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Asterisk Open Source PBX 1.0 Release
An anonymous reader writes "Today at Astricon (the first Asterisk conference), Mark Spencer announced the release of version 1.0.0 of Asterisk. For those of you that don't know: 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 users can be up and running, making phone calls in under an hour using the various guides found at the VoIP Wiki. Connectivity to the PSTN is provided by companies like VoicePulse, Nufone, Gafachi and VoipJet."
...right here.
It's a bit out of date - it suggests you check out the code from CVS - but lots of good info there nonetheless.
The Army reading list
google is your friend.... in short its a phone system.
AC, meet my friend Google.
(PBX = Private Branch Exchange)
http://www.webopedia.com/TERM/P/PBX.html
Free GMail invite with Free iPods!
http://en.wikipedia.org/wiki/Private_branch_excha
Keep your eyes to the sky.
...is quite nice and easy to get going with a cheap $40 FXO card. With that and a decently powered machine you can easily replace your home answering service or machine with something a bit more complex. As great as Asterisk is though it definitely is a 1.0 product, hopefully now that the functionality has stabilized somewhat, more work will be put into rearing the myriad of control files into something more managable and some work will be put into better troubleshooting tools. Odd or weird problems can be a real PITA to diagnose on your Asterisk setup.
My dialplan (which works all but the analog portion 100% of the time) is that a call comes in -> rings the analog line a few times -> asterisk then picks up and gives the user a menu, from there one can pick my sip client or my girlfriend's or a global that rings the analog line and the sip clients at the same time. In case of no answer voicemail then picks up and fires off an email to us containing the message. Eventually I hope to have it sharing functionality with some friends in different states so we can all have free local dial-ins for family and friends who are scattered.
--- I do not moderate.
Mirrors can be found at http://asterisk.paperwork.com
By far my fav are Cisco 7960 (I haven't tried the 7970) the only problem is they need things like DNS entried and tftpservers to work optimally.
For lower-cost alternatives, I really like the SNOM phones. I've used an snom 200 for quite a while and it's a very nice phone.
I also have a Pulver WiSIP which is nice but not exactly featureful, and the audio quality goes down when WEP is used.
For ATA's the SIPura, and the Linksys models there of ($50 or so) are a good bet, and the dirt-cheap Grandstreams work okay too.
Use the voip wiki to find optimal phone and sip.conf configs for a bunch of different phones.
Please send all UCE to scally@devolution.com so I can f
In my apartment, I've got a Cisco 7960 and Budgetone 100 both connected to * via SIP. They're at opposite ends of the cost spectrum, the 7960 being about $400 MSRP and the BT about $100. Both work fine.
See http://www.voip-info.org/ for more.
--
Phil
It depends on your needs. There have been suggestions that some CLECs are using Asterisk internally, and there are certainly a ton of VoIP startups using it. The general impression that I get is that you don't want to run more then 100 simultaneous connections through a single Asterisk server. If you want more, then add more servers and share the load. If you're doing a lot of compression on the server, the number may drop below 100.
Fortunately, Asterisk does a decent job of sharing information between multiple servers, but setting up a large multi-system PBX still isn't going to be trivial.
If you're using VoIP phones (Cisco, Polycom, etc), then there's no real limit to how many employees you can service with a single server. If you're using analog phones, then you should probably limit yourself to around 4 T1s worth of phones per server.
Ignoring the free iPod issue, this is free software (GPL, even) that we're talking about. It's not v1.0 of some random commercial program. It's v1.0 of the premier Linux VoIP package. That makes it news.
I've just barely started playing with it, but it's pretty easy to use once you get the hang of it. It even comes with prerecorded messages such as "all members of our household are currently dealing with telemarketers", "somethings *terribly* wrong", and one that's just angry monkeys screaming for 20 seconds.
Here are some great resources for getting started:
http://www.digium.com/handbook-draft.pdf
and a good soft phone (x-lite) at http://www.xten.com/
The Polycom IP300/IP500/IP600 line seems to be the best combination of price and performance right now, at least for a business environment. You can get cheap phones (the Grandstream Budgetone is around $70), but they're cheap and missing some features.
Asterisk doesn't have native LDAP support, but it's not very hard to write a script that produces a set of Asterisk config files out of LDAP data. With a bit more work, you could script Asterisk to do LDAP lookups, but it'll take too much work to be worth it for small (100 users) sites.
Don't forget asterlink.com and tollfreeexpress.com
Well, Asterisk already lets you send voicemail via email, with the message attached as a WAV file. It can suck its VM config out of MySQL or Postgres, or it can use text files. It'll also send mail to a pager email address; I get a SMS message on my cell phone whenever I get new voicemail at home. The message includes the caller ID information as well, which makes it a snap to return calls.
There's a patch out there somewhere to tie Asterisk into Request Tracker. Done properly, you could build a really interesting support phone system--it'd record calls, stick them into the ticket queue as needed, and give you a great way to keep track of who's bugging you the most.
Voop offers PSTN termination over IAX and SIP for Asterisk users in Norway. Both business users and private individuals welcome.
Disclosure: I work for Voop.
And remember kids: Never trust a computer you can actually lift.
There are un-locked linksys Sipura's out there, look for -NA on the model # PAP2-NA and RT31P2-NA are the two models available according to VoIP Wiki
Please send all UCE to scally@devolution.com so I can f
We use Asterisk where I work -- about 30 Zultys ZIP 4X4 phones connected to a dual Xeon server with a Digium ISDN adaptor card (4 x E1 spans). One span is used to connect to the outside world (the full 30 lines; was just 12) and another connects to our "old" Siemens HiPath exchange.
We did have a problem with call quality which seemed to be related to recording calls; it turned out that it was due to having far too many files in the recording directory, and once we had that sorted, it was clear as a bell again.
My boss has even set up an Asterisk server at home. I haven't, but I've a spare machine I might use for the job if I can scrounge a spare IP phone. I'm not using a softphone -- we tested every one we could get the source for and one we couldn't, and they were all lousy for one reason or another.
Je fume. Tu fumes. Nous fûmes!
I have seen a couple, bought none:u cts_id=48 9 34028032.htm?search=asterisk
http://voipstore.pulver.com/product_info.php?prod
and
http://voipstore.atacomm.com/shops/Search.aspx/27
also some much worse deals on ebay.
Only odd thing we had to correct was switching off the Linux screensaver, as it was causing voice quality to occasionally stutter under high network traffic volume.
Support Hint: an office PBX is a mission-critical system for a commercial business. You can't run it on an old piece of leftover trash! You need to put it on a high quality 1U server racked in your air conditioned computer room behind a secure door where the night cleaner can't plug his vacuun cleaner into your power bar!
You also have to ensure it's properly backed up to off site tape/CD-ROM storage, and that the disk is RAID so that it can be QUICKLY restored when the disk fails.
Anything less than this level of proper support means your ass is grass when something bad happens and the office comes to a screeching halt!
You have been warned.
Ocelot Wreak
"I figure you're here 'cause you need some whacko who's willing to stick his finger in the fan. So who are we helping?
Ok so I install this thing on a Linux server. Then what? How do I make calls to say, someone in New York from LA? And who would I have to pay still? No one?
You set up Asterisk servers in NY and LA and make them communicate (hint: IAX2 rocks). Now you pay for net connectivity for both servers and that's about it. If you want a normal phone number attached to your Asterisk server, you need to sign up with a VOIP provider (there are plenty of them in the US, I wish there was at least one here...) - check the Asterisk Wiki (link in the summary), there's a list somewhere.
Also, how would I interface my phone with this thing? Would I need to get a VOIP phone?
You can use:
- a VOIP phone (just about any SIP/H.323/MGCP phone you fancy although some don't work with Asterisk) for about $70-$500 (from Grandstream to Cisco)
- an ATA (analog telephone adapter, IIRC) which costs a little below $100 per port (check out IAXy and the Sipura gear) - it's a device to plug in your analog phone(s) which then lets it communicate via VOIP
- a TDM400P card by Digium with an FXS module (1-4 on a single card) - you plug in an analog phone and it works with Asterisk
- a softphone (X-Lite for example) but it feels somewhat weird
If you want PSTN (public phone network) connectivity, you need either a VOIP provider account or a FXO interface card (check out X100P and TDM400P with FXO modules on Digium site).For a home installation I think I'd recommend a Sipura SPA-3000
Pavlov. Does this name ring a bell?
My hosting business, Binhost Technologies, uses Asterisk behind its IAX/SIP Origination and Termination and wholesale VOIP operations. It works well -- the price is right and the features are many. Most phones work if they speak SIP, IAX, or H.323 and the system comes configured from the start in a pretty usable state.
A few things we've found out: The scripting system is a bit of a learning curve. Also, the configurations are one of those Jenga configs -- breathe too hard and it falls down. You have to be really, really careful when messing with the configs because one place can easily mess up another thing. But once you get a good, working config, it just works.
Processor usage is reasonable, too. A P-266 would do well for a couple of lines and maybe up to 10. After that you'll want a bit more horse.
</plug>
Blog,Twitter
The Offical Asterisk IRC channel!
irc.freenode.net
#Asterisk
Note: you must be registered and identified with NickServ to join the channel as we've had a lot of problems with spambots.
To do so simply
then
Come on in and say hi!
Some links
The Wiki
The Asterisk Documentation Project
Andy's Getting Started With Asterisk Guide (it's written for a old version of asterisk, but still useful)
ManxPower's site
For some advanced examples see John Todd's site
Also read all files in
more links (look at the "Unnoficial Links")
Mod me up!
.sig
I have one of those TDM400 cards populated with 3 FXS modules. Each module is wired into my house's pre-existing phone wiring and drives anywhere from 2 to 4 phones. There is no issue with DSPs running out of steam; you can still only have one conversation on the party line that makes up that set of phones. It turns out the limitation is more the amount of loop current you can push through the line drivers before they fry.
Been running this config for months with great success.
Now if only I could teach my wife and daughters the concept of extensions and the trick of transferring a call, I'd be one happy camper indeed
PaulW, IT Consultant