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This looks like it might be helpful:

http://www.voip-info.org/wiki-VOIP+GSM+Gateways

The site www.voip-info.org itself looked promising during my brief visit....

Mind clueing me into an app that uses less than 13Kbps ( http://www.voip-info.org/wiki-Codecs )?

It's very sensitive to latency and dropped packets, but it's not a bandwidth hog. In fact, I can fit at least 50 calls through a t1 using ulaw ( uncompressed voice codec ). Using gsm, that number jumps to over 200.

Over at voip-info.org they have a bounty on Skype functionality for Asterisk.

I'm kind of sorry that the above could be seen as flamebait. It does contain at least some points that are worth clarifying.

Asterisk is one of several different VoIP open-source / freeware software PBX solutions. One of the things you can do is program a phone menu system into it. It is I admit somewhat of a black art still to actually configure Asterisk but if you can get the hang of it, it is very powerful. If you don't like it, try one of the others. It runs on many platforms, some with hardware limitations and of course the underlying security as a whole. Once a call is in your PBX you can then of course program it to do anything that you can devise.

I personally think the hardware adaptors are expensive for any number/combination of ports (FXO - foreign exchange office and FXS - foreign exchange station - see http://www.voip-info.org/ for a wiki), especially here in the UK if you source locally. I do like the Sipura/Vegastream adaptors for their hardware simplicity though. It may be much better to consider IP phones such as GrandStream or SnomPhone if you are starting from scratch. A mixture of the two is of course what most people will do if they have relatively expensive analogue DTMF telephone handsets.

YMMV especially if you have to deal with a non-US type telephone system as you will need some kind of adaptor at least a one point in your network.

Obviously your IT guys just don't want to be bothered all the time. If you get past the menus then you must have a good (read important) reason to require their time. Time is money especially to four guys supporting many more poeple than perhaps they should. Not many have escaped IT cutbacks.

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This is just being lazy

• http://www.voip-news.com/1/voipwifi.htm
• http://www.zyxel.com/product/P2000W.php
• http://www.vonage.com/
• http://www.webopedia.com/DidYouKnow/Internet/2005/ voIP_WiFi.asp
• http://digital-lifestyles.info/display_page.asp?se ction=platforms&id=1761
• http://www.voipsupply.com/home.php
• http://www.voipuser.org/forum_topic_1072.html
• http://www.voip-info.org/tiki-index.php

Store the call logs in a database. Indicate in your dialplan which types of call are billable and which are free, or "documentation." Then at check out, the hotel billing system queries the database and adds the proper line items to the bill.

You can even match different types of calls and assign different rates. i.e. local, long distance, intra-lata, toll-free, etc.

There are several packages available from various sources to do prepaid phone cards or accounts. Check out http://voip-info.org

I think what you mean to say is that you would like to pick up an incoming call to your PSTN line at home while you are somewhere else, having the incoming call delivered to your via VoIP.

If so, this is quite straightforward. You could buy a little box called an FXO gateway. It's got at least one phone jack for connecting to a POTS line on one end and an Ethernet jack for connecting to a VoIP client on the other end. One of the most popular devices is probably the Sipura-3000.

The Sipura-3000 is actually a combo device as it has both an FXO and and FXS port. So you could also connect an analog phone to it, in addition to your POTS line. When you are out of the house and online with an IP phone or a softphone somwhere else, you could log in to the Sipura's web admin interface and change it's settings so that it will deliver incoming calls to your VoIP phone. By default it will pass incoming calls to the FXS port. You will need to have the Sipura on a public IP address though.

There are other FXO gateways, too. Check ...

http://www.voip-info.org/wiki-VOIP+Gateways/

1. POTS lines will work. You will need an FX0 card per line. Not practical if you need a lot of lines. There are some multi Line FXO cards available. FX0=Hook up to telephone lines. There is a flavor of Intel Modem that will work as a single Line FXO card. They are pretty cheap and would be a good way to build a cheap test or home system.

2. To hook up just plain old phones to Asterisk you need FXS cards. FXS= hook phones up to Asterisk.
Or you can get VoIP phones and hook them up to a 100BaseT or 1000BaseT network. I will probably also want to use a power inserter so you can have power over ethernet or PoE. That way the phones will get their power over the network connection and will not have to have a wall wart.
Or you can use a softphone. A softphone is a program that runs under Windows, Linux, BSD, PalmOS, WinCE, or the Mac that uses your computers soundcard as a telephone.
Your best place to look is the VoIP Wiki http://www.voip-info.org/tiki-index.php.
Another good site is the Asterisk@Home project http://asteriskathome.sourceforge.net/. It is a Linux/Asterisk distro. Pop it in and you get an Asterisk box. Warning! This is NOT a live CD. It will reformat your hard drive and install Linux and Asterisk on it.

You might try the Polycom IP-500 SIP phones. They are supposed to have great speakerphones, just barely under $200 at many places.

We're about to upgrade at my work. Its between the SPA-841's and the IP-500's. Both look pretty nice!

More info from the Asterisk wiki
http://www.voip-info.org/tiki-index.php?page=Polyc om%20SoundPoint%20IP%20500

I'm doing the Documentation for AMP which is probably (IMO) the best admin tool, and it's what is used for 99% of the administration of Asterisk@Home. AMP is rapidly becoming more than just a basic interface to Asterisk tho - the current CVS handles LCR, ZAP Trunks (eg, physical connections to the PSTN via ISDN or normal 2-wire FXO/FXS), Call Groups, Inbound call queues with everything you'd expect ("Your call is 4th in the queue. Your expected wait time is 3 minutes"). The current CVS of Asterisk, when used with AMP, gives you attended transfers, call (audio) recording, and a whole pile of other stuff.

Probably the best thing for someone new to VoIP is to get the latest version of Asterisk@Home (which is 0.9 at the time of this post) and an old machine, a couple of soft-phones (VoIP software that lets you make calls from your PC using your sound card) and a FWD number and start playing.

Feel free to leave me voicemail on my FWD number - 47876 - if you have any questions or comments!

You are mistaken.

VOIP doesn't mean that your computer becomes a phone. All it means is that voice is encapsulated into IP packets. Period.

In fact, a software that allows you to use your computer as a phone, a so called softphone is a very bad compromise.

You will always get far superior quality if you use a real VOIP phone, that is, a device that looks just like an old fashioned telephone, but instead of a phone jack (RJ-11) it has got an ethernet jack (RJ-45) at its back and circuitry to convert between analog voice and IP packet encapsulated digitised voice.

http://www.voip-info.org/wiki-VOIP+Phones

You can do this already with Asterisk

http://www.voip-info.org/wiki-Asterisk+Bluetooth+c hannels

Asterisk runs on OSX, though I am not sure if anybody has tried the BT channel modules with Asterisk on OSX yet.

AstWind using colinux under windows and AsteriskWin32 using cygwin under windows are two ways to get asterisk to run under windows. Basicly if you want to run asterisk under Windows it will involve using a linux emulator, with no hardware support, and lacking a lot of features (no conf!). I would suggest using a Asterisk bootable CD if you're curious about Asterisk.

AstWind using colinux under windows and AsteriskWin32 using cygwin under windows are two ways to get asterisk to run under windows. Basicly if you want to run asterisk under Windows it will involve using a linux emulator, with no hardware support, and lacking a lot of features (no conf!). I would suggest using a Asterisk bootable CD if you're curious about Asterisk.

Yes, all of that is very doable (not that I've done it on mine since I don't have callerid). You'd just need some logic to check the value of the ${CALLERIDNUM} variable.

Take a look at the Asterisk wiki, particularly the variables and commands pages.

Yes, all of that is very doable (not that I've done it on mine since I don't have callerid). You'd just need some logic to check the value of the ${CALLERIDNUM} variable.

Take a look at the Asterisk wiki, particularly the variables and commands pages.

For the people that just want to try things out on the cheap, there is no reason to sign up with a company that charges a setup fee and/or a high monthly fee. Several voip players have no setup or monthly fees and a relatively cheap 2cents/minute. In most cases thats comes out much cheaper that the places that sell you "unlimited" service for $20/mo - $40/mo and then get mad at you if you use over 1000minutes per month. One example of a provider that makes it painless to try out voip is gafachi.

Here is a relatively complete list of voip service providers. The voip market is still very much in flux and the offerings are always changing. It is a good idea to check that list periodically.

I don't buy anything but if I'm bored I try to string them along for as much as possible wasting their money. With any luck I'll eventually be blacklisted and not receive any more calls.

Say "Wow, yes, I'm definately interested. Hold on while I go get my credit card information". Then place the call on hold, and go do something else.

If you run Asterisk, there's a great script to have some fun with them.

Anyone that runs a voip system can always have the system route UNKNOWN or ANONYMOUS callers to a computer based screening tool. One bored gent wrote an elaborate voice-mail maze for telemarketers to wander into.

Telemarketer Torture

So far the only prank SIP call I have received was one from a buddy that was testing his SIP knowledge and wanted to see if he could really make my phone ring.

Broadvoice's config details are misleading, and Asterisk itself can be a bit arcane at times. However check out this wiki page. The "Second example" is actually mine. I've spelled out a bit more clearly what each element is, and used the terminology that BV does for the elements, as well as removed some superfluous elements that don't even do anything. I know of several people who couldn't get it working before tried it, and haven't had problems since.

Oh, and then there's the "Broadvoice patch". It used to be BV people had to patch their own installs to work with BV, but 1.0.6 works unpatched, which saves some headaches. It's more likely this was actually your issue than any configuration.

~Lake

/msg nickserv register mypassword

/msg nickserv identify mypassword
then /join #asterisk

Come on in and say hi!
Some links

The Wiki [voip-info.org] bar none the best resource.

The Asterisk Documentation Project [asteriskdocs.org]

more links [digium.com] (look at the "Unnoficial Links")

Mod me up! :)...

Skype's success can be attributed to its ease of use and the "it just works" factor. Part of that is the proprietary protocol that makes it work nicely through nat's and firewalls, something the SIP doesn't do so well.

However, there is an open VoIP protocol developed by the asterisk project call IAX that works great through nat's and firewalls as well.

http://www.voip-info.org/wiki-IAX

If you're willing to tolerate it, you coud cram 18 or so simultaneous calls using the lpc10 codec. "Domo arigato, Mr. Roboto, Mata ah-oo hima de!"

I don't understand this discussion, or maybe I understand it better than others. I have VOIP. I have a hardware firewall which blocks all but port 80.

My understanding is that, when the SIP device finds all ports blocked, it works around the block. Maybe with STUN.

Skype also works around firewall blocks.

Can anyone explain this?

... if it's alrady a feature.

Asterisk is an open sourced pbx project that has really matured. Some people still gripe about its scripting, and difficultly to set up, but I find it a breeze. If you're relaly interested check out the Asterisk Wiki.

Even better, for those with a landline or VoIP phone, would be a system that automatically picks the cheapest route out for any given call.

Basically, you're looking for something like Least Cost Routers (anybody wanna translate this?). These things have been very popular in Germany ever since the telecom market was deregulated. In Germany you can use other (landline) telecom providers through a Call-By-Call system, dialing the provider's prefix before your actual phone number if you want to use a provider other than your default one (e.g., 01033 for German Telekom, 01013 for Tele2). There's whole websites dedicated to providing lists of the cheapest call-by-call providers. These LCRs can store such lists of providers and their rates for different types of calls (i.e., local, long-distance, other countries, cell phone networks, etc.) at different times of the day/week, and the automatically prefix the number you dial with the cheapest provider's. Of course, lists can be updated manually or automatically. Now, I'm not sure if anybody has built such a device with cell vs. landline vs. VoIP in mind, but if that exists, other Slashdotters who can be bothered to look it up instead of working ;-) will surely post links...

FWIW, there's also an isdn4linux-based LCR tool and corresponding phone rate databases (see English summary at bottom) available. For cell/landline/VoIP solutions, if there's nothing else available, there is probably a good starting point.

VoIP == Voice over IP.

We've run Asterisk http://www.asterisk.org/ for about a year now and when we relocated our offices decided that Verizon wanted too much for long distance. I signed us up with Broadvoice http://www.voip-info.org/tiki-index.php?page=Broad voice and configured our phone server to send all long distance calls out over our 1.5 MB DSL line. It's been PERFECT! I did have to upgrade our Asterisk to the latest tarballs, because my older CVS version couldn't register with Broadvoice though. Dirt cheap long-distance.

but not good for businesses. Skype doesn't offer the "carrier grade" telephony quality/reliability/features businesses are looking for. It's great as a additional line but that's it.

Check out http://voip-info.org/ for a listing of business class VoIP solutions. The best part of something like Skype is outsourcing your communications. You no longer have to be running a PBX in your business. It's what CENTREX was supposed to be.

Asterisk is a big cost? It's GPL. Free. As in Beer. Runs on Linux, BSD, and Mac or even a linksys router. While I don't know how well it works, there is even a version that runs on Windows. Asterisk gives you SOOOO much more than just VoIP and cheap phone calls. It's an extremely flexible full-functionality PBX. It's really a matter of time before someone does a windows port.

You really need more than just an ATA since you want to be able to lock-down the dial-plan. Example: you allow 800 numbers because they are free, right? So someone calls 1800-call-att and makes an operator assisted long-distance call to Japan... Not so good.

That said, I tried bellster. Easy setup, works. I'm in...

So what kind of speeds do you get??? For faxes, you have the T.38 protocol that allows them to work (requires support at both the VoIP provider AND the ATA you are using). Getting modems to work over 9600 is Much more of a trick. First, you can't use any codec that does compression so it sucks a lot of bandwidth, and second, the latency and packetization of the modem signal is going to be quite problematic. See this page for more info on modems over VoIP.

If you can get your modem to work at all over VoIP, good for you (I am VERY surprised to find that someone is using it succesfully.) It doesn't work at all for most people at this time however.

I would like to see an Skype Asterix extension.
http://voip-info.org/tiki-index.php?page=bounty%20 skype
That would be the real killer POTs killer app.
Can any reverse engineering pro give a hand ?

IIRC, doesn't Asterisk let you use any ASDI phone? As such, there should be lots of nice cheap phones you can use (such as the ones specifically recommended by Digium). You need to make sure you can get the programming codes for the phones you buy, but that isn't as difficult as it once was.

The codec that skype uses is freely available.
See:
iLBC

Yep,

Our shop split into two buildings - the old one (sales) and another round the corner across a highway (service).

Using a wireless link (w/IPSec for security on top of WPA, since we have some "intranet" app thingies too), two Debian GNU/Linux boxes with a Digium TDM400 card in each one, we can now:
1) Make internal calls for free as much as we like
2) Dial out using a collective line pool of 7 PSTN lines between both shops
3) Transfer calls from one shop to the other
4) Answer with voicemail after hours
5) Music on hold!
6) Watch caller ID to decide if we want to answer the call (or perhaps transfer them to the MOH test extension, BWHAHA)
7) Use Areski CDR searchable call history/stats database. In fact, my boss(es) have been so completely impressed with this one app it would be very hard for them to do without. We can track: who has rang, who we've rang, when, from which phone, on which line, for how long. The stats we've gathered have been phenominal - hours spent on the phone per day - we never would have imagined. We can also now say which is the busiest day of the week for calls.

All this for the cost of our original commander system which was 8 phones.

We use a mix of Budge-Tone 100s and Snom 190s. If someone could point me in the right direction to make distinctive ring on the Snom190 work, that'd really make my day :-)

The best site for answers to VoIP questions is voip-info.org. Be sure to check the reviews service providers there before shelling out any money. I've tried four so far, and have had a bad experience with two of them.

Josh

Flash is even fronting for Asterisk

sorry, this is what I meant to type in . . http://www.voip-info.org/wiki-VoIP+Training Asterisk is worth looking into - install if you like. It is compatible with sip and cisco's "skinny" protocol and works with cisco phones as well as many others. It also does h.323 (if you like pain), iax2 (used by some big companies such as VoicePulse), adsi, sip, etc. http://www.voip-info.org/tiki-index.php?page=Aster isk You can download an iso and install it from the iso or run from a live cd: http://www.voip-info.org/wiki-Asterisk+installatio n+tips all above are from http://wwww.voip-info.org/

sorry, this is what I meant to type in . . http://www.voip-info.org/wiki-VoIP+Training Asterisk is worth looking into - install if you like. It is compatible with sip and cisco's "skinny" protocol and works with cisco phones as well as many others. It also does h.323 (if you like pain), iax2 (used by some big companies such as VoicePulse), adsi, sip, etc. http://www.voip-info.org/tiki-index.php?page=Aster isk You can download an iso and install it from the iso or run from a live cd: http://www.voip-info.org/wiki-Asterisk+installatio n+tips all above are from http://wwww.voip-info.org/

sorry, this is what I meant to type in . . http://www.voip-info.org/wiki-VoIP+Training Asterisk is worth looking into - install if you like. It is compatible with sip and cisco's "skinny" protocol and works with cisco phones as well as many others. It also does h.323 (if you like pain), iax2 (used by some big companies such as VoicePulse), adsi, sip, etc. http://www.voip-info.org/tiki-index.php?page=Aster isk You can download an iso and install it from the iso or run from a live cd: http://www.voip-info.org/wiki-Asterisk+installatio n+tips all above are from http://wwww.voip-info.org/

sorry, this is what I meant to type in . . http://www.voip-info.org/wiki-VoIP+Training Asterisk is worth looking into - install if you like. It is compatible with sip and cisco's "skinny" protocol and works with cisco phones as well as many others. It also does h.323 (if you like pain), iax2 (used by some big companies such as VoicePulse), adsi, sip, etc. http://www.voip-info.org/tiki-index.php?page=Aster isk You can download an iso and install it from the iso or run from a live cd: http://www.voip-info.org/wiki-Asterisk+installatio n+tips all above are from http://wwww.voip-info.org/

if you can travel . . . http://www.voip-info.org/wiki-VoIP+Training/

Take a look at Asterisk http://www.asterisk.org/. The wiki http://www.voip-info.org/wiki-Asterisk has more useful information. It is a full VoIP softswitch solution. In addition to SIP, H323 and MGCP it also supports the IAX protocol, which was designed to be NAT friendly. You won't be able to run it point-to-point. You will have to run an Asterisk server somewhere in your network, but since you are already running Linux on the desktop, it should be fairly easy to run it on one of them for a small network. Combine that with a dynamic dns service like dyndns.org and voila!

Your point (1) is faulty. Linux may be Asterisk's primary platform but since it's open source you are just as free to go ahead and run it on, say, BSD or Solaris or Mac OS X. Meanwhile ports to more esoteric platforms are certainly an option, and it's already possible to run it on Windows if you have compatibility layer software. Linux will probably be the most likely platform to benefit from Asterisk being popular, but Asterisk definitely has potential outside of Linux.

Your point (1) is faulty. Linux may be Asterisk's primary platform but since it's open source you are just as free to go ahead and run it on, say, BSD or Solaris or Mac OS X. Meanwhile ports to more esoteric platforms are certainly an option, and it's already possible to run it on Windows if you have compatibility layer software. Linux will probably be the most likely platform to benefit from Asterisk being popular, but Asterisk definitely has potential outside of Linux.

Right.

The world would be far better off if everyone installed Linux on a spare computer so they could run Asterisk. You then just need to buy a bunch of hardware, and then either spend a few hundred dollars each on WiFi phones, or spend tens of hours recabling your house.

Oh, and then you get to configure the mess, after learning all about such eccentricities as G.711, G.723.1, GSM, IAX, and SIP, SCCP, plus a whole lot of other defacto telephony standards and Ways Of Doing Things that were obviously developed in a cave.

Once you solve the echo problem, all you gotta do is make DUNDi work, and you can finally call other people Just Like You. Or, you can sign up with any of dozens of shady small VOIP telephone companies and pay a few tenths of a cent per minute to talk to regular people via a SIP, IAX, h.323, or MGCP connection.

Sweet.

Alternatively, one could always download and install Skype. I understand that it does work fairly well, and is easy for mere mortals to use. It seems that Siemens now has an easy way for you to use their handsets with it. Neat.

>but the telcom admin of a large corporation isn't going to want to look at a text file to figure out his dialplan or use some arcane interface
As with most things Linux, the install problems usually get simplier through heavy use.

I tried installing asterisk 18 months ago, and wasn't getting their, not much like http://voip-info.org/wiki-Asterisk that I could find then, not much hardware in the market place...

3 weeks ago, I tried again, succesfully. 10* better. now dozens of voip phones, and config scripts...

it's still at the state of hundreds of config files floating around, but you already got:
http://dontpokebadgers.com/rss2cisco
giving scripts that auto configure your cisco config files, and extensions, passwords,... with a querry from a mysql database, and http://sourceforge.net/projects/jasterisk/ that lets you do all real time call routing/handeling from a PC app.

granted both of these take a bunch of config options, but so do those pripriotory systems yo mention. just you send them to a company to configure off site before the bring the $100,000 box's to the site, a couple months later.

Get a linux guru a couple months, and you will have all those fancy java/cgi/whatever scripts doing all this from a single website, thats more intuitive than the costly box. I feal confident saying this, because I got the PBX all loading, and sending XML content on scripts I customized, in 3 weeks, and I had no idea what the rc.d directory was good for, and never wrote a cgi script before.

The Asterisk Wiki
Note: the wiki search is useless. Search with google instead, use "searchterm site:voip-info.org" (without quotes).

The Asterisk Documentation Project

The Asterisk Mailing Lists
Note: to search the lists use google again. "searchterm site:lists.digium.com" (without quotes)" in google.

the #asterisk chat room on irc.freenode.org. Drop by and say hello.
Note that due to problems with massive spambot attacks regisitration is required to join the channel. Simply type
/msg nickserv register mypassword
/join #asterisk

The next time you join you will need to type
/msg nickserv identify mypassword

This company is probably nothing more than someone running Asterisk, using Nufone for the PSTN service.

A simple php script will dump a callfile into /var/spool/asterisk/outgoing and bridge the two calls together.

Then all you need to do is write something to manage user accounts, and accept paypal payments and bam. You've got camophone.com.

This whole configuration could probably be whipped up in a day.