How to setup Asterisk with Ooma voip using a Linksys SPA-3102

23 Comments

• 

  On May 17, 2010 at 11:47 am Suresh said

  Hi ,

  Nice article.

  I have Nslu2 , ooma core and planning to buy Linksys SPA-3102 and follow your guide to setup my system.

  Once I have this one , how can my friends can call ( do they need anything from their end except a phone ?) from outside USA and how it will route to my USA OOMA number or my cell number. could you please explain .

  appreciate your help on this.

  Thanks
  Suresh

• 

  On May 20, 2010 at 9:29 am Blog Master said

  Hi Suresh,

  All you will need is a phone ( i like the linksys spa942, and grandstream bugetone bt-100) or a softphone like xlite to register with asterisk (wherever you are in the world), and make outgoing calls through your ooma device.

  You will just have to add extensions in the users.conf (or sip.conf) for each phone that will be registering with asterisk, and like in my above example for [pstn], send them to a context that contain outgoing call commands that you define in the extensions.conf file.

  For example, if aunt sally in australia wants to call out through your ooma to call anywhere in the US (such as your cell phone), youll create an extension for her in your users or sip.conf and then define context = auntsally. Next, go to your extensions.conf file and make a context called [auntsally] and under that line just type in the lines I provided above for ;make outgoing calls to ooma pstn.

  An additional and easier way to do this provided you have the lines under ;make outgoing calls to ooma pstn in your default context ([default]) in your extensions.conf, would just be to point to the default context in the section related to auntsallys extension in the users.conf or sip.conf file. Using this method is quicker, but you relinquish customization for call handling (limiting international calls for example) per extension since everything you do to the default context applies to everyone unless specified in a specific context.

  Another way to do this is by utilizing the include = command. You may use the include = default command to add whatever is in the default context to be available to [auntsally] context. Just add that line under the [auntsally] context in extensions.conf and you wont have to type all those lines under ;make outgoing calls to ooma pstn, but remember that everything in the default context now applies to the auntsally context.

  One thing to keep in mind with the configuring of phones over the internet is that you must have your RTP ports forwarded to your asterisk box (in this case your nslu2), and then define the rtp port range in the phones that you are setting up remotely. You can find (and define) your rtp port range by editing the rtp.conf file in the /opt/etc/asterisk directory on your nslu2 (personally i find the rtp range too large and limited it to 5061-5090).

• 

  On May 24, 2010 at 7:50 am Suresh said

  Thanks for your detailed explanation. I will give it a try and let you know how it goes.

• 

  On August 23, 2010 at 2:41 pm Hany Philip said

  I think you know a lot about the Cisco SPA3102 and I hope you can help me out
  I have Ooma core at home and I am very happy with the service. I have an alarm system that is always having problems (constantly sending invalid signals to the central monitoring station)

  This all started after I switched to Ooma aka VOIP

  I read somwhwere that an alarm system is designed to send its signals over an analog phone line. To transmit emergency signals properly using VOIP, the signal must be converted to digital, then converted again to analog. It is during this conversion that problems develop. Usually the signals arrive at the central monitoring station with errors, or not at all.

  Can the Cisco SPA3102 help me with this problem? If so, how does the wiring work?

  Currently, I have the Ooma connected to my router and then I have a splitter in the wall jack where one cord goes to a nearby phone and the other line is connected to the phone port of the Ooma. Therefore, the whole house is wired to the Ooma

  Thanks in advance for your help

• 

  On August 24, 2010 at 10:41 pm Blog Master said

  Hi Hany,

  Unfortunately, thats one of the limitations/difficulties of VOIP. Its not necessarily the analog/digital conversion, its the latency issues thats inherent with internet data traversal. You see, data is sent through the internet through tcp or udp packets, it has to bounce from one switch/router to the next, sometimes over several hops… TCP is a controlled protocol which has error correction, and udp doesnt really have anything like that. TCP will check for lost packets of data and retransmit when necessary (even if it does, itll end up getting to its destination later than the rest of the voice data has arrived, resulting in errors). UDP just spits everything out and hopes for the best. VOIP traffic normally isnt sent via TCP, but UDP. Sometimes, UDP packets get lost on the way to its destination, and you have lost data. Sometimes using ooma, you find that your voice is cut off, or voice is garbled a bit. Thats because the internet is screwing your voice data, and those datagrams have been lost.

  Thats why ooma says to dial *99 to send a fax. Faxes work just fine using analog lines when the signal doesnt get garbled or the data isnt received later than it should be. *99 disables oomas qos (qos reorganizes voip data based on priority) type of stuff to allow faxes to work. Sometimes it works, and sometimes it doesnt. Depends on the network congestion at the time. So I take it that your alarm system uses some kind of communication much like a fax. Using the SPA-3102 wont help you because in the end, your connection will be still be routed through the internet and not through an analog line. The spa will allow you to dial out using a regular pstn, but you got ooma so you dont have that. It also allows you to receive calls from the pstn, but again, you dont have that anymore since youre using ooma.

  Your best bet would be to try and configure your alarm system to dial out with the *99 prefix, or talk to your alarm system company and see if they have any configuration or adapter that allows better communication via VOIP. If they are a big company, they should have something like that. Hope that helps.

• 

  On December 4, 2010 at 8:10 am ukhan said

  Hi,

  Really great article. I am considering buying a Ooma Telo VoIP device. My wife and I live in the U.S. and we call abroad a lot and also receive a lot of calls from abroad. There are two options would like to have:

  1. This is a lot like what Suresh was asking. When we receive a call from abroad, could I configure asterisk to present the caller with a menu option to:

  a. Forwarded the call to a choice of cell phone numbers?
  b. Leave a message .

  2. An option to call from our cell phone to the ooma phone and have a secret option(with pin) to have the call forwarded to abroad phone?

  I would greatly appreciate your thoughts and wisdoms. Thanks

• 

  On December 10, 2010 at 11:32 pm Blog Master said

  Hi ukhan, to answer your questions:

  1. Yes, for a. and b. But the way you set this up largely depends on if the people who are calling from abroad are calling your ooma number, or calling using voip phones setup as extensions already connected to your asterisk system. If they are extensions, youll just have to configure asterisk for the call handling to go out through the spa3102, if they are calling your ooma phone number and then asterisk picks up the phone via the spa3102, youll either have to jump through a few hoops (using the ooma scout with another spa3102 to allow asterisk to call out using the instant second line feature) and also have to pay for ooma premiere to keep the instant second line feature beyond the trial period if I remember correctly. Or use a sip provider to call out.

  Setup #1:
  That complicated setup looks like this: international call —> ooma (your ooma #) —> spa3102 —> asterisk —> a second spa3102 —> ooma scout —> ooma second line —> cell number

  Setup #2:
  The easiest method would just be to use another sip provider with asterisk to do the outbound calling.

  it will look like this: international call —> ooma (your ooma #) —> spa3102 —> asterisk —> outbound sip provider (like sipgate) —-> cell number.

  Youll have to setup an auto attendant with asterisk, its essentially like a menu that you get when you call businesses asking you to press 1 for this option and 2 for another option. You will have to do some major reading on how to set one up and program the prompts, but its not to difficult (youll have to read about asterisk dialplan commands). You can even record your own audio files and have them as the prompts.

  Asterisk allows you to do many many neat things with the extensions and call handling features that you write in the extensions.conf file. If you have a good technical background in computers and dont mind tinkering around with command line stuff, you should be ok with this.

  2. Yes, asterisk coupled with the spa3102 will allow you to call into your home number, provide you a prompt when dialing into it, and then enter in the number that you want to call whether be it a (local or international call) then dial out with ooma and only pay the local mobile rates. Once again, this involves learning how to configure the asterisk dialplan.

  youll also have to keep in mind the configuration example I wrote above about calling into the ooma, or using your cell as an extension to asterisk using voip. If you have a smartphone like an iphone, you can install a voip client like acrobits softphone and a hack to make voip work over the 3g network. It will register to your asterisk server at home and then it can dial out through your ooma. If not, and youre calling your ooma number, youll have to use the two spa3102s and an ooma scout setup to make that work.

  Just be warned that ooma if I remember correctly, is having trouble with inbound dtmf tones (although I just tested it on my ooma hub and I can hear dtmf tones just fine), so you cant press the keys on your mobile phone and have it register with asterisk (I just googled it and found that ooma is working on a fix for the telo).

  On a side note, I believe google voice will allow you to do this as well, with pin number, and with no configuration needed. Just sign up to google voice with your gmail account, and then youll be all set to do what you want for question number 2. Google voice can even be integrated into asterisk when used in conjunction with gizmo5…only downside is that gizmo isnt taking new registrations at the moment.

  One last caveat that will render all this cool phone stuff useless is what I just realized happened to me. If your ISP blocks port 5060 and 5061 or VOIP traffic, you will not be able to register with any sip outbound trunks like sipgate and google voice. I had used sipgate and google voice for my DID numbers which called my individual asterisk extensions, but were rendered unusable when my ISP decided to roll out their own VOIP services and block those ports for their internet customers. If these ports are blocked, you wont be able to utilize Setup #2 above.

  So sadly, since last week I am scrambling to figure out a way around this…My only thought is to load up a linksys wrt54g with asterisk, set it up at my parents house who use an ISP which does not block those voip ports, and then have that connection do all the ITSP registering and then route the inbound calls through that connection. Who knows though…

  hope this helps…I kept thinking about this post and more and more details kept emerging…hence the huge reply!

• 

  On February 12, 2011 at 5:53 pm Carlos N. said

  Hello, great tutorial!
  I am a student and am very interested in your setup. I have a ooma Telo, and a Trixbox server. I have been playing around in the server to set up thing locally but now I want to be able to receive and make outgoing calls. I have ordered the SPA 3102 and it will be arriving in a few days. Would you instructions still work with the Ooma Telo and the current version of Astrix/Trixbox?

  Thanks!
  Carlos N.

• 

  On February 12, 2011 at 8:17 pm Blog Master said

  Hi Carlos, yes the instructions will work with asterisk 1.4 based distributions. Trixbox comes in both asterisk 1.4 and 1.6 flavors. If youre using a 1.6 based version, I believe you just have to change the peer type to make it work correctly, but I would try it without changing anything in my examples first, and only do so if there is a problem.

  The spa3102 will also be able to work with the telo.

• 

  On February 14, 2011 at 6:02 am Carlos N. said

  Thanks,
  I have one more question, how many people can call the pbx at once with ooma?
  Thanks
  Carlos N.

• 

  On February 14, 2011 at 9:53 am Blog Master said

  Only one call at a time after your ooma premiere trial expires. You’ll have to get an ooma scout to have it work with the instant second line feature to have two calls at once but that requires buying the premier yearly subscription. I think you are better off using sipgate or another low cost sip trunk if your want several outgoing calls at once. Plus with sipgate you also get free inbound calls to a DID number. So really you get two seperate phone numbers people can call into and dial out from. It’s pretty sweet.

• 

  On February 14, 2011 at 11:26 am Carlos N. said

  I have looked in to sipgate and I will be signing up for sipgate one. As for the ooma I cant return it so I will use it. I don’t mind paying for the Premier as I think Ooma is such a great device and I recommend it to everyone. You said, “Only one call at a time after your ooma premiere trial expires.” Does that mean that if I have Ooma Telo (premier) I will be able to use both ooma numbers to be able to have two people call the Astrix box at once? Even though he hub is not can’t be used with the Telo.

  Thanks
  Carlos N.

• 

  On February 14, 2011 at 8:52 pm Blog Master said

  Hi carlos,

  My mistake, Ooma telo does not use the scout, and therefore cant be used to do the instant second line out with trixbox. Telos use the telo handset which does NOT allow you to plug in another spa3102 to make two outgoing calls out with trixbox through ooma, like the scout does (see my response to ukhan above for info regarding using the ooma hub/scout setup). So youre pretty much stuck with using sipgate or another sip provider instead of the ooma telo device for multiple outgoing calls with trixbox. You should still definitely use it for one outgoing trunk though, since it provides free US calling and very low international call rates.

  It sounds like you would like several DIDs (basically phone numbers) by which people can call your extensions, and also numbers which you can dial out from. What I did before my ISP blocked port 5060 and 5061 was that I registered three sipgate accounts (one for each family member), and then tied their asterisk extensions to those DID numbers (when one called phone number 1 it called me, when one called phone number 2 it called my sister in australia, etc..these could also be called at the same time). These three sipgate accounts also act as sip outgoing trunks so I could configure my dialplan to route outgoing calls through any of those numbers (provided there was money in those accounts) in case my ooma line was already tied up with an outgoing call (the more trunks you have, the more simultaneous outgoing/incoming calls you can have).

  But, after my ISP blocked VOIP ports and effectively disabled any VOIP SIP based ITSP service, I had to resort to using Ipkall and IAX instead of sip trunks for my DID numbers, and only use ooma for my single outgoing trunk. I am effectively limited to one outgoing call at a time placed through ooma (unless I buy another spa3102 and use the ooma scout/ooma premier). I am fine with just one outgoing trunk since my family members just call their internal asterisk extensions to talk to each other from wherever they are.

  So you might have to weigh the cost/benefits of what your needs are in terms of necessary outgoing calls vs whether or not your users can just call each other using their internal asterisk extensions. Hope that helps, and sorry for the mix up with the ooma telo/scout!

• 

  On February 26, 2011 at 7:27 pm Andy said

  Hi,

  Wow, you know a lot SPA3102, Ooma and telecommunication.

  I have Ooma telo and verizon landline with bare bone service (10c per minute for local call)

  Can SPA3102 help me to route all outgoing calls (local and long distance) to Ooma but 911 will use landline and in case of internet down or power down, SPA3102 will route all calls to landline?

  Ooma can intergrate my landline with Ooma telco but all local calls (including 911) will use landline and long distance calls will use Ooma. I don’t want to do that because it cost 10c per minute for local calls.

  Thanks,

  Andy-

• 

  On February 28, 2011 at 1:37 pm Blog Master said

  im not exactly sure about what you are asking, but the spa-3102 allows you to customize your dialplan to allow for calling etc. I would look at the capabilities of the spa-3102 dialplan in order to find out if it does what you are looking for.

• 

  On May 2, 2011 at 2:56 pm charles said

  First I want to say thank you so much for this guide. In our home office we have a Ooma Hub MODEM connected to our NETGEAR router and the Ooma Hub is also connected to our SPA via PHONE > SPA LINE. Our Hub is connected to our Scout via Hub WALL > Scout WALL and Scout Phone > SPA Phone. We entered in all the settings in your guide and still are having troubles with the Line light on the SPA not lighting up.

• 

  On May 4, 2011 at 11:32 pm Blog Master said

  Hey there,

  There might be a few things you can check to make sure the configuration you have is working correctly.
  1. Test to see that a regular phone hooked up to the scout (connected to the wall port) allows you to access the instant second line on the ooma when calling out through it.
  2. Make sure you can call out by connecting a regular phone to the ooma hub phone port.

  If both these tests allow you to make calls out through your ooma, then the configuration problem resides in the SPA(s).
  If they do not work, youll have to make sure you have the ooma correctly configured and able to dial out. As well, if you are using the instant second line with the scout, youll need to make sure you have the ooma premier service paid for and activated.

  Are you using asterisk 1.4 to register with the SPA units? If so, make sure they are registering with asterisk otherwise the LINE LED may not light up as well. Let me know how it goes.

• 

  On May 6, 2011 at 11:51 am Yvette said

  Hi Blog Master,

  Thank you for your generous sharing! We are a small business here. Does Asterisk provide the functions of automatic extension transfers and music onhold?

  Thank you!

• 

  On May 6, 2011 at 1:38 pm Blog Master said

  Yes sir. Asterisk does that and more. Email voicemails, ring groups, conference rooms, and even call files which make calls unattended and can play a pre recorded message!

• 

  On March 14, 2012 at 5:53 pm Matt said

  Any idea how you could make this work on a asterisk system like trixbox or similar?

• 

  On March 18, 2012 at 10:07 pm Blog Master said

  it should be the same to set up as long as the underlying version of asterisk is the same.

• 

  On May 6, 2012 at 4:49 pm akavMAC said

  Sorry to bring back up souch an old post. However I’ve followed this tutorial to setup my OOMA with Asterisk. However I don’t know if you’ve looked into this issue. When Asterisk picks up the call, if I have someone in the house pick up the Telo handset, or if the call is answered with the telo handset, is there anyway to get it back to Asterisk? Right now if I receive a call and use my SIP phones then pick up with the Telo there is no way to return the phone back to asterisk? Is there any solution to this or is this inherent of the specific setup?

• 

  On May 14, 2012 at 10:40 pm Blog Master said

  as far as i know, no. Once a handset picks up the call, it will effectively disconnect the line from asterisk. Without being able to have asterisk be notified to pick up another call, there would not be a way to get the line connected to asterisk. I would suggest not having any handset on the telo, and instead have all the calls routed through asterisk use only voip phones.

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