Asterisk is one of the coolest pieces of open source software that I have come across. Its possibilities are endless, and its almost completely free (aside from all the cool gadgets you buy to expand its functionality). The reason for this blog post is to provide a better guide for setting up asterisk to communicate with an spa-3102 and interface it with ooma. In this setup, I have asterisk 1.4 running on an NSLU2 running unslung 6.10beta. Being that the nslu2 does not have much in terms of support for fxo/fxs built in or through its USB ports, the super handy dandy and small form factor of the SPA-3102 is a perfect option to get an FXO port to interface with asterisk via ethernet and be able to make and receive calls with asterisk to the PSTN (in this case to dial out and receive calls from my ooma hub).
The real motivation for me to use asterisk and ooma was to save moolah. With ooma, albeit with the up-front cost of $200, one can cancel their phone line and stop paying those pesky monthly bills. Our monthly bills were not as absorbitant as others, yet the reasoning behind me getting it was to get more for the same. I canceled my ATT $10 internet (768k down) and $10 phone with only local calling…(total monthly charge of $32-$37 with taxes and long distance charges that we didnt make), and opted for a 12mb down 512k up cable internet connection for $37.99. I would then still have a home phone number which I could take with me if I moved, and have super fast internet (in my standards). With asterisk in the mix, I could then share out my ooma with my family and allow them to make calls to the US for free as well (my sister lives in australia). Also, if I am overseas, I can make free calls to the US. I also have gizmo5 and google voice working together to provide my individual extensions in asterisk with DID numbers, but thats for another post.
For those of you who do not know what ooma is, its a VOIP hardware solution which gives you a dedicated phone line and “unlimited” calling to the US, all you need is an internet connection. I got my ooma core from radio shack for $199 and will never have to pay phone bills again (ooma core does not have an annual regulatory fee, while the ooma telo, and ooma hub only, does). Ooma makes its money off the upfront cost of hardware, and also by selling its ooma premier service. This service gives you cool features, but features that asterisk allows you to do and with more customization (and maybe for a little more effort). The Linksys SPA-3102 is an ethernet voice gateway with FXO port that has the added functionality of routing, and it also acts as an ATA to allow your regular analog phone to connect with a VOIP provider using the FXS port (If you don’t have an SPA-3102 yet, you can get one from here).
Ooma hub wiring setup
The Ooma hub can be hooked up to your existing phone lines in several ways. Currently, I have my ooma phone port plugged directly into my existing home wiring jack with a splitter which also has my fax/answering machine plugged into it. This configuration allows all the phone jacks in my home to access the ooma hub without the use of the ooma scout. This is essentially the same wiring configuration as one receiving phone service from the telco. However, you lose the instant second line feature you would otherwise be provided when using the ooma scout adapter. To connect the SPA-3102, just plug a phone cord from the Line port on the adapter to a jack in the wall, or if it is near the ooma hub, into a splitter which shares a line plugged into the Phone port on the Ooma hub (or directly into the hubs phone port without a splitter). If you are on a call using a phone plugged into my wiring configuration and dial out using asterisk through the Line port on the SPA, the adapter will report a 503 message to asterisk and stop the call from taking place and interrupting.
If one wants to make sure the line is not busy when receiving/making a call when using the ooma as a regular analog telephone line as well, another configuration one can use, is to hook up the ooma scout and connect it directly to the ooma hubs ‘wall’ port via phone line. One would then connect a phone line from the SPA-3102 Line port to the scouts ‘wall’ port. This enables the scout to communicate with the ooma hub and enables the instant second line feature should the first line be active when a call out from the asterisk box takes place.
Linksys SPA-3102 Configuration
SPA-3102 Remote Management for LAN Setup
The SPA-3102 has four interface ports in the rear, Internet, Ethernet, Phone, and Line. If you plug in a computer to the ethernet port via cable, it will provide your computer with an ip address with which you can then enter in the gateway address from an ipconfig and hit the spa3102 web gui. With this web gui, youll be able to configure the device. We dont want to have to plug in a cable each time to configure the device, so we will enable the web interface on the spa3102 when it is connected to the Internet port (with which it will receive a dhcp address handed out to it from your router currently on your network).
-Log into the webgui for the spa-3102 when you are connected to the ethernet port
-Click on the admin and advanced links at the top right to get the elevated setup access
-Goto ‘WAN Setup’Â Tab
-Change ‘remote management’ option to ‘yes’
-Click the ‘submit all changes’ button at the bottom.
-Connect your spa-3102 to your network via the internet port.
-Log into your spa-3102 and look at the status screen with the computer still plugged into the ethernet port on spa3102. You will find the dhcp address the spa-3102 received from your router which is connected to the Internet port.
-Disconnect your computer from the ethernet port on the spa-3102
-Log into your 3102 via the dhcp address that it received from the internet port.
SPA-3102 PSTN Line Setup:
Now we begin the configuration of the SPA to be used with asterisk. In this setup, I will not enable Line1 which makes the SPA-3102 an ATA adapter as well (allowing calls made to your voip provider to ring the analog phone connected to the PHONE port). In this setup, I just use the SPA as a gateway which allows me to make and receive calls (using the LINE port on the SPA) from any extension that is connected to my asterisk pbx. Under the LINE 1 tab in the SPA, ive set “Line enable” to no.
NOTE: When things are configured properly, and the PSTN Line is registering with asterisk, the LINE LED on the SPA will light up and remain lit (same with the Phone port if Line 1 is enabled). If things arent communicating correctly, the LED will not be lit (I have the spa registering to asterisk).
-Log into the SPA web gui
-Click on the admin and advanced links at the top right to get the elevated setup access
-Click the ‘PSTN Line’ tab
Where you enter in your asterisk server IP info and whether or not it will register to asterisk.
Proxy and Registration
–Proxy – Change 192.168.1.77 to the ip address asterisk is on your network. I put the ip in the outbound proxy, its not necessary as ‘Use Outbound Proxy’ is set to ‘no.’
–Register – ‘yes’
–Make call without reg and Ans call without reg – Change options to ‘no’
Subscriber Information
–Display name – can be anything
–User Id – can be anything but for simplicity sakes when configuring asterisk, use a name without spaces
–Password -Â can be anything
This is where you enter dialplan information, and options to configure sending calls from asterisk to the pstn port.
Dial Plans
This section of dial plans are accessed by the entire page of the SPA PSTN Line tab. There are eight DP fields because it allows you to create different dial plan options to be used throughout this tab. Voip-to-PSTN, and PSTN-to-VOIP sections both reference these dialplan fields as ‘DP.’ As you can see in my screenshot, Dial Plan 2: is filled out. In my setup, this command tells the SPA that any calls answered after the PSTN-to-VOIP gateway option answer delay is reached, to be sent to the S extension in asterisk. You may enter any extension such as (S0<:102@asteriskIP>).
VOIP-to-PSTN Gateway Setup
–VOIP-to-PSTN gateway enable – ‘yes’
–Line caller DP – set to ‘1’ (this option references Dial Plan 1: and the default (xx.). This just passes anything sent from asterisk to the SPA without any change)
–One Stage Dialing – set to ‘yes.’ If set to no, then the SPA uses 2 stage dialing, and it screws up asterisks calling out to the Line port.
The PSTN-to-VOIP configuration, where you configure how to send calls from the Line port (pstn) to asterisk
PSTN-to-VOIP Gateway Setup
–PSTN-to-VOIP Gateway enable - ‘yes’ (in the screenshot above, i have it set to ‘no’ as I do not want the spa to pick up the line and forward to my extension s@192.168.1.77 as defined in Dial Plan 2:. This essentially turns off any calls going to my asterisk system. I have my asterisk system setup to forward the call from the pstn to my cell phone when this is turned on, and only used while traveling far away from home. When I am not traveling I have a fax/answering machine on my ooma and want it to pick up instead, so it is disabled.
If you are in an asterisk/voip only configuration and want all calls to be routed straight to your asterisk system without worrying about any analog answering machines or fax picking up/ringing, then set to ‘yes’
–PSTN CID For VoIP CID – set to ‘yes’ if you want callerid to be passed onto your asterisk system
–PSTN Caller Default DP – set to ‘2’ as in my Dial Plan 2: it allows calls to be routed from Line port (pstn), to extension S on my asterisk pbx
FXO timer values (sec)
–PSTN answer delay – this option is to change the length of time the SPA-3102 picks up the call coming in from the PSTN and forwards it to your asterisk system. The default is 16 which allows the line to ring for a little too long before sending it off to Asterisk. A number of 3-5 should be good for callerid to be gathered and sent along with the call to asterisk.
–PSTN Dial Digit Len – set to .1/.1 otherwise calls may take longer to start connecting. This essentially shortens the speed at which digits are dialed at. You dont want digits to take forever to be entered do you?
Awesome. You are now finished configuring your SPA-3102 to act as a SIP trunk on the SPA-3102 side. All you need from here to configure asterisk is the username and password you specified in the Proxy and Registration section. Now on to the Asterisk side of things…
Asterisk Configuration
This setup is for asterisk 1.4. I found that many guides found on the internet do not seem to work for my setup. I was not able to set my SPA as a peer, but had to configure my 3102 to register to asterisk as an extension in order for everything to work correctly. I also found many internet guides had sip trunk settings which were no longer used for version 1.4. Now lets tell asterisk theres a device to communicate with in the Users.conf (for you it might be Sip.conf) file in the asterisk directory.
Users.conf or Sip.conf
Users.conf or sip.conf configuration in asterisk. this command sets up the SPA to be used as an extension from which calls can be made and received.
–[pstn] – Put the username you specified on the SPA-3102 in between brackets. In my example above, replace [pstn] with your username. The tricky part here is that the name between brackets is your username, even if you specify username = as something else, asterisk will not allow your 3102 to register with it.
–type = friend – Sets as an extension which can be dialed out from (see here for more info)
–port – I saw several guides saying to put in port = 5061, this is unecessary as this configuration automatically connects to this port. You may need this if it was set to type=peer, but I was never able to get it to work as one.
–disallow – This configuration also disallows any voice codec other than ulaw, alaw, and gsm as the NSLU2 does not have enough horsepower to transcode the higher compression the other codecs use.
–context = pstn-in – this is the label for where incoming calls are sent to in my extensions.conf file.
–host = dynamic – as the spa is registering to asterisk, the ip address of the 3102 does not need to be specified here and will be obtained during the spa-3102 device registration. I could change the ip address of the spa-3102 on the unit and it would still register with asterisk.
–secret = passwd – replace ‘passwd’ with the password you entered in the spa-3102
Other than that, this is all thats needed for the PSTN Line to register to asterisk and send and receive calls.
Extensions.conf
Extensions.conf is the the file which tells asterisk how to handle incoming/outgoing phone calls.
Lets configure outgoing calls first, as it requires very little configuration. Below is all i need to put into my default context in order to make outgoing calls out of the Line port (we previously specified ‘pstn’ as a trunk/user in users.conf/sip.conf) on my SPA-3102. Below, the _XXXX. tells asterisk that any number dialed with more than 4 digits should go out through the SPA (I have 4 digit extension numbers configured). Without this configuration, asterisk would try to place a call to an internal extension number, only to find that the extension (the phone number dialed) did not exist.
Simple outgoing dialplan used in the default context in my extensions.conf
In my users.conf (maybe your sip.conf) file, I had the pstn user (pstn is the username specified in the spa3102) use context pstn-in as the label to goto which contains the code for how to handle the calls coming in from the Line port or PSTN.
Extensions.conf tells asterisk how to handle a call. pstn-in is the context defined in my users.conf/sip.conf
In my spa-3102, my Dial Plan 2: had code (S0<:s@192.168.1.77>). This essentially sent the incoming call from the PSTN Line to the S extension in asterisk. Asterisk knows that the call is coming from the pstn user defined in the users.conf/sip.conf file, and found that context pstn-in was specified. It then initiated the commands under the pstn-in section. The commands listed above, answers the call, plays a sound file that nobody is available to take the call, then says “call-forwarding” and proceeds to ring my cell phone number (8001112222 is the cell number, change it as desired. proxy01.sipphone.com is my gizmo account which forwards it out through a google voice DID number).
Alternatively, if you just want to have the incoming pstn call sent to several or all extensions on your asterisk system, you can substitute my ‘Dial’ line with the one below and change my extension numbers to match yours (where 6000, 6100, 6200 are my extensions/phones registered with asterisk):
exten = s,n,Dial(SIP/6000&SIP/6100&SIP/6200,20)
Thats all there is to it, good luck!
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
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).
Thanks for your detailed explanation. I will give it a try and let you know how it goes.
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
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.
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
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!
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.
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.
Thanks,
I have one more question, how many people can call the pbx at once with ooma?
Thanks
Carlos N.
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.
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.
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!
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-
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.
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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.
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.
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!
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!
Any idea how you could make this work on a asterisk system like trixbox or similar?
it should be the same to set up as long as the underlying version of asterisk is the same.
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?
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.
I have a problem :
I am able to receive incoming calls from PSTN to the phone connected to phone port of the spa. But when I tried to make outgoing PSTN calls from the phone (it is set to make calls to PSTN unless number starts with # in which case it goes to voip) I hear a ring tone, but calls don’t go out. The ooma history also doesn’t show it.
I tested this with another spa Line (which acts as phone service) connected to the Phone port of the above spa and with that it works fine.
So I am guessing it has to do with some voltage/cadence or other settings ? (incoming calls from ooma to spa are working though).
there could be plenty wrong…id start first with looking at your outbound dialplan and making sure everything is correctly specified. If asterisk isnt even sending the calls out to the SPA, it wont dial out.
try setting the asterisk verbose to 10 and dial out with your asterisk and watch the output. you should be able to trace what asterisk is doing use what you find as a jumping off point.
I have an Ooma Telo.
I also have an Ip phone system.
How do I make it so that the Ip phone can use the Ooma line too.
Hi! Thanks for the post. I am very interested in it however, I am not a very techie person. From What I could understand after I do your setup, Could my mom from manila call me in Canada? Would she need an spa linksys box as well?
To simplify things, why not get two ooma boxes? Calling is free from ooma to ooma numbers. My friend got an ooma box and lives in manilla and gets great quality calls to/from San francisco for free
Thanks for you very interesting article, did you know if the other Linksys Ata model PAP2T can work in this configuration?
Thank you