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Communications Google Open Source Software Upgrades IT News

Asterisk 1.8 Released With Support For Google Voice 83

Thinkcloud writes with a note that long-standing open-source VoiP software Asterisk has just been updated, and it's packed with more than 200 enhancements, security updates, and new features — including calendar integration and support for Google Voice and Google Talk. Asterisk's fully-featured PBX includes call waiting, hold and transfer, caller ID, and other useful tools so it's a great option for small businesses that need to watch costs."
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Asterisk 1.8 Released With Support For Google Voice

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  • by Animats ( 122034 ) on Saturday October 23, 2010 @06:53PM (#33999818) Homepage

    Google Voice is useful and fun, but its interface with the telephone network remains flaky. See "Can't send text message from sprint phone to my Google voice number" [] and "Google number not callable from certain numbers" [] Google Voice seems to have an ongoing problem keeping their blocks of phone numbers properly visible to other carriers. Troubles have been reported with Sprint, Verizon, and T-Mobile numbers for over a year now. There are also some limitations on calling into and out of Google Voice from VoIP systems.

    The problems may stem from the fact that Google didn't implement Google Voice. It's just "Grand Central", acquired and re-branded. It's not a "telco quality" system. It's not ready for prime time as your main phone system for a business.

  • by sampas ( 256178 ) on Saturday October 23, 2010 @07:25PM (#33999994)

    I replaced my home landline with an Asterisk box running on a Supermicro Atom D510 mboard, specifically PBX in a flash [], which is the Cliffs' notes version of FreePBX []. FreePBX is based on Asterisk, but provides a spiffy web interface for configuration that's more advanced and free-er than the others. That said, you'll still need to be comfortable at the command line on Linux and a text editor such as vi.

    With Asterisk, you can do voicemail, have your voicemail emailed to you, have multiple boxes, pay $1.50 per month for a phone number plus $.01/minute for calls with a SIP provider such as Vitelity []. You can have conference calls (you'll need to pay $10/channel for g729 if you want to scale at all on home bandwidth, though.)
    You can have ring groups, different extensions, etc. I have one for emergency late-night network issues, which only those with the secret extension can access to wake me up.

    There's an unlimited number of stupid tricks you can do, but you'll need to learn the difference between trunks, routes, and dial plans. That said, it's pretty cool. But then you'll want to buy Aastra SIP phones, which come with open-source phone applications, so it will cost you more. If you want to light up your in-house phone lines, it's $200 for an FXS card. If you want to use an existing landline as a trunk, it's $200 for an FXO card. (Double check which is which before you buy because I can never remember which is which.)

    The best part about running your own PBX is that (1) you can send telemarketers to voicemail hell and (2) it's trivial to fake callerID, which helps with (1).

  • Re:Vs Freeswitch (Score:5, Informative)

    by kasparov ( 105041 ) * on Saturday October 23, 2010 @07:54PM (#34000230)
    You are wrong. Asterisk 1.8 supports SRTP. I know, because I merged it myself. :-p
  • by kasparov ( 105041 ) * on Saturday October 23, 2010 @08:04PM (#34000294)
    You get an inbound number with Google Voice. You can now have that routed to an Asterisk box. From there, you could do any kind of filtering of the call you want. You could have Asterisk check your calendar to see if you are currently in a meeting and handle the call differently from there. You could get a SIP client for your smartphone and register it to your Asterisk box at home and then make make free outbound calls with Google Voice w/o having to have any voice plan. Asterisk makes it possible to do just about anything you want with a call. If you want to set it up so you only receive calls from one group of numbers M-F 8-5, and want only certain contacts to be able to call you after midnight--no problem. If you are a control freak, you should love Asterisk.
  • by wrook ( 134116 ) on Saturday October 23, 2010 @08:26PM (#34000442) Homepage

    As others have said, Asterisk becomes much more obvious if you have an ITSP (Internet Telephone Service Provider). Here's an example of what I have done with it. I moved to Japan 3 years ago, but I still wanted to keep in touch with my friends. Calling long distance to/from Japan is expensive, no matter what plan you have. So I bought a DID (Direct Inward Dial) for my old home town. This gives my friends a local number to call. It routes over the internet to my Asterisk box and rings a softphone on my computer in Japan. The DID costs me $5 a month. Of course, there is a huge time change between Canada (where I moved from) and Japan. Asterisk has voice mail. If my phone isn't running on my computer, Asterisk takes a message and emails me the contents. When I wake up in the morning, I can listen to the message from my email and call the person back. Outgoing calls cost me 2 cents a minutes to North America and there are unlimited plans with many ITSPs (I don't bother because I don't make many outgoing calls).

    Even without an ITSP Asterisk is useful. Perhaps you are used to using Skype or Google Talk to make computer to computer voice calls. Asterisk lets you talk to your friends using SIP (and now I guess Google Talk), but still have all the PBX features. So for instance, if your friend wants to send you a voice mail via SIP they can. You can set up conference calls fairly easily as well. You can buy very inexpensive USB handsets that look like telephones and hook them up to your computer. If you set your softphone's audio device to the handset, it ends up working pretty much like a normal phone. Or you can buy a SIP handset (a bit more expensive) and simply plug it in anywhere you have a network connection. This allows you to set up as many extensions controlled by your Asterisk PBX as you want. It's handy if you have kids, especially since DIDs are really, really, cheap.

    Finally, for some fun you can easily set up ring groups on Asterisk. Talking on a cell phone is generally expensive. Instead, you can set up a DID for your Asterisk box and everyone can call you there. If you have your softphone up, it will ring that first. If it isn't up (or you don't answer it) you can get it to call your cell phone with an outgoing call. You can even set up a voice mail menu that asks the calling person if they would rather leave a message or try your cell phone. And to be even fancier you can vary the response based on who's calling. If it's someone you don't know you can direct them to voice mail immediately, if it's someone you don't care about much you can just allow them to ring the softphone, if it's likely to be important than you can forward to your cell.

    This should give you a few ideas. There are really an endless number of options. Especially since you have source code with Asterisk you can make it do whatever you want.

  • by wolrahnaes ( 632574 ) <> on Saturday October 23, 2010 @08:32PM (#34000482) Homepage Journal

    I've been intermittently experimenting with VoIP over mobile networks for the past few years, and it seems very carrier dependent. Myself and a coworker have both used softphone clients on jailbroken iPhones over the AT&T network, both EDGE and 3G, with decent success. EDGE was a bit flaky with G.711 and G.722, but G.729 was solid and 3G worked fine with all. Latency was a bit unpredictable on both, though much more stable on 3G (150-400ms vs. 200-900ms). On 3G it's never made a call intolerable.

    For the past few weeks I've been using Sipdroid on an HTC Evo on the Sprint network. I've had it on Sprint native 3G as well as roaming on Verizon 1xRTT and in both cases had usable G.722 and G.711 calls. I've never had to fail down to a sub-64k codec and I've even used it while in a moving vehicle (which tends to hurt mobile data performance).

  • by adamstew ( 909658 ) on Saturday October 23, 2010 @09:00PM (#34000650)

    Asterisk has AGI. Think CGI, but through asterisk instead. You can hook asterisk in to PHP, Perl, Python, etc. You can use your scripts to create your own voice menus, and program your own functionality.

    It's fairly simple. I setup asterisk at my company, which is a fairly large health clinic. I wrote a script (executed by cron) that connects to our practice management software, pulls down a list of appointments for the next day, and makes an automated phone call to each patient reminding them of the appointment. It's fairly sophisticated: It detects answering machines and will wait until the "beep" before it leaves a message, tells the patient the date, time and the name of the clinician that they are seeing and asks them to "press 1 to confirm" or they can "press 2 to speak with a receptionist". It also manages the outbound bandwidth, it never has more than 5 calls going simultaneously. It will also try busy numbers and no answers 3 times, waiting 5 minutes between each try.

    After it finishes the calls, it e-mails the log of what happened to one of our receptionists who handles the answers, busy, disconnected numbers, etc. It also keeps a verbose record of exactly what happened throughout each call... time it dialed, when it was answered, if it left a message, what the user did, etc. Finally, it has a "do not call list" that the system won't call a patient if they've asked not to receive them.

    Overall, it's pretty limitless. If you have some API for your garage door, thermostat, etc. that you can interface with PHP, Perl, Python or a number of other languages, then you're good to go.

  • by GNUALMAFUERTE ( 697061 ) <almafuerte@g[ ] ['mai' in gap]> on Saturday October 23, 2010 @11:20PM (#34001210)

    Agreed on the network-dependent part. Bandwidth is usually not the problem, with high-compression codecs and good silence detection you can squeeze voip on plain old GPRS (I've done it using GSM and g729). Latency is always your biggest enemy. Remember, you are measuring your latency up to your server, from there, you have to add termination latency, which of course varies according to your destination. Calling overseas becomes an issue, because you'll be dealing with at least 200ms from 3G latency + another 150-250ms to the destination. I guess it all comes down to your phone usage (that is, what kind of destinations you call) and what coverage you have on your area. In my case, I mostly deal with companies (that tend to use VoIP on their side, that means increased latency), and the coverage in my area isn't great sometimes (Buenos Aires, Argentina), so I have to deal with > 500ms overall (that is, phone to phone, counting 3G latency, termination, endpoint latency, etc). Unusable for ~50% of my calls. I just said fuck it, and went back to using my cellphone carrier.

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