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Build Your Own PBX
Kerbo writes "Kerry Garrison has written up a complete guide to building your own PBX with Asterisk@Home to create your own working PBX system. In the article, he shows how you can build a complete, working system for under $20 (assuming you have some old hardware laying around the house)."
For those with extra hardware to run Astrerix@Home, consider running Folding@Home! ;-)
Hulk SMASH Celiac Disease
While I ddid focus on using softphones, you can use either SIP phones or normal phones using an ATA adapter. SIP phones range in price from $69 - $500.
If you don't know what a PBX is, you're not a geek.
If you want to avoid the hardware costs of a PBX, consider a virtual PBX system, which often offers many more features than you can get even in a professional PBX setup. For as low as $9.99 a month you can get a system at http://www.freedomvoice.com
I admit, I do work for this company, but it definately is a relevant option.
For those that don't know...Private Branch Exchange[wikipedia.org]
Thank you. I was going to post, too, but given my already "Bad" Karma, I thought better of it. Glad to see someone agrees.
In any case, here is an explanation of PBX -- it certainly helped me.
PBX = Private Branch Exchange
(Your work phone system)
Just in case it's not clear yet
http://www.webopedia.com/TERM/P/PBX.html
Why worry? Each of us is wearing an unlicensed "nucular" accelerator on his back.
Sig changed for readability by G.W.
In a PURE VOIP scenerio, you could get PSTN/POTS termination from a company like Vonage, Broadvoice, VoicePulse, or a number or other companies. This basically gives you Dialtone-over-internet. What you need on your side is either a single softphone, ATA adapter, or something like an Asterisk box communicate with the provider. VOIP is suitable for everything from 1 user to any number of business class users.
The parent submission is a serious troll. I mean, I started working the lowest level helpdesk in 97, and when I got my first tour of the server room, I was shown the PBX. Now, I had no idea what PBX stood for, or how it worked (and to be honest, I still don't), but I was told that its primary purpose was to handle all the phone switching.
I mean, come on. It's almost all but impossible to work in the IT field and not at least see a PBX.
Slashdot isn't supposed to spell out every single technology that it features. It's assuming that you either have an idea what the article is about, or you know how to at least use google. Go google for "what is a PBX". There, done.
Way off. There's no need to have POTS lines if you want to be a telecom, you need end termination.
Check out http://voip-info.org/ and good luck! =) It's a fun, crazy learning curve.
You can use any of the following: Regular phone lines (multiple modems or multi-line modems) ISDN Trunk lines T1 Trunk lines Internet-based carriers If you have 2 regular phone lines, you can handle 2 concurrent calls, a T1 can handle up to 23. It is a very effective cost point considering the alternative standard PBX costs.
Make sure your connection is up for it:
http://testyourvoip.com/
Wikipedia VoIP Article:
http://en.wikipedia.org/wiki/Voip
You can have 2 FXO cards (Wildcards ) in one computer, though sometimes it gives people problems. The *best* solution to what you want is a full fledged TDM400P with 2 FXO and 1 FXS modules. But thats expensive.
You can get 2 Clone FXO cards for cheap, and then use a Sipura 1000 or 2000 for your FXS. I use a similar setup (1 Clone FXO, Sipura, some Grandstreams), which works well.
--- Kicking the Cheat since late 2002
You either use something like the Wildcard X100P Generic (mentioned in the article as available on eBay for $6.95) to connect your standard analog line to the Linux box.
Or, you use a VoIP provider that does DID/Termination and will port your number. I believe Broadvoice does this. Then it's purely a VoIP line -- you use it by setting up the Asterisk box to use SIP to connect to Broadvoice. i.e. once you port into Broadvoice you'll no longer have a real landline.
Broadvoice is $5.95/mth for DID (unlimited incoming, pay per minute outgoing) or starts at $19.95 for their unlimited package (includes North America and quite a few European locations).
I have two Broadvoice DID's, one Broadvoice line that has the unlimited World that I use for outgoing calls, 1 Nu-Fone IAX outgoing line, 2 Unlimitel DID's (one of the only places to get Canadian numbers), 1 POTS line.
One of my favorite side tricks after getting the basic PBX features running is to setup one of the DID lines to give me DISA (direct-inward-system-access) when I call from my Cell Phone (based on Caller ID) so I can then dial out one of the outgoing accounts to anywhere in the world and only have to pay for a local call on my cell phone. (Here in Canada long distance is rarely included on cell phones like it is in the states)
Don't get into the things you were talking about. You only go there if you're running a business and want things like 24 flexpath lines on a T1.
That's because the articles haven't been very good. I know that I go through phases of interest and when the articles are as bad as they have been slashdot gets replaced with other activities.
if you currently have a single, POTS line from your local telco then you'll need one of them modem cards. And that's it. Plug the phone into the card, tell Asterisx about it, and you're done - you have a 1 line PBX. Good for call screening and voicemail.
:)
You don't need to tell your telco anything - this isn't the same as getting a block of DID numbers and a T1.
-EvilMagnus
FYI: T1 can handle calls on all 24 channels without extra digital info, Caller ID, ANI/DNIS, etc.
You are referring to a PRI (Primary Rate ISDN) that uses 23 channels for voice and 1 for call setup/teardown. This provides all modern phone convieniences.
The answer to your question is in the Step by Step Home Wiring Setup" in the Vonage Faq.
What I would like to know is if this Astrix PBX they talked about in the article can be used to replace the ATA Vonage hands out. I'm pretty sure Vonage won't let me return the ATA for any value, but Astrix looks to have more features.
What the parent said should make you happy. It works well.
.5-1 ring delay).
Please do not try and combine Vonage with Asterisk unless you're prepared to use an FXO device/card. It can't be done. (Before I get flamed, technically you can do it through software but Vonage won't give you the information to do it.) You'd have to use an FXO card to connect the "analog" line coming out of your Vonage ATA to the Asterisk Box -- it would treat it just like a standard POTS line.
A very crude setup, if I do say so. Your calls would be going through multiple conversions and using much more bandwith (Vonage -> ATA, Asterisk Box -> SIP-phone/ATA vs. Broadvoice -> Asterisk then bridged to Broadvoice -> SIP-phone after call setup).
Plus it's frankly just going to be annoying -- when you start doing things like this you're just asking for a long delay between when the person calls you and your phone rings (even with immediate=yes, the Wildcard X100P often introduces
Not a commercial, but you can add regular telephones to this great PBX system by going to www.voipsupply.com Most consumer SIP equipment is locked for Vonage, CallVantage, or Net2Phone. This shop sells non-configured versions which you can simply plug into your network, configure, and go. There are single- and dual-port analog adapters with FSX support, and fairly nice (and cheap) desktop phone sets, all SIP compatible.
Kriston
You can just use a regular phone line or a voip line if you prefer. No need to talk to the phone company. Busy signals for incoming calls are the phone companies problem, you don't have to worry about that. In a simple case you might have your Asterisk box pick up on the second ring (caller ID comes in between ring 1 & 2) and ask the caller to "press 1 for Jack, 2 for Jill etc. When the caller makes a selection you then ring the phone distinctivly (ie. different kind of ring for different people) and if noone answers the call goes to voice mail.
Those Lin/Winmodems are simply a soundcard with a telephone interface. They are of no value to most of us that have had fast Internet connections for years. Wait a minute: They have been approved to be connected to the PSTN! That is by far the greatest expence of producing this hardware.
Might I add, with a bit of experience, and perhaps a 'trade school' education in electronics it is trivial to reverse them and make 'ATA' devices. (Actually I think they are called 'FXS' devices in telco lingo. The devices that must be approved for connection to the PSTN are 'FXO' modules.) At somewhere between zero and a couple Euros per linmodem, used, this is a great idea.
As for the software, I'll stick to FreeBSD or Gentoo Linux and install my own Asterisk. Binaries give me the creaps. Computers a couple or three years old are likely to have more PCI slots and maybe an onboard 'Lin/Winmodem and lan card. While ee100s are a very good NIC (get them for as little as EUR 5,--) Linux and the BSDs have drivers for just about all types -- far more than Windows ever had.
Why do you need two FXOs? The outgoing to packet8 would be via sip uplink. It would be INCREDIBLY silly to go FXO-> DTA 310 -> Ethernet again!!!
+ and+A sterisk
Here is how to do it:
http://www.voip-info.org/wiki-Packet8+DTA310
So now you're down to 1 FXO and 1 FXS. However, you can, if you choose, "downgrade" your now unused DTA-310's firmware to an older version that lets you set which server to log into. Set it to log into your Asterisk server, and suddenly you don't need an FXS because your DTA-310 does the job!
Email me if you want details. it's my slashdot username @ happychinchilla . com
And even saying PRI uses 23 channels for voice and 1 for call setup/teardown is specific to a single PRI T1.
The spec for PRI allows that one channel for call setup/teardown to handle as many as 20 PRI spans. After the first PRI (which is shorted the control channel) each of the remaining 19 PRI T1 circuits can use all 24 channels for voice or whatever service you configure them for with your provider.
In other words if you can afford the expenditure, you can do variable bandwidth connections across 1 x 23 x 64 + 19 x 24 x 64 ~= 30Mbps with variation down to 1 channel or 0 if you and your remote end have some other method of signalling to activate from 0 channels (i.e. your isp sends you a reverse call request via the data channel.)
Note, while this is an option, I have never actually seen it implemented. I have seen multiple PRIs in use, we use it as a dial backup mechanism to maintain connectivity with branches at times, however our PRI's are all 1D23B setups. I wasn't involved in the design, I just use the stupid things at times.
-Rusty
You never know...
# Post Early: If an article has over a certain number of posts on it already, yours is less likely to be moderated. This is less likely both statistically (there are more to choose from) and due to positioning (as a moderator I have to actually find your post waaay at the end of a long list.)
# Post Early: If an article has over a certain number of posts on it already, yours is less likely to be moderated.
This is far different however than posting early. The tactic he/she used was to post early in a parent post to try to boost karma by being closer to the top of the page by astroturfing on someone elses high ranking post as well as asking for mod points.
The parent is right in this case. The kid needs to be sent back to school.
zosxavius photography
It's pretty much general knowledge what a PBX is.
Some trivia: PBXs used to be known as PABXs, when it was new and highly advanced for a company to have a private AUTOMATIC branch exchange, rather than one where a switchboard operator would plug plugs into the right holes to manually route calls. The original PABXs were electromechanical - some large companies had Strowger private exchanges the size of a typical telco exchange unit.
Oolite: Elite-like game. For Mac, Linux and Windows
DO NOT BOOT THE Asterisk@Home ISO !!!!!!
IT WILLE AUTOMATICALLY FROMAT YOUR / PARTITION WITHOUT AKSING YOU A THING.
OMG this is freekin' Unbelievable!!!
I've downloaded the Asterisk@Home and like every other n00b around I just booted up from it (no I've not read any warning s or FULL documentation). I bet 98% of people around to the same thing. After the kernel boot and X detection phase it started formatting my / partition. NOT even a ONE question asked. Of course it screwed my system.
The good part is that I booted from this ISO in a virtual machine (not my real box) and I just lost a default Debian installation (installed on a virtual machine).
Is this the latest way to distribute bad things on the net? Post an article regarding an open source project that do a lot of good things after it FORMAT your / partition?
Please if you wanna try Asterisk@Home ISO do it in a blank new virtual machine not your development/home primary box.
Regards,
the3ngineer
You can use a Digium TDM400P with 3 FXO modules. The TDM400P can accomodate up to 4 modules, with whatever combination of FXO or FXS. Works great here. :)