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The problem isn't the *obvious* issue. I mean, it wasn't obvious to me in 1995, or most other people I'd wager.

Grand Central doesn't have my area code, but thanks for the tip. Does Gizmo connect to the PSTN, though? Or would I still need an outbound provider

I'm running Vonage on phone jack 1 and Vitelity on jack 2 of my adapter since I've unlocked it. $3/mo + $0.011/min inbound and $0/mo + $0.0139/min outbound is way better than what I've been paying Vonage, based on my usage. But I wouldn't mind staying with Vonage till the end of my second year at $5/mo rather than pay the $40 termination fee. I just wouldn't call out ($0.04/min sucks...).

I should call them up, and tell them I'm planning to switch, I suppose.

Anyway, I'd check over voip-info.org before settling on a DID (incomming) or termination (outgoing) provider.

There is another post closer to the end of the comments, it explains more about FastTrack. Basically, if the IP addresses match and are public then you open your connection to that host:port without further ado. If the IP addresses differ then you take the public one (which is in your IP envelope that you received) and try to open that address:host in hope that the NAT will forward it to the right box. It's called "firewall penetration" and there are some more ways to do it, all completely legitimate and intended to keep the devices on the private network functioning - things like VoIP phones, for example, use STUN protocol.

though the defense lawyer didn't seem to pick up on that detail

Direct physical evidence will outweigh the indirect one. They seized the computer of the accused, and it's clean. IMO, there isn't much to base a trial on. I think the lawyers are simply going through the motions to get their pay.

There's a great script on voip-info.org to use to torture telemarketers.

I have been told that there are certain softmodems -- ones that use a particular Intel chipset -- that you can find for under $10 that will work under Asterix in place of the proprietary cards. I think it's the Intel 537 series chipsets that are the key. Since all they do is bridge the phone line to the computer, they can be used for voice or data given appropriate software. Given how few modems I've seen in stores lately, eBay may be your friend.

Might want to see this article:
A $10 Linux Answering Machine

Now that would mean basically one line, either to the telco PSTN system or to an extension, per PCI slot, which unless you had some abnormally large mobo is probably not too great. However, you could load up the PCI slots with them, use them all for the outgoing lines (to the telco) and then do your internal extensions over VOIP. That would also have the advantage of not having to run analog phone lines throughout the office; everything is just on your data network provided it can handle the load. Might be a bit of cash though, if you don't have VoIP phones already.

Also, there is hardware besides Digium's which is compatible with Asterisk, see this list for an overview of vendors.

A few months ago I tried to attach my Mother-in-law (who lives overseas) to my Asterisk server. I bought a SIP analogue telephone adapter and went to visit. I spent at least 2 hours a day for 8 days trying to get it to work. I fiddled with every setting on the ATA and (via an SSH tunnel) loads of settings on my router and I never got the ATA or any SIP softphones to work. Finally, I tried an IAX2 softphone and it worked first time.

So, when I returned to the UK I bought an IAX2 phone for the same price as the ATA and sent it to Estonia. My Mother-in-law plugged it in and it worked first time. Now she and my wife chat for hours without it costing me a penny.

Conclusion: SIP works fine on a LAN (where I now use the above ATA), but if you have any NAT anywhere in the link then the pain of making SIP work isn't worth it when you can get IAX2 kit for the same price and it "Just Works".

It's possibly the most used, but it is _NOT_ the only open source GUI.
There are lots of others: http://www.voip-info.org/wiki/view/Asterisk+GUI

The snoms, on the other hand, do _not_ have echo cancellation in their speakerphone, which means it can't be all that loud.

I have trouble believing that -- our single Snom 360 sounds as good as the (POTS) polycom units when on speakerphone, and we certainly don't run it quiet. Looking through their release notes, it says they added echo cancellation as of firmware version 3.60b. I don't see any complaints about lack of echo cancellation at http://www.voip-info.org/wiki/view/snom+360.

I haven't had a chance to work with a Polycom SIP phone, which is a little surprising since they're literally right down the road a bit (before we moved, they were the next building over). We hired one of the engineers who used to work on firmware for their videophones, though, and he had nothing good to say about how their software group was run. That said, I don't doubt their speakerphone support is excellent -- it's what Polycom is known for, after all.

It's ok if you are not interested but there are some who are. Here is an article about some of the benefits. http://www.analogzone.com/nett0307.pdf as well as a wiki entry from http://www.voip-info.org./ http://www.voip-info.org/wiki/view/Wideband+VoIP

It's ok if you are not interested but there are some who are. Here is an article about some of the benefits. http://www.analogzone.com/nett0307.pdf as well as a wiki entry from http://www.voip-info.org./ http://www.voip-info.org/wiki/view/Wideband+VoIP

I took a WiFi VoIP phone (zyxel) home and it used the Asterisk server at work from behind my Linky's NAT just fine

1. Because it works in your situation, it's not a good idea to generalize.

2. Conveniently, you fail to mention how you are connecting to the office network.

3. http://www.voip-info.org/wiki-Asterisk+SIP+not-pro xy explains the difference between it and a proxy. Please read it and consider carefully.

The task of proxying over heterogeneous security appliances and public/private networks is not as easy as you claim.

It's called the Session Initiation Protocol, and pretty much every VoIP service OTHER than Skype uses/supports it. (With a few small exceptions such as Google Talk which uses the Jingle VoIP extensions to the XMPP (aka Jabber) protocol). Note in that particular case that Google and many of the big proponents of SIP (especially Project Gizmo/SIPPhone) have been working on solutions for XMPP+Jingle interoperability with SIP.

There are a wide variety of SIP softphones available, just as there are a wide variety of SIP service providers. Many of these also support the IAX protocol, which is primarily used by Asterisk PBX systems.

Examples, most of these service providers provide their own SIP client, but in most cases SIP clients are interchangeable between SIP services:
StanaPhone (http://www.stanaphone.com/) - Free incoming DIDs (dial-in phone numbers) in various New York area codes
SIPPhone/Gizmo Project (http://www.gizmoproject.com/) - Free PC-to-PC, DIDs and outgoing PSTN cost money (not much though)
Free World Dialup - Primarily PC-to-PC (or Asterisk-to-Asterisk or whatever), but with some PSTN in/out capability
The list goes on and on, and I haven't even included the "landline replacement" VoIP providers. (Vonage in the U.S. is the most well known example, but most educated consumers hate them as they have some rather customer-unfriendly policies such as locking telephone adapters to their service and forbidding the use of your own telephone adapter without paying a significant extra fee). A few other providers do use other (although usually still known and standardized) protocols such as AT&T CallVantage (which uses the MGCP protocol).

See http://www.voip-info.org/ for LOTS of addition information on hardware, setup, and cheap providers.

Your wish is my command... VoIP wiki.

You could always do a code bounty.

If I wanted to get around it, I would just Interface my cellphone with Asterisk and forward all calls to a friend's or prepaid cellphone. Thanks for the suggestion, I think I am going to start my own "phone company" soon!
Too bad I spend all my free time reading Slashdot. As a result I don't have any friends to go anywhere with.

Asterisk is too complicated for you to configure? Unable to add the FreePBX web interface? Can't manage to get the Flash Operator Panel working?

Let me introduce you to Asterisk@Home which is uber-easy to configure (get your PBX up and running in an hour or two!), or if the "@Home" name is too objectionable for your PHB, the shiny Asterisk@Work logo so you can convince him that an open source project is suited for business use.

I think a little more dilligence in your research is in order.

Google query
VERY first hit
Unlimitel Inc.

You only need an instrument capable of switching to GSM when Wifi is not available.

Something like this perhaps:

http://www.gizmodo.com/archives/motorola-cn620-sea mless-wifi-to-gsm-voice-calls-017270.php
http://www.voip-info.org/wiki-VOIP+Phones
http://blog.tmcnet.com/blog/tom-keating/mobile-pho nes/zyxel-dualmode-gsmwifi-phone.asp

Enjoy :P

This is true, but the only requirements of the GP were that it use "SIP or h.323 ... be installed and used easily, [and cope] ... with NAT routers transparency (sic)"

Gizmo is not open source, but it uses open standards (you can use it with asterisk!) and it is a heck of a lot better than skype.

There are many VoIP providers that specialize in SME and SOHO. Check out http://www.voip-info.org/ for a large list.

You mean like:

http://www.utstar.com/Solutions/Handsets/WiFi/
or
http://www.vocera.com/ (star trek - like)
or
http://www.voip-info.org/wiki/view/ZyXEL+P2000W

All are wifi SIP phones and work well with Asterisk

Looks like the thread full of insults and lookie heres RE: you timing post missed the point. I will assume you are referring to this:

grep ZAPTEL `find . -name \*.c` ./apps/app_meetme.c:"Please note: A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING TO WORK!\n\n" ./apps/app_meetme.c:"A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY.\n"; ./channel.c:#ifdef ZAPTEL_OPTIMIZATIONS ./channel.c:#ifdef ZAPTEL_OPTIMIZATIONS ./channel.c:#ifdef ZAPTEL_OPTIMIZATIONS ./channel.c:#ifdef ZAPTEL_OPTIMIZATIONS ./channel.c:#ifdef ZAPTEL_OPTIMIZATIONS ./channel.c:#ifdef ZAPTEL_OPTIMIZATIONS ./file.c:#ifdef ZAPTEL_OPTIMIZATIONS ./file.c:#ifdef ZAPTEL_OPTIMIZATIONS ./file.c:#ifdef ZAPTEL_OPTIMIZATIONS ./file.c:#ifdef ZAPTEL_OPTIMIZATIONS

More or less, the timing lesson you were given as a response is not untrue but is also not relevant to your question. Where it may be true that for TDM you have an enormous dependency on time sync you have much less with VoIP. The bigger reason for the Zaptel timing for non-hardware stuff comes from the hatred of threads that Asterisk has.

In a typical call in Asterisk, 2 channels exist within 1 thread. This thread must poll both channels and service them based on socket events. The first problem is that not every channel type has a socket (most do but IAX2, for example, does not which is another can of worms). The second problem is that most routines in Asterisk make the assumption that if you read from the channel in a loop you can write to the channel on each pass and have perfect timing. Sorry, thanks for playing, but you cannot do that. Many protocols (SIP for example) have VAD (Voice Activity Detection) support.
This means that the far end of your call may not send any voice because the other guy is not talking so why bother. When this occurs the poor Asterisk channel expecting reads to dictate writes simply halts in place DoH!

This problem is fixed by a friendly "Don't use VAD" console message. And they are trying to improve on it by somehow trying to weave some logic in there to make that poll timeout and let it work still using good ol' Zaptel timer but it's like digging a hole too deep and just makes the mess bigger if both channels we in their own independent I/O thread the problem would be gone.

The next place Zaptel is wedged into place is in the channel I/O routines. A pipe is opened inside the channel to allow the queuing of frames meaning you can write them as fast as you want and the channel will deliver them at a steady interval. Thank goodness Asterisk is hard coded to only operate at 8000hz! This also compensates for the channels with no socket you can just queue all the inbound frames instead of write them.
This constant pulse is also courtesy of Zaptel timer.

Next there is the file delivery system which is really the one place you really need a timer for sure since the file is simply a large collection of data so it must be fed at a timed interval this also is somewhere the Zaptel timer is used.

The timer is also used by the external music on hold system. The default system uses an external pipe to an mp3 player that also spits the data out too fast since it has no soundcard for timing so the Zaptel timer to the rescue again!

The one place you cannot avoid the timer is the MeetMe application. This is a conference system that depends on the Zaptel to do the audio buffering/muxing and the timing. There is, however, a purely soft-timed algorithm-based conference application available.

http://www.voip-info.org/wiki-Asterisk+app_confere nce

Nearly all of these uses can be replaced with low resolution timers and normalization techniques (gettimeofday with incremental waypoints) a

I got mine on the monopolistic online auction site eBay.

Here is one for $5:
http://cgi.ebay.com/Digium-Wildcard-FXO-card-for-a sterisk-X100P-OEM_W0QQitemZ5858067510QQcategoryZ61 841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
$10: http://cgi.ebay.com/Clone-of-Digium-X100P-X101P-Wi ldcard-for-Asterisk-PBX_W0QQitemZ5856427611QQcateg oryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
Buy it now $13: http://cgi.ebay.com/Asterisk-FXO-PCI-Card-OEM-X101 P-100-Compatiable_W0QQitemZ5856065846QQcategoryZ11 182QQrdZ1QQcmdZViewItem
Buy it now $15:
http://cgi.ebay.com/Digium-Wildcard-X100P-OEM-FXO- PCI-Card-for-Asterisk-PBX_W0QQitemZ5858250361QQcat egoryZ61839QQrdZ1QQcmdZViewItem

Yes you have to factor in shipping, but if you buy multiples you can get that down.

These are generally Clone cards that work every bit as good as the originals.
The orginals were just Intel v.92 modems anyway.

http://www.voip-info.org/wiki/view/X100P+clone

Hope that helps.

http://voip-info.org/wiki/index.php?page=Asterisk

While there certainly are the Vonages of the world, there are far more VoIP services that permit you to connect any phone you like.

Having a good permission system enables systems where you can decide how people interacts with the system (i.e. adding or viewing content) and where. The extensions enables i.e. drawings editing in a wiki-like scheme and its integration enables i.e. putting portions of blogs in wiki pages or including editables spreadsheets in.

My biggest use of it was for documentation, mixing blogs for tracking progress of projects, adding editable diagrams of networks, organizing and grouping the information in wiki pages and giving different kinds of priviledges to the development people (editors) from different kinds of viewers. But a lot of people gives a lot of different kinds of uses for it i.e. KDE project or voip-info.org.

Software depends on how you're intending to do the encryption. If you're planning on hooking up to a regular phone system as well as doing VoIP, then you're certainly looking at using Asterisk for your exchange system.

At the kernel level, for pure VoIP, you probably want to use either Linux with either the StrongSWAN or OpenSWAN patches applied, OpenBSD or MirBSD. (I believe FreeBSD and NetBSD have IPSec, but I'm not sure.) This allows you to encrypt from your machine to the destination, using a grade of encryption that will be proof against standard wiretaps. Rijndael (AES) is (as far as anyone knows) uncrackable on existant technology and if you combine it with SHA-2, you've a system that would be impervious to any wiretap you're likely to encounter.

For working with Asterisk, you want to use a stream cipher, not a block cipher. Asterisk with Encryption WIKI has more information on how to set it all up. You basically link up a VPN and encrypt the VPN. Because some line loss is inevitable, you want an algorithm that is resistant to loss and is reasonably fast. There are phones that have hardware encryption in them, which provide higher levels of security and they are also listed.

Of the stream ciphers out there, FEAL/SEAL and Chameleon seem to be the most reputable.

Asterisk is a wonderful program, but for a real company, let someone who knows what they're doing design your phone system. Having done it myself, it may have been easier to pay someone else, the number of variables in a phone system are staggering and the thing is still a beast to maintain. What happens to your company when the person who designed the Asterisk system leaves? What kind of phones should you buy?

Saying any IT department can design you a phone system with Asterisk is like saying any IT department can design you a web site with Apache. AMP is good for a small setup, but I found it lacking and inflexible for even the needs of a small company. Asterisk is definitely the way to go, but try and see if you can find someone to do it for you. You'll save yourself a lot of headaches.

I'm not saying Asterisk is something you can't/shouldn't learn, but there is a pretty good learning curve if you're new to VoIP and phone systems in general. This wiki has a lot of good information about VoIP in general, though mostly how it relates to Asterisk.

First and most important, is quality of calls. If you want to ensure voice quality, it is best to seperate your voice and data networks. They should be seperate lans, or at least VLANs if your switch can handle that and ensure that something like SQL Slammer won't fill the backplane on the switch and impact voice as well as data.

Second if it is ever necesarry for the voice traffic to be "trunked" over the same link as the data at any point. You will want to ensure you put some sort of traffic shaping box in there to reserve as much bandwidth as you need for your voice traffic. (about 80 kbps per concurrent call for uncompressed voice) If you use GSM, or any of the other different companders/compression you can change that number, but it will change the quality of the voice.

Phones: I like the Polycom SIP phones, the Soundpoint IP 501, is a 3 line phone, with a great speaker phone, and is a reasonable cost compared to some other phones.

If you are looking at Asterisk, study voip-info that is where all the info is. Also know that until you get everything working the way you want it to work, at least someone in the IT department will be kept busy. People need phones to work, and they have certain things they expect out of them.

example: If you don't put a 1 or 2 second pause of dead-air into the line between when a user makes a selection on the AutoAttendent and when the next AA or VM starts talking you will miss the first fraction of a second of the announcement because it will start before the user takes his finger off the button. And there will be tons of other little gotchas.

There are plenty of companies out there that will let you outsource your PBX and have IP phones on your premesis, and some even make financial sense (mainly because you don't have capital expenditure, and need to learn anything except how to run the handset). This doesn't sound like what you want, but it is worth checking them out. (full disclosure I work at CoreDial so I of course wouldn't mind if you checked them out.)

And were are the ACD capabilities for Asterisk?

http://www.voip-info.org/wiki-Asterisk+Wishlist

Plus if one's running a call-center one needs to be able to keep track of overall phone system stats, as well as per-agent.

• IP Phones are readily available starting at around $45US a set for cheapies (new, but low frills and crappy docco), up to several hundred a set for top-o-the-line units from folks like Cisco. I would personally recommend at least two or three for your pilot project, and not all the same model.
  
  
• Beware the "power adaptor problem.' Some VoIP phones are designed to use POE (Power over Ethernet), where the switch provides the power over the ethernet cabling just like the phone company. If the phone sets are designed for this, they may not come with power bricks, and these particular bricks can be very expensive, and add considerably to the cost of the phone set.
  
  
• ATAs (analog telephone adaptors) let you plug a phone (or a fax, or both) into an ethernet link connected to a VoIP lashup. These are what a LOT of the commercial VoIP providers furnish or provide at low cost. There are LOTS of these available on the secondary market, and many can be unlocked to use with any provider. I'd recommend you play with a couple different ones of these as well.
  
  
• There is a metric a$$load of information on VoIP, Asterisk and Asterisk @ Home at VoIP-Info.org. Among other things, you can find info on which phones (soft, hard and ATAs) are well supported, and config info for lots of specific models.
  
  

Troll? No. Clever? Thanks, yes I know, but don't let that frighten you.
As an AC I don't get any real chance to defend my side but I can
promise you that I have built at least three _working_ systems.
Jitter can be solved many ways, simple reordering, windowed backbuffering,
lots of fancy stuff - or for a budget DIY system just don't bother about
it too much, it's not like the OP asked for a fully commercial system is it?
Here's a few obvious links to help. I'm guessing you are enraged by my suggestion
because you work for a commercial VoIP provider. What can I say? Please grow up.
I also guess you have never actually built such such a system either, try it, go on,
fire up that C compiler and amaze yourself at how easy it is. Really, you don't need a load of fancy stuff, VoIP hardly even needs a processor, man you could get a 4MHz Z80 to
do most of what is required.

here
here
here
here
here
and here

I set up a small voip system in our office in NJ (3 lines) using broadvoice paired with asterisk - and while the service (most notably broadvoice tech support) leaves some things to be desired - our phone system is much better in terms of its feature set than it was on our POTS pbx. That said, most of the reliability issues we've encountered were the fault of our service provider, and we're generally quite happy with the switch.

The website i found myself constantly referring to in terms of making phone, software, hardware and other choices - as well as finding out the quirks and perks of each and mountains of setup info is the voip wiki.

Cheers, and good luck - you may need some in the process.

I've switched to using http://asterisk.org/ along with http://www.broadvoice.com/rates_compare.html. I think you'll find this Wiki to be a very useful resource: http://voip-info.org/

The plan I'm using is BYOD-Lite which costs me only $6 a month and there was no activation fee, since I had my own VOIP equipment in the form of an Asterisk PBX installed on Linux. From what I can tell, they are one of the few providers who allow the use of customer supplied VoIP hardware/software, in my case Asterisk.

Something you'll have to research is what technology you want to use for hooking up individual phones to Asterisk. One possibility would be to use hardware from Digium: http://www.digium.com/index.php?menu=product_categ ory&category=hardware or any other Analog Telephone Adapter (ATA), or you could use Softphones installed on employee PCs such as X-Lite (free), or similar.

Good Luck!

http://www.gloryhoundz.com/

Here's a page with a list of Open Source VOIP applications, both clients and servers.

I would agree with the other posters about it being the protocol designer's fault, not NAT. Contrast the SIP protocol with the IAX (Inter-Asterisk eXchange) protocol. Not only does it support NAT without any problems, it has a host of other features that make it superior to the SIP protocol in almost every way. Just take a look at http://www.voip-info.org/wiki-IAX+versus+SIP for a point-by-point comparison.

For whatever reason, IPv6 is still a long ways off from being universally adopted. Whether or not people like NAT, it is a useful tool, heavily deployed, and an inescapable reality. New protocols should be designed to work with it.

I put all my phone numbers on the "Do Not Call Registry" in the US, and have experienced a great reduction in the number of telemarketing calls, but, the Do Not Call registry does not apply to Charitable institutions, and a few others, and the volume of these calls has grown exponentially over the months... it seems the charities sell each other their call lists, and if you give anything to one, soon you will have them ALL calling at regular intervals.

I've been playing with Asterisk for a couple of years now. I've implemented every privacy option in my dialplan, and have finished the coding of the call filtering option, and had it incorporated into the 2.x releases.

First the 3-tone is played (the da-dee-doo that precedes "The number you have called is no longer in service!", if no CallerID is present.

Next, if no CallerID is present, and the autodialer has not hung up, the calling party is forced to supply a phone number, or the call is terminated, and if they are stupid, and give my number instead of theirs, the call is "terminated with prejudice".

Then, they get a menu, where they must choose the person with which they would like to converse. They get music on hold, and if no answer, they are thrown into voicemail.

One of the first menu options they are presented with is a number to press if they are telemarketers. This option runs them into what I titled the "Telemarketer Torture Script". (See http://www.voip-info.org/wiki-Asterisk+Telemarkete r+Torture)

I have complete CDR logs of the incoming and outgoing calls, and have totalled up how many calls and from what sources, and at what stage each call was terminated.

And I found that each measure I implemented removed a fairly small percentage of the callers, with the total result that I have not received any telemarketing calls or requests for donations in almost two years now.

The single most effective measure is the menu prompts. It defeats autodialers, which aren't programmed to answer prompts. It somehow defeats all the rest of the live callers, who don't seem to have the courage to ignore the telemarketing option, and choose a person. Only once has a charity ever been brave enough to actually select my extension, and in that case, all they did was thank me for past contributions, and hung up.

My conclusion is, that if you truly want to eliminate unsolicited calls from your business/home, you need to implement a simple IVR menu system.

A more detailed explanation of the privacy measures are outlined in http://lists.digium.com/pipermail/asterisk-cvs/200 5-July/006992.html

And, some details of my research results are in: http://lists.digium.com/pipermail/asterisk-users/2 004-September/062571.html

Best of luck!

What if some [terrorist, child porn, etc.] group decided to set up a network of Asterisk or Bayonne servers, virtually circumventing any established VoIP providers? I'm not sure about Bayonne, but Asterisk is extremely easy to throw together and set up. Will they make setting up such "unlicensed" servers illegal? I shudder to think what that would do to the community at large...

>Skype does use encryption:
>Where as something like Asterisk doesn't:

Asterisk does use/support encryption, just doesn handle the mentioned SIP version.

http://www.voip-info.org/tiki-index.php?page=Aster isk+iax+rsa+auth

this is the IAX protocol, which is what asterisk uses between asterisk box's, and at least one VOIP provider uses (IAXtell, if they still exist.)

Why is the parent insightful? Skype does use encryption:
http://support.skype.com/?_a=knowledgebase&_j=ques tiondetails&_i=145

Where as something like Asterisk doesn't:
http://www.voip-info.org/tiki-index.php?page=SIP+e ncryption+for+Asterisk
even though there is a bounty to implement it:
http://www.voip-info.org/tiki-index.php?page=Aster isk+Bounty+SIP+encryption

You might not like skype for a number of reasons, but lack of secure communication isn't one of them.

Why is the parent insightful? Skype does use encryption:
http://support.skype.com/?_a=knowledgebase&_j=ques tiondetails&_i=145

Where as something like Asterisk doesn't:
http://www.voip-info.org/tiki-index.php?page=SIP+e ncryption+for+Asterisk
even though there is a bounty to implement it:
http://www.voip-info.org/tiki-index.php?page=Aster isk+Bounty+SIP+encryption

You might not like skype for a number of reasons, but lack of secure communication isn't one of them.

"...if you represent a market research organization, please dial 4..."

http://www.voip-info.org/tiki-index.php?page=Aster isk+Telemarketer+Torture

Cheese and rice... EBAY paid this much money for simple software VoIP. Imagine if they had simply thrown $100 million at the Asterisk project with the following criteria:

1) NAT-friendly Windows and MAC VoIP client
2) NOTHING!

Seriously... If you provide clients, Asterisk already has the capability to do EVERYTHING that Skype has. They even have a NAT-friendly IAX protocol. They'd just need to develop the clients and then build a large PSTN interface.

The inefficiency in business is astounding. EBAY had money burning a hole in their pocket so their remedy was to remove the money and throw it into the fire.

I'm breathless.

Apparenly, that changed about 4 months back... Asterisk SS7 support became available in April.

http://www.voip-info.org/tiki-index.php?page=How+t o+connect+VoIP+GSM+gateway+to+Asterisk+PBX

The only company I have found that is interested in actually serving the customer so far is braodvoice. They will let me use Asterisk or my own equipment and let me retain control of my equipment. All the others I found refuse to.

There are several VOIP providers that allow you to BYOD (Bring Your Own Device). Check out http://voip-info.org/tiki-index.php?page=VOIP+Serv ice+Providers+Residential and http://voxilla.com/compare/compare.php?typeid=1&mo de=type

Personally, I'm looking at voipex - https://www.voipexinc.com/ using a Sipura 3000 so I can still receive calls on the PSTN number I've had for the last 13+ years...

There are several services that allow free inbound calls from pstn - but require you to pay for outbound.


sipgate.co.uk - free inbound UK and German numbers
stanaphone.com - free inbound NYC and Area
messagenet.it - free inbound italy number

I have one asterisk PBX here in Toronto with inbound phone numbers in all of the above - I don't pay for a single one of the inbound numbers (I pay for local service thru Vonage).

This site has a lot of useful information on SIP providers and Asterisk http://www.voip-info.org

Well, there IS the Telemarketer Torture script...

I don't understand the 300mS cross-US latency figure -- my informal ping'ing has an average roundtrip less than 20mS across the US. And AFAIK, the QoS requirements of VoIP require a max of 120mS to 150mS (eg, http://www.voip-info.org/wiki-QOS). I don't use VoIP or Skype on a regular basis...has anyone ever experienced 300mS?

I wasn't aware of this paper. Looks like someone is gunning for the prize