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Using BroadVoice with Asterisk How-To
Kerbo writes "With all the hype surrounding open source PBXs (telephone switches) such as Asterisk, the user community is clamoring for more help in getting these systems up and running. The Geek Gazette has published an article on how to configure Asterisk to work with BroadVoice VoIP service and eliminate the need for the phone company."
but how long until the phone company - who owns the backbone somewhere up the line, puts its foot down?
billy - remembering DSL
What Is Asterisk?
Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Check the Features section for a more complete list.
Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium(TM). Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks as well as a single port FXO card and a one to four-port modular FXS and FXO card.
http://www.sandstorming.com
For those who are interested in VoIP for business, I've written some online guides:
VoIP for business
How ISPs can sell VoIP services to their customers
You can set up your own free pseudo-micro-telco with Asterisk and a bunch of software phones. Asterisk, afaik, runs best on Unix, although I think I remember seeing a Windows version around recently. Software phones are available for many platforms.
Seeing as how featureful, small, internet-connected telephone switches might just attract some phr34ks, you may want to run your Asterisk on an older, dedicated machine.
I have been playing with it on a Debian machine recently, and it was really easy to install. The configuration has been non-trivial, but not too terrible.
I like the idea because you are in complete control of an entire PBX, including switching, extensions, and voice mail. You can even include on-hold music of your choice.
The best advice I can give is to install Asterisk (behind a firewall, to start) and begin playing with it.
Wil Langford - opinionated bastard - Linux rules
I'm working on a system where each of my computers (at home, work, and my parent's house) is fitted with a Bluetooth dongle which will discover if my mobile phone is nearby. If it is, calls to my Asterisk server will be routed to the landline phone I'm sitting at.
Geeky, I know, but I started my engineering life in telecoms and can't seem to shake it off...
The Slashdot story How Do You Make International Calls? drew 420 comments. The best suggestion by far, I found, was BroadVoice. It's amazing: $25/month for unlimited calls to land lines in 35 countries.
The BroadVoice service has been excellent. Note this comment earlier in this story: BV = Poor Support, #11989921. It's very important when using any VOIP service to test your internet connection quality. It's easy. Linux users need no help, probably. Windows users left-click on Start/ Run/. Enter CMD and press the Enter key. In the Command Line Interface (DOS) box that appears, enter
and press the Enter key. The times may be about 60 milliseconds, and should all be below about 300 milliseconds, and there should be no times far (5x) larger than the average time. Hold down the control key and press the C key to exit from Ping before the 100 tests are completed.
If you get highly variable Ping times, you will have trouble with VOIP, both in dialing and in talking. Call your ISP and tell them to repair their equipment. I did that with Telefonica here in Brazil, and, after hours of talking to many people, they did do the repair.
If you call your ISP, I suggest you don't complain about VOIP, because that is a painful issue for some ISPs. Instead, complain about these things:
1) Ping times definitely show there is a problem. Tell tech support to try it themselves.
2) Web pages give error messages or don't load unless they are clicked on more than once.
3) Email cannot be received or sent except by trying several times.
4) Music on internet radio is periodically interrupted.
BroadVoice customer service has been excellent for me.
"I'm the only one in the family running Linux ..."
Asterisk also runs on *BSD, MacOS X and Solaris. With the help of Cygwin it even runs on Windows now.
In fact, talking about an easy to set up home PBX, you might actually find MacOS X to be far more likely to suit your needs.
There is an Asterisk installer for the Mac, so you don't have to built it yourself and there are GUI based setup wizards, or assistants as they're called in the Mac world, which allow non-geeks without tech skills to set up a basic home PBX in just a few minutes.
A driver for using the Mac's built-in modem as a voice port to connect to a POTS line is on its way.
But even if you don't have a Mac nor want to buy one, I assume that similar tools will eventually show up for Windows now that Asterisk runs under Cygwin.
Asterisk on Linux will probably remain a "mostly for geeks" affair. Then again, there are some promising efforts under way to package Asterisk and Linux in a "works out of the box" fashion, for example Asterisk@Home.
Anyway, you shouldn't compare Asterisk with Skype because Asterisk is a _server_ application that can be linked to just about _any_ service and Skype is a _client_ application that is _locked_ to one single service.
the macintosh asterisk mailing list http://www.astm
I agree, very poor support, and generally poor service. Great price is the only plus.
The only codecs they allow are alaw and ulaw, which are bandwidth hogs, leading to poor call quality.
3/10'th the time they can't complete my outgoing call for one reason or another, sometimes I even get error messages in russian(?).
The worst part is their SIP implimentation changes regularly. One day everything works fine, the next day they've changed something and I have to fiddle with your configuration for a few hours to get it working again.
Good luck getting in touch with support, they don't answer e-mail and rarely answer the phone. I've called them a total of 27 times in the past, and got a rep on the phone only once. The rep was nice and tried to help at least. I am now trying to cancel my account to no avail. I'm putting a stop-payment on it tomorrow.
In short, don't use BroadVoice unless you are a bit of a masochist. There are other services such as VoicePulse Connect and SimpleTelecom that work MUCH better, and are fully supported.
Look for a provider that uses IAX instead of SIP. IAX is great for people behind nat, and the fact that they allow IAX connections means they run Asterisk too so compatibility is almost guarenteed.
"What I would like to know is how I may build a free (as in speech) Skype-like network with my friends, using Asterisk (or something else)?"
That's rather easy to do with Asterisk.
The first thing to do is - surprise - to set up an Asterisk server. Next, you configure a user account for yourself and one for each of your friends.
Then you tell your friends your server's address or DNS name, username and password and ask them to download a software phone that supports any of the open standards Asterisk supports, eg. SIP, IAX and H.323 to name the most important ones.
For Windows, your preferred choice would probably be the Firefly softphone, which supports both SIP and IAX, another one is called X-lite which is SIP only. For Linux there are quite a few open source softphones supporting various protocols, SJphone, Kphone, GnomeMeeting and more. For MacOS X there is X-Lite and the cross platform iaxComm (Win/Lin/Mac). All those are free.
Then all that remains to do is to tell those softphones how to find your Asterisk server and what their username and password is. In some cases a little fine tuning may be needed. For example, if someone is behind NAT, you may have to work around NAT traversal problems.
The easiest way to avoid NAT problems is to use the IAX protocol and a softphone that support IAX, eg. Firefly or iaxComm. IAX doesn't have NAT issues, so no work arounds are needed.
Note, that Asterisk supports multiple protocols concurrently. So, some of your friends might come in using SIP while others use IAX and yet others use H.323. The overhead for Asterisk to translate between protocols is negligible.
Everybody can now call everybody else by their username, which could be a nickname or an internal phone number. If a user isn't logged in, calls will go to voicemail. You can also set up chat rooms for multi-party voice conferencing.
In addition, you can set up so called SIP URIs, which is akin to an email address. In fact, your email address may well be identical to your SIP URI. Using that SIP URI, anybody with a SIP device can now call anyone on your DIY VOIP network, if you want to allow that.
Your friends can also register their ordinary phone numbers with a directory service like E164.org and if somebody with an appropriately configured IP-PBX calls that number, the call would not pass over the PSTN but over the internet via your Asterisk server to the owner of that number.
All this is not very difficult to do and you don't need a very powerful box either. So, all I can say is: Go for it!
the macintosh asterisk mailing list http://www.astm
The benifits as I see them:
Multiline. Several people can be calling at once. (Provided the voip provider doesn't mind. Unlimited plans usually forbid this, but per minute plans have no such restrictions.)
Multi-mailbox. You can assign a different mailbox to different members of the household.
Multi-number. You can have multiple phone numbers in different geographic areas. You can even get cheap 800 numbers that cost 2cents/min. All these can be funneled to the same phones.
Telemarketer avoidence. You can have a top-level voice menu that asks people to press 1 for person-a, 2 for person-b etc. If they don't press anything the call is dropped. The predictice dialers that telemarketers use won't press anything, so the call never rings any of your phones.
Per-callerid call-routing. Calls from people you'd rather not talk to can go direct to voice mail or get blocked. (jokingly refered to as the ex-girlfriend option in the asterisk documentation.)
Better voice quality on the voicemail. Most home answering machines compress the crap out of the incoming and outgoing messages. Computer disks are cheap enough and voice only takes 64kbits/sec uncompressed anyway, so you can just keep it in the native telco-format and not lose any voice quality on the messages.
call accounting. If you do consulting, you often want to keep track of how long you spent on the phone with each customer. Asterisk automatically logs every incoming and outgoing call with the exact call start and end times.