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Asterisk 1.8 Released With Support For Google Voice

timothy posted about 4 years ago | from the undersung-hero dept.

Communications 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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I love poast (1)

Jowr (226732) | about 4 years ago | (#33999502)


Small businesses that need to watch costs? (2, Insightful)

ickleberry (864871) | about 4 years ago | (#33999518)

I don't think any business can "spend like they just don't care" in these recessionary times. Except maybe Google themselves

Re:Small businesses that need to watch costs? (1)

niftydude (1745144) | about 4 years ago | (#33999580)

Additionally - asterisk is quite scalable- so I think large businesses would also benefit quite nicely from using this. My experience is that ever company I've ever come across spends too much on telephony.

Re:Small businesses that need to watch costs? (4, Interesting)

icebike (68054) | about 4 years ago | (#33999910)

Unfortunately, the small business is HIGHLY unlikely to have any of the skill set to get asterisk running, negotiate the Voip mine-field, and not lose calls and voice mail, and generate crippling telephone disruption for the entire business.

The benefits it offers just start to make it worth while at about 25 users or so, due to the falling prices of cat5 phone systems which you can pick up for pretty cheap these days, especially on the used market.

Asterisk is probably not really warranted until you have a hundred desks or so, and only then if you happen to have a fairly good geek on staff.

It is scalable, but not for the entry level technician unless all you want it to do is handle voice mail.

Re:Small businesses that need to watch costs? (4, Interesting)

Low Ranked Craig (1327799) | about 4 years ago | (#34000220)

There are a number of packaged implementations, like PBX in a Flash and Elsatix. If you have basic knowledge of TCP/IP networking you can work your way through it. There are lots of tutorials. I wrestled my way through it and found that it's not that hard unless you need copper lines. Just need to pay special attention to security so you din't end up with a $10,000 phone bill. We're getting ready to dump Vonage for straight VOIP by the minute. For what I'm paying Vonage every month I can have 3 numbers, each with 2 "lines" and 5 times the minutes. We went from PacBell, to Cox to Vonage and now to ala carte VOIP - our phone bill will have gone from $75 to $15 a month...

Re:Small businesses that need to watch costs? (1)

0100010001010011 (652467) | about 4 years ago | (#34000516)

How good is the programming language to setup and use? Back when GV was new I registered (xxx) xx-HOUSE. I have 1-wire setup throughout the house for stuff like the furnace. Garage door, etc.

I want to be able to dial into my Asterisk box and enter my pass code and see if the garage door is open (and optionally close it). Set thermostat temp, etc.

Re:Small businesses that need to watch costs? (3, Informative)

adamstew (909658) | about 4 years ago | (#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.

Re:Small businesses that need to watch costs? (0)

Anonymous Coward | about 4 years ago | (#34005256)

Sounds awesome! Can you share?

Re:Small businesses that need to watch costs? (1)

Just Some Guy (3352) | more than 3 years ago | (#34011618)

May I suggest that you run this setup through your legal department? That sounds utterly HIPAA-noncompliant: "Hello, this is the office of Dr. Obstetrician calling for Teenage Daughter to confirm her appointment for tomorrow at 9:00AM. Have a nice day!"

Chances are that's not a huge deal for you. For example, my wife's a podiatrist, and few people are going to be mortally embarrassed to have an appointment for her to see their ingrown toenail. Still, it's something you might want to think about. Maybe consider having a line on your patient questionnaire like "I agree to have Dr. Obstetrician's automated phone system call my house to remind me of my appointment: YES___ NO___" with a place for their initials.

Re:Small businesses that need to watch costs? (1)

adamstew (909658) | more than 3 years ago | (#34013022)

It has been run through the legal department. Whenever we get a phone number, we always ask if it's okay to leave messages at that number. It won't leave a message otherwise.

Also, we NEVER mention anyone's name or any personally identifiable information about someone. Just that "This is a reminder phone call for an appointment at with Dr. . To confirm this call..."

Re:Small businesses that need to watch costs? (1)

Just Some Guy (3352) | more than 3 years ago | (#34013270)

Gotcha. I think it's a great idea, but thought I'd mention that stuff just in case. I'm sure you're all too aware of how nitpicky HIPAA gets about stuff. For a while, the local gov't liaison person wanted my wife to have a "take the next number" machine in her lobby and call patients back by number. I'm not sure if they ever understood why my wife flatly refused to have any part of that idea.

Re:Small businesses that need to watch costs? (3, Interesting)

Albanach (527650) | about 4 years ago | (#34000272)

Crikey, I've run my house on asterisk with very little maintenance for the best part of a decade. I have multiple incoming numbers, least cost routing, Direct Inward System Access on a 1-800 number which I can use from hotels/airports etc, conference calling that gets used for family calls and work. An added bonus is the easy NAT traversal of the IAX protocol. It's easy to get a box up and running behind a domestic router.

For anyone looking for a really easy set-up, there's things like Trixbox.

Certainly it can seem daunting and there are pitfalls to beware of. However, small businesses are often spending a fortune on telephony that could be better placed, or they could be enjoying a feature set well beyond that currently available to them. As an example, we had a Nortel PBX with about 20 extensions and were looking for an extra two lines. Either we needed a card installed to provide the extra lines, or they could make a software change to enable VoIP and we could have added two SIP phones using our existing networking. The cost of either option was > $3,000.

Re:Small businesses that need to watch costs? (2, Insightful)

icebike (68054) | about 4 years ago | (#34000350)

But you sir, are a geek. /tips hat/.

And presumably no one paid for your time.

Re:Small businesses that need to watch costs? (1)

Albanach (527650) | about 4 years ago | (#34000642)

Yes, I'm a geek. My time was more than paid for in reduced call costs. A day or so configuring asterisk probably saved me about $1,500 in international call costs in the first year of ownership.

There were two points I was trying to make. My setup has been almost maintenance free, meaning a very small TCO. Secondly, there are GUIs available that really should make this within the competence level of many small business tech's.

Re:Small businesses that need to watch costs? (1)

wrook (134116) | about 4 years ago | (#34002416)

Yes, especially Trixbox is very easy to set up if you have a dedicated machine for it (and you don't need much for an Asterisk box). For an installation cost, Asterisk is much cheaper to install than your average hard line PBX in my opinion. Well, I guess Nortel is out of business now, but those Nortel PBXs were a bitch to set up and maintain (add new lines, etc). My only real complaint about Asterisk (as long as I don't look at some parts of the code ;-) ) has nothing to do with Asterisk itself. I find that I have problems getting consistently good quality sound through my ITSP. Sometimes the lowest cost route is really bad. Some ITSPs give you some control over the route so I need to play with that a bit, but it's a pain in the ass to deal with. If you have your own FXOs its less of a problem. Of course that increases the cost a bit (and buying good FXOs seems to be an art in and of itself).

Re:Small businesses that need to watch costs? (1)

wgoodman (1109297) | about 4 years ago | (#34003096)

You can buy a preconfigured Asterix setup including the computer it runs on for ~2k. Even for people without an inhouse geek, that still seems better than 3k.

Re:Small businesses that need to watch costs? (1)

Albanach (527650) | about 4 years ago | (#34006652)

That 3k was for TWO additional lines on a phone system that had already cost over $20k, or $1,000 per extension!

Re:Small businesses that need to watch costs? (0)

Anonymous Coward | about 4 years ago | (#34002214)

I'm a VoIP newb researching how to switch from Vonage to my own custom system based around FreeSwitch. What termination services do you use for your least cost routing setup?

Re:Small businesses that need to watch costs? (1)

Albanach (527650) | about 4 years ago | (#34006666)

I've used Gizmo5, Google Voice, Broadvoice and Junction Networks for US based calls.

There's an example of LCR using extensions.conf here: []

It demonstrates how to test for carrier availability and failover if necessary.

Re:Small businesses that need to watch costs? (1)

dskoll (99328) | about 4 years ago | (#34004346)

Totally untrue. We have 8 employees and run Asterisk. We hired a consultant to set it up, and then maintained it ourselves. It was much cheaper to set up and run than our former proprietary PBX from NEC.

I liked Asterisk so much I even run it at home. I have implemented countermeasures to avoid telemarketers and I love it.

Re:Small businesses that need to watch costs? (1)

aaarrrgggh (9205) | about 4 years ago | (#34006502)

As a /.'er, I can say that Asterisk was cost effective for us when we got over 6 extensions. I had no real knowledge of Asterisk or creating a dial plan, but we got it together and it has worked reliably for years.

Compared to paying Digium or one of the other providers the recurring fees, it is marginally cheaper... even with out-sourcing our linux administration. Compared to the proprietary systems that you need to have a telecom vendor re-program, Asterisk seems to pay for itself in less than a year with minimal specialized skills.

The only time the value proposition starts to disappear is when you want specialized features; the way we would like our night call switch to work would cost too much to implement, as would find-me/follow-me or a outbound dialing directory. 1.8 Might be the version we upgrade to though for improvements in some of these things. It's about time to replace the server anyway...

Re:Small businesses that need to watch costs? (3, Funny)

mrsteveman1 (1010381) | about 4 years ago | (#33999612)

in these recessionary times


Re:Small businesses that need to watch costs? (0)

Anonymous Coward | about 4 years ago | (#33999842)


Re:Small businesses that need to watch costs? (1)

Crypto Gnome (651401) | about 4 years ago | (#34002726)

Maybe not in America, but there are other countries out there, and their economies are not necessarily sucking as hard as yours is right now.

Re:Small businesses that need to watch costs? (1)

Eunuchswear (210685) | more than 3 years ago | (#34012402)

No, many of them are doing worse.

At least the US isn't (yet) being run by people who think the way to deal with a recession is to cut spending.

Vs Freeswitch (0)

Anonymous Coward | about 4 years ago | (#33999522)

Freeswitch had these and more for a while now.

Re:Vs Freeswitch (2, Insightful)

man_of_mr_e (217855) | about 4 years ago | (#33999882)

Yeah, I may be wrong, but I think Asterisk *Still* does not support SRTP after all these years.

This leads me to wonder why so many protocols are unencrypted on the internet. most IM's, SIP, H.323, etc.. it's just ridiculous how much information people are giving away across unencrypted connections.

Re:Vs Freeswitch (5, Informative)

kasparov (105041) | about 4 years ago | (#34000230)

You are wrong. Asterisk 1.8 supports SRTP. I know, because I merged it myself. :-p

Re:Vs Freeswitch (1)

man_of_mr_e (217855) | about 4 years ago | (#34001186)

As I said to the other guy, I'm glad to be wrong on this. I've been using FreeSwitch for years now because it support SRTP and Asterisk didn't, despite it's immaturity.

Re:Vs Freeswitch (0)

Anonymous Coward | about 4 years ago | (#34002640)

The information at is not correct then.

Re:Vs Freeswitch (1, Funny)

Anonymous Coward | about 4 years ago | (#34003676)

Wikipedia being out of date or incorrect? Inconceivable!

Re:Vs Freeswitch (0, Redundant)

IMightB (533307) | about 4 years ago | (#34004564)

That word... I do not think it means what you think it means.

Re:Vs Freeswitch (1)

Albanach (527650) | about 4 years ago | (#34000292)

Asterisk 1.8 includes contributions from hundreds of community developers, as well as the Digium development team. Asterisk 1.8 updates include:

  • Secure Real-time Transport Protocol (SRTP) support: End-to-end VoIP encryption of signaling and media
  • Security event framework: Modular capability for collecting and distributing system security events []

Re:Vs Freeswitch (1)

man_of_mr_e (217855) | about 4 years ago | (#34001136)

I'm glad I was wrong. Still, it's about time.. This is something that's been sorely needed for ages.

Re:Vs Freeswitch (0)

Anonymous Coward | about 4 years ago | (#34004468)

On behalf of actual Asterisk users to devs, I apologize for the asshole who posted above me. We don't think it's "about time" any feature was added. We appreciate all your hard work, and when we're wrong about something, we don't feel the need to add snarky comments like "about time" as if it somehow makes us sound less wrong.

Re:Vs Freeswitch (1)

TooMuchToDo (882796) | about 4 years ago | (#34005636)

Any hardware SIP endpoints that support SRTP? Or just softphones/software SIP clients?

Re:Vs Freeswitch (1)

man_of_mr_e (217855) | about 4 years ago | (#34006174)

Many do, although there are a number of compatibility issues.. many of them implement the standard poorly or incompletely. I have successfully used Polycom phones (IP-501 or greater.. the lower end models don't have enough horsepower), but Cisco and Aastra also support it.

Re:Vs Freeswitch (1)

TooMuchToDo (882796) | about 4 years ago | (#34006198)

Excellent. I have a Cisco 7960 I'm currently using with Asterisk/Google Voice/Twilio. I'll have to dig how to convert it to use SRTP after I upgrade my Asterisk instance, since my Asterisk VM is at my datacenter but I have phones at several physical locations (never know who is listening).

Re:Vs Freeswitch (0)

Anonymous Coward | about 4 years ago | (#34000242)

So has asterisk... release isnt covering all the real improvements...

Re:Vs Freeswitch (0)

Anonymous Coward | about 4 years ago | (#34006952)

I completely agree and I won't touch Asterisk with a ten-foot pole and won't use it for anything serious until they fix their issues.

For example, the way in which they handle their channels is laughable.

This has been in the Asterisk source code forever now, and I can't believe they haven't fixed it yet.

FreeSWITCH does these things at lot better. /*
  * XXX This is a seriously whacked out operation. We're essentially putting the guts of
  * the clone channel into the original channel. Start by killing off the original
  * channel's backend. I'm not sure we're going to keep this function, because
  * while the features are nice, the cost is very high in terms of pure nastiness. XXX

        \par Masquerading channels
        In some cases, a channel can masquerade itself into another
        channel. This happens frequently in call transfers, where
        a new channel takes over a channel that is already involved
        in a call. The new channel sneaks in and takes over the bridge
        and the old channel, now a zombie, is hung up.

Re:Vs Freeswitch (0)

Anonymous Coward | about 4 years ago | (#34008060)

FreeSWITCH sucks, they fork every possible lib and their lack of FHS can't make them into distros.

Can I do this with Asterisk? (3, Interesting)

nikomo (1338131) | about 4 years ago | (#33999582)

This is extremely off-topic, but this is a good place to ask this question. Could I host Asterisk on my VPS and then call other people through Asterisk with my Nokia phone that supports SIP (both through WiFi and 3G/3.5G). Basically, SIP-call to my friend's SIP-supporting phone who's connected to my Asterisk, from my SIP-supporting phone that's connected to the same Asterisk. Figure it would basically be free calling since we all have unlimited 3G plans.

Re:Can I do this with Asterisk? (3, Insightful)

sslayer (968948) | about 4 years ago | (#33999618)

Yes, you'll be able to do that. That is, probably, the most basic functionality of Asterisk. You can, as well, make your Asterisk server connect to another SIP provider (like voipbuster and the like) and redirect your phone calls through the trunk it provides. And that's just the beginning...

Re:Can I do this with Asterisk? (1)

lidocaineus (661282) | about 4 years ago | (#33999630)

Since Asterisk is a sip proxy/pbx and this is its main function, I would say yes.

Re:Can I do this with Asterisk? (2, Insightful)

Bert64 (520050) | about 4 years ago | (#33999690)

You can, this is exactly what asterisk is for..
Be careful about the ToS with your telco - they might explicitly forbid the use of voip over 3G, and might actually block it at a technical level. You might need to hide the traffic in a VPN, or one telco i'm aware of blocks SIP but doesn't block IAX.

You can do a _LOT_ more with asterisk than just that tho, for instance you could sign up with sipbroker which would let people in many countries around the world call you for the price of a local rate call, you could sign up with a selection of providers to make cheap outbound calls and you can setup fancy menu systems and have hold music playing like large companies have been doing for years, you can even put telemarketers into a queueing system or have scripts to annoy unwanted callers based on callerid etc.

Re:Can I do this with Asterisk? (1)

nikomo (1338131) | about 4 years ago | (#33999906)

I don't think that'll be a problem, I live in Finland and we actually still have rights here as far as telecommunications goes. Nothing is filtered except for the website blocklist that the police forces ISPs to block childporn on a DNS level. I have yet to see anything get blocked through my 3G, I've downloaded lots of Linux LiveCDs through P2P networks through 3G.

Re:Can I do this with Asterisk? (0, Redundant)

hweimer (709734) | about 4 years ago | (#33999772)

As others have pointed out already you can surely do this with Asterisk, but I think in your case the question is whether your 3G network operator allows SIP traffic on their unlimited data plans.

Re:Can I do this with Asterisk? (1)

GNUALMAFUERTE (697061) | about 4 years ago | (#33999828)

Except that 3G tends to have a very shitty latency.

I've worked on VoIP for almost 10 years now. I've tried to pass voice through mobile networks since the times of CDMA 1X. It just doesn't work very well. 3G is a huge improvement, and it all depends on what network you are on, and your current signal, but don't expect it to work like a regular call.

Regarding asterisk on a VPS, that shouldn't be a problem.

Re:Can I do this with Asterisk? (2, Informative)

wolrahnaes (632574) | about 4 years ago | (#34000482)

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).

Re:Can I do this with Asterisk? (2, Informative)

GNUALMAFUERTE (697061) | about 4 years ago | (#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.

Re:Can I do this with Asterisk? (1)

EdIII (1114411) | about 4 years ago | (#33999870)

You might be able to go one step better...

SIP traffic usually sucks balls on cell phones. I just don't see enough quality in the 3G connection to support it in most cases. Making a voice call is usually much higher quality.

If you have friends and family plans on your phones you could add a PSTN number that is connected to your Asterisk box. Setup a reverse IVR.. and you now have a working gateway that allows you call anywhere in the world you want at the prices of a decent VOIP provider. Think of like a calling card and you are the only customer. Lock it down with a PIN, and by caller-id if you want, and it will be fairly safe In fact, calling card programs that exist for smart phones would allow you to route all your outbound calls through your Asterisk box without you even having to call the number, enter the PIN, and then enter the transfer number. The software does that all for you and makes it very easy for you to dial.

There a lot of VOIP providers that will give you unlimited inbound calling too. My provider does this but charges .9c per minute for all outbound calls within the US.

So what I end up with is a system where all of my calls, inbound and outbound, are reduced to .9c per minute on my cell phone. Granted it is not completely free like your idea, but I am not dealing with VOIP across my phone. I get the full quality of service for a regular phone call at that extremely reduced price. Compare that to most cell calls which are upwards of 25c per minute.

This does not address calls to foreign countries though... but if you really want to take it even further to the next level you could setup Asterisk boxes on hosted or dedicated servers in the EU, Australia, etc. Purchase a DID (the PSTN) number and create the same gateway system in that country at their local VOIP prices. Connect the Asterisk box in the US with the Asterisk box in the EU via an IAX2 trunk and you now have country to country traffic being routed at local prices. It's as a simple as your buddy in Australia calling his local gateway, entering a specific 4 digit extension and being routed to your cell phone in the US. All of that at the local outbound VOIP costs for both gateways which has to *loads* cheaper than an international cell phone call.

Asterisk and VOIP is awesome. If you have multiple corporate branch offices spread out across multiple companies there is NO REASON to pay international calling costs if you have a little skill with Asterisk :)

Re:Can I do this with Asterisk? (1)

numbski (515011) | about 4 years ago | (#34004312)

Granted, I know we're talking internationally, but 25cents for a minute of cell phone airtime???

T-Mobile here, I pay $125/mo for 2 phones, "unlimited" data, sms, and mms. 1500 shared minutes. Reasonably speaking, $25 of that is data, fees, and taxes. 1500/100 is 15 cents per minute, but more than that, we get unlimited mobile-to-mobile calling, so we get much more than that.

Re:Can I do this with Asterisk? (1)

arndawg (1468629) | about 4 years ago | (#34000004)

Timing in a virtual machine could be an issue. I do run asterisk at my job on Xenserver and it works pretty good. But i'm hearing that vmware might have issues.

Re:Can I do this with Asterisk? (1)

nikomo (1338131) | about 4 years ago | (#34000928)

Do you happen to know about Virtuozzo containers? The VPS I rent from Germany runs on OpenVZ.

Re:Can I do this with Asterisk? (1)

moonbender (547943) | about 4 years ago | (#34001032)

If that is all you want, you might as well get an account with any of the SIP providers out there. With the one I used, calls are usually free as long as they don't touch the "real" telephone system. So calling any of your friends who's also using SIP is free. With the one I used (sipgate), I actually got a POTS number for free, as well, so people could call me with their regular phones; often for free since they have free calling for "landline" numbers. I've tried this using two Nokia phones over Wifi, never bothered to try it over 3G. Staying logged in indefinitely was a huge battery drain, killing most of the appeal.

Why? (1)

Slashdot Parent (995749) | more than 3 years ago | (#34026770)

Two reactions to your plan:

  1. Yes, you can definitely do it, but no, you will probably not be happy with the quality. As of the last time I looked at Asterisk, a finer-grained real time clock was required for optimal performance. That could either be via the Digium hardware (which you wouldn't have on a VPS) or a RTC/ztdummy/?? module compiled into the kernel, which you most likely don't control.
  2. Congratulations. You just reinvented fring [] .

Dahdi support? (1)

JamesTRexx (675890) | about 4 years ago | (#33999754)

I hope DAHDI is better supported this time. I had a lot of trouble getting it to work under a plain (Ubuntu server) installation as it involved modifying settings that weren't really documented. :-/
I reverted to using the AsteriskNOW installation and even that needed a manual change to one of the config files for it to recognise two FXS channels...

Re:Dahdi support? (1)

gratuitous_arp (1650741) | about 4 years ago | (#34000838)

I've been through DAHDI and Zaptel-hell many times in 1.4, 1.6, and now 1.8. Playing around with the 1.8 code (beta has been out since mid summer), DAHDI/dahdidummy is working just fine on my Ubuntu 8.10 box for conferencing. Coaxing Zaptel into behaving was hell the first time around, this isn't bad at all. So be happy. :-)

Don't use Google Voice for anything critical (3, Informative)

Animats (122034) | about 4 years ago | (#33999818)

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.

Re:Don't use Google Voice for anything critical (2, Interesting)

Anonymous Coward | about 4 years ago | (#33999868)

These issues you name actually happen with all companies.

Re:Don't use Google Voice for anything critical (1, Insightful)

Anonymous Coward | about 4 years ago | (#34000490)

Really? Not any telco ive ever dealt with.

Re:Don't use Google Voice for anything critical (-1, Flamebait)

Anonymous Coward | about 4 years ago | (#34001984)

Really? Not any telco ive ever dealt with.

You're speaking from experience with lots of corporate PBX deployments I see. NOT!

Unable to call phone blocks (1)

phorm (591458) | more than 3 years ago | (#34012320)

I've definitely run across it once or twice. The first time was with Bell phone cards, which for some reason had issues dialing the "778" prefix (newer prefix in BC, Canada). It really sucked when I'd bought a few cards to keep in touch with my GF while out-of-town, only to find the damn cards didn't recognize her number.

The second time was with 1-8xx routing issues from the local Fido (Rogers) network. NO 800-type number would work, they'd always come up as busy. Other networks worked fine. This one cropped up twice in about 14 months, but I think they may have fixed it now.

Re:Don't use Google Voice for anything critical (1, Interesting)

Anonymous Coward | about 4 years ago | (#34000528)

I've been testing Google integration in Freeswitch and there's WAY too much latency to use this for anything other than a novelty.

What can a home user do with askerisk? (1)

frovingslosh (582462) | about 4 years ago | (#33999866)

I've long heard of asterisk and I've visited the website (even downloaded it at one point) but what I fail to grock is what can an individual actually do with it? Maybe the Google link in will help, but I call Google voice with my land line, not voip, and Google chat's communication doesn't give me an inbound number. If I can use the technology for something interesting at home I would set up an asterisk system, but I just can't see what I can do with it.

Re:What can a home user do with askerisk? (2, Interesting)

Anonymous Coward | about 4 years ago | (#33999944)

You just need a SIP provider like Gizmo5. They are now owned by Google. This lets me make and receive unlimited free calls to anywhere in the US from any device that has a SIP client like my PC or my iPhone. As long as data is unlimited then calling is unlimited since its all VoIP on my end. Google Voice provides the link to PSTN.

If I call out, Google voice calls my Gizmo5 SIP number and my other party. If someone calls my Google Voice number, Google calls my SIP and connects me with the caller.

Re:What can a home user do with askerisk? (1)

nikomo (1338131) | about 4 years ago | (#33999948)

Free SIP calling between friends? That's basically what my thread was about.

Re:What can a home user do with askerisk? (3, Informative)

sampas (256178) | about 4 years ago | (#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:What can a home user do with askerisk? (1)

complete loony (663508) | about 4 years ago | (#34000166)

$200 for FXS and FXO? I can get an ADSL modem with both for less than that.

Re:What can a home user do with askerisk? (1, Interesting)

Anonymous Coward | about 4 years ago | (#34000432)

"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."

A Linksys 3102 has both 1 fxo and 1 fsx and costs about $60. A Linksys PAP2T has 2 fxs ports and costs about $50 (only downside is the lack of t38 (but that shouldn't matter much for home use)).

Re:What can a home user do with askerisk? (1)

arndawg (1468629) | about 4 years ago | (#34000134)

You can route between your land-line and SIP or IAX enabling you to do all kinds of fun things. You could use a SIP client via the internet to be able to answer you landline when you're on vacation .
You could use bluetooth to detect proximity to your server and then auto-switch your landline to forward to your mobile, or just go straight to voicemail which in turns sends an email with the voice-message.
You could set up a callback service so that you can call your land-line and press a special key and asterisk can give you a dial-tone so you could call for "free" from your mobile
Set up queues so callers can wait in line. +++++++

Re:What can a home user do with askerisk? (2, Informative)

kasparov (105041) | about 4 years ago | (#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.

Re:What can a home user do with askerisk? (4, Informative)

wrook (134116) | about 4 years ago | (#34000442)

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.

Re:What can a home user do with askerisk? (1)

Eunuchswear (210685) | more than 3 years ago | (#34012614)

Calling long distance to/from Japan is expensive, no matter what plan you have.

Bizzare. It's free for me.

Re:What can a home user do with askerisk? (1)

dskoll (99328) | about 4 years ago | (#34004752)

I have a POTS line, but I still use Asterisk plus some home-brewed programming to implement some nifty features:

  • Telemarketer Deterrence: Any call coming from a non-local area code (except for a few hand-whitelisted numbers) goes to a recording asking the caller to press "1" to prove he/she is not a telemarketer. This completely kills automatic diallers. Based on my logs, I see that it stops between 3 and 7 calls per week.
  • Number Recall: All my call records are logged to a PostgreSQL database. If I forget someone's number, I dial a special extension and a little AGI app I wrote prompts me to enter as much of the person's name as I can remember. It then rummages through the call log looking for matches and reads them out to me using the Festival voice synthesizer.
  • Blocking of 1-900 numbers. That's a no-brainer; I don't want people in my house to be able to call a 1-900 number.
  • Automatic prefix for long-distance calls. I don't have to remember to dial the magic prefix to get cheaper long distance; Asterisk remembers for me.
  • 7-digit dialling. Our area code now requires 10-digit dialling. I hate it. So if I dial a 7-digit number, Asterisk automatically prepends our area code.

These are small features, but I find them really handy. Our work Asterisk setup is a lot more sophisticated [] (PDF slides)

cheap jerseys (0, Offtopic)

nfl jersey online (1901968) | about 4 years ago | (#34001034)

Many football fans want to wear a jersey of their favorite team or player, but it can be difficult to shop for right size one. Typically, our ootball jerseys [] fits different than a normal website, so your regular shirt size might not necessarily be the same size as an NFL jersey [] . Using a measuring tape, it is easy to figure out what size red wings jerseys [] [] will fit you best. []

Skype for the Mom and Pop shops (1)

flyingfsck (986395) | about 4 years ago | (#34001224)

I am finding more and more that ordinary folks use Skype for their business and private calls. It works fine for a one or two person Mom and Pop Shop. So ordinary folks are getting very open towards VoIP in general.

Press Release (0)

Anonymous Coward | about 4 years ago | (#34001314)

Sometimes these stories just come off as press releases.

Re:Press Release (1)

Steauengeglase (512315) | about 4 years ago | (#34002082)

If it were any other company I would agree, but Digium have done a lot and deserve any good karma that anyone wants to send their way. They seem to genuinely "get" FOSS and haven't turned into a MBA driven POS itching for the day they can cash in.

They have always, honestly answered any questions that I've had and when I reached the point where it looked like Asterisk wasn't the solution I needed, they pointed me to the right direction instead of trying to sell me vaporware and a service contract. They may not have gotten me as a customer, but they earned my respect and I'll happily point others towards their offerings.

For something more polished, try SipX (1)

hmbJeff (591813) | about 4 years ago | (#34003952)

I looked at various Asterisk-based distributions for the residential phone system I run in my condo complex, and I found them all to feel pretty messy and hacked together. I kept looking and found SipXecs (, which is an amazingly polished open source project used in many companies.

It has a completely-integrated web management interface that controls all aspects of system config and operation. It is highly scalable, allows for clustering, comes pre-integrated to support SIP trunks from many carriers and phone devices from many vendors. It includes voicemail, IM, ACD (call center functionality), all completely managed through the web interface, and provides a user web portal as well for allowing access to these features plus call routing rules, phone directories, etc.

Architecturally SipX is cleaner, I think, than Asterisk, and is fully SIP-based. Calls between phones on a SipX switch pass their audio directly to each other, rather than passing through the switch, as is the case in Asterisk. Integration for SIP trunks is built-in. However if you need to use physical phone lines, SipX connects through external SIP gateway hardware from many vendors, rather than depending on the messy integration of cards into the server itself.

I now run phone service for myself and 25 of my neighbors for less than half of what we were paying, with way better features. Many SipX installations scale easily to hundreds of users.

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