who do use FreePBX, I’ve done a series of articles on my blog about the Obihai devices. The nearest equivalent to your post for FreePBX users is this one:

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

    In later articles I cover how you can also use one of these devices as a gateway (trunk) to Google Voice or another SIP provider, and to the OBiTALK network.

  2. Pingback: Review of the Obihai OBi110 VoIP device, Part 4: How to use the OBi110 as an FXO port with Asterisk and FreePBX « The Michigan Telephone blog

  3. voipnewbie

    Thanks for the great article. Quick question.
    Reading the article, I am not sure whether asterisk (in this setup) can differentiate incoming calls between google voice and your ooma. Does this mean all incoming calls from gvtrunk (eg google voice and ooma) will all go to ext 4000?

    In other words, how can asterisk differentiate incoming calls from gv or ooma?

    Thanks again for the great write up.

  4. Blog Master Post author

    This is what I got from obihais support team regarding differentiating incoming calls to asterisk.

    Set SP1 InboundCallRoute to, say SP2(1234)
    Set LINE Port InboundCallRoute to, say SP2(5678)

    Furthermore, the calls from OBi SP2 to Asterisk/Ext
    have caller-id from the original caller; you can easily config OBi to prefix
    GV incoming calls or PSTN LINE incoming calls to help identifying
    where the calls originated.

  5. Keith T

    I’ve got a similar configuration (OBI 110 + Ooma). I’ve configured mine the same as outlined above but do not get caller ID in FreePBX on an inbound Ooma to OBI FXO. The caller ID is being passed from the Ooma because I can see it in the OBI dashboard history and well as on a phone plugged in the OBI phone port. I’m replacing a dead SPA3000 that handled caller ID perfectly and sent it to the PBX, so I know the setup SHOULD work. Any idea what I’m doing wrong?

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

  7. Pingback: Nerd Vittles » netTALK to the Rescue: Dumping Google Voice for Less Than 10¢ a Day

  8. York

    Any idea why I am getting rejected when calling from asterisk side? I expect the call goes through SP2 and forwards to GV on SP1. From my asterisk console, I can see the INVITE message got a response SIP/2.0 404 Not Found. Do I need to do anything on OBI to accept SIP/12125551234@obi? Here “obi” is the name of my obi device in asterisk.

Leave a Reply

Your email address will not be published. Required fields are marked *