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... ."the right long distance company."...

or more specifically the right SIP trunk provider. I worked for a company that made lots of automated calls (legally, within the bounds of market research). We had plenty of providers who would just say no, but there were enough that didn't even bother to ask. They all made sure we sent an ANI, although they didn't seem to care what that ANI was, or if we even owned it, (which we did), and they made sure our call duration on connected calls wasn't below 12 seconds (I don't recall what the short duration regulations are, but I believe that was why we had to stay connected at least that long.)

Those trunk providers are also rampant with this: https://www.voip-info.org/fake...

Pretty much. We looked at the cost and challenges for encrypting SIP communications on our local LAN, and it just wasn't worth the hassle. We will segregate the phones onto a separate VLAN, but the value is limited; SIP deployments really aren't focused on security yet.

We control the financial aspect by carrier-enforced rules which prevent toll calls. Much more effective. (We do have a way to make calling card calls through our Asterisk system that is sufficiently locked down and only has $100 or so at risk.)

What system are you using that doesn't inherently support SIP authentication?
http://www.voip-info.org/wiki/...

The biggest risk for most implementations is toll theft so while encryption may not be necessary you should still be able to authenticate call setup and control.

If you talk to people over the internet, latency is an issue. Like, you say something in Skype. The person at the other end hears it and replies. By the time you hear the reply, a regime change has taken place and there's a new president in power. Currently internet video chatting over long distances is an unpleasant experience due to the lag.

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

Callers usually notice roundtrip voice delays of 250ms or more. ITU-T G.114 recommends a maximum of a 150 ms one-way latency. Since this includes the entire voice path, part of which may be on the public Internet, your own network should have transit latencies of considerably less than 150 ms

I'm already getting cross country ping times of 65ms (round trip), so to be compliant with ITU-T G.114, my codec has 235 ms to do its work. I regularly talk on the phone with colleagues on the other side of the country using a VOIP system hosted here, and haven't noticed any latency problem. Even video calls using our Polycom have good latency (but not great, there's still a noticable lag, even when connected to local users)

I find voice latency on cell phones (even local calls) to be far more noticable and annoying than with cross country VOIP calls. I suspect the lag you're seeing is due to client side buffering and codec compression more than network latency.

It's surprising because VoIP can do a lot of things to make it not work, and indeed the most reliable way (if supported) to use a fax machine over a VoIP system is to use the protocol developed just for that purpose: T.38. If the fax machine instead uses the audio channel to encode the fax, then the VoIP system cannot use a compressing codec, and latency, jitter and packet loss must all be close to optimal for a successful transmission.

On the topic of sounds we no longer hear: modem synchronization, the sound of trains rolling on rails with gaps, the rumble, static and pops from vinyl records, the screeching emitted by dot matrix printers, the charging whine of photoflashes, the whirr and track-to-track clacks from floppy disk drives, chalk on blackboard. Good riddance, I say. The world is too loud anyway.

This is not your run of the mill ogg or mp3 codec. These codecs are typically VBR (variable bit rate), this means that the level of compression varies based on what the codec and network conditions dictate. If you swing the bitrate enough you WILL hear changes in volume and quality that sound like a flanging effect.
http://www.voiceage.com/Audio-...
http://www.voip-info.org/wiki/...
You can :rolleyes: when you know more than just bb smiley code. Until then, hand in your geek creds please.

There is no adapter that will make a rotary phone work on a touch-tone-only phone network.

Oh, but there is!

http://www.voip-info.org/wiki/view/Dial+Pulse+to+Touchtone+DTMF+Converters

Or you could skip DTMF all together and go straight to SIP.

http://www.realtonetech.com/product/voip-gateways/83-sip-gateway-ata.html

Factually inaccurate statements make me doubt your story about deploying VoIP on a large scale. UDP makes no difference, and your incorrect assertion that VoIP has ZERO forward error correction is something I wouldn't even expect from an entry level CCNA.

Will not solve the problems associated with running large VoIP deployments on the public Internet. Period. You have clearly never done this, you're just guessing because you used Skype once. You don't seem to understand how sensitive voice traffic is, so I'd suggest you do a little reading.

And for the record, you most certainly can throttle incoming connections, it's just not as finely controllable and beneficial as outgoing QoS, which is more common. But I made it clear the first time around I was talking about controlling both ends of the connection.

Not on the Internet. Let me explain this in a concrete example so you understand why you can't run QoS on the Internet.

1. User A starts a VoIP call over your Internet connection. For some reason you QoS outbound VoIP traffic (even though it's "non-criticial" - your words)
2. User B starts to download a large file
3. As traffic leaves your router, towards the Internet, the VoIP traffic is given priority in the output queue.
4. Now, the traffic starts coming back. Your providers router now starts sending the traffic BACK to you. He has no outbound QoS policy to prioritize the RTP stream in his output queue. The large file traffic now completely stomps on the VoIP traffic, causing packet loss and delivery delays.

If QoS worked on the Internet, we'd all mark our traffic with the highest priority AND THEN QOS WOULDNT WORK ON THE INTERNET. Use your brain.

For that matter, I think I was extremely clear that I was talking about latency guarantees by the rest of my statement you didn't quote... They are empty promises of service you'd be getting anyways, and only worth the price of the penalties your ISP has agreed to pay. It's not at all unusual for companies of offer ridiculously impossible SLAs, knowing it'll bring in more business, and paying the penalties is cheaper than actually maintaining that level of service.

We monitor and alert on our RTT in our NMS. I've had half a dozen LSP reengineered over the MPLS core and I've had loops reprovisioned to make sure they were meeting their SLA. Once again, you're speaking far beyond your level of experience.

Let's go over how completely wrong you are again:
1) Doesn't understand RTT SLA
2) Doesn't understand the need for inbound QoS
3) Thinks QoS works on the Internet
4) Thinks an interruption in voice service is acceptable
5) Thinks voice is a "non-critical" service.

Anyhow, be gone with you, Mr Jr Network admin for whatever unfortunate company.

You are, a fucking dipshit. Bye.

Security Systems: http://www.zoneminder.com/
http://www.linuxjournal.com/article/8513
http://linas.org/linux/secure.html

Alarm Systems: http://www.voip-info.org/wiki/view/How+to+implement+an+alarm+system+with+Asterisk+and+a+webcam
http://www.linux-support.com/cms/diy-burglar-alarm-system/
http://www.linuxjournal.com/content/interfacing-disparate-systems

CCTV: http://www.tuxradar.com/content/build-your-own-surveillance-zoneminder
http://www.seattlesurveillance.com/

Building Automation: http://www.sciencedirect.com/science/article/pii/S092658050500097X
http://dl.acm.org/citation.cfm?id=1029022

Engineering Apps: http://loll.sourceforge.net/linux/links/Software_Applications/Science_-_Engineering/index.html

You get the idea I hope. So what can't run on Linux?

Asterisk is great for this, google "Asterisk Telemarketer Torture" sometime: http://www.voip-info.org/wiki/view/Asterisk+Telemarketer+Torture

If you must hide your IP address, you can use one of many Skype/Sip gateways. SIP to SIP to the gateway then Skype to Skype from the gateway. Since Skype does not work well in Linus, I use SIP instead. SIP is P-P too, a SIP call will reveal my IP to a SIP caller. A Skype caller will only see the gateway.

There are several gateways. IPPI.fr is only a representative example.

You can Skype me in France anytime. I have never been to France.

http://www.ippi.com/

I don't use this to hide my IP address. I use it with an ATA so calls ring my phone, even when I'm not online. With their speed dialer, I can make Skype calls without turning on the computer.

I can be called by Google Voice, an INUM number, SIP, Skype, or IPKall number and any will ring my SIP phone, provide voice mail, caller ID, etc.

Analog Telephone Adapter (ATA) http://www.voip-info.org/wiki/view/Linksys+PAP2T

Asterisk Reverse Hold! You call them up and then put them on hold through your Asterisk system. It parks the call and plays a loop telling them not to hang up and to dial a number when you come off hold. Then it makes your phone ring and connects the call. Genius! And if they want to leave you on hold for a month, that's fine. At least as long as you come in through their 800 number...

and let's be honest, in the corporate world no-one runs Asterisk

Wow. So the thousands of SIP compliant handsets and dozens of Asterisk servers I've deployed myself to private companies (many over $1 Million a year), not to mention several large cities and even an International airport doesn't count evidently.

You might want to let the solution providers that roll Asterisk based solutions out on a daily basis know that as well, despite the fact that it's an easy moneymaker for them and a cost saver for their customers

Seriously, how does statements like this get modded +5 insightful?

List of Open Source VOIP Software. Feel free to verify or modify the source to your liking. I think Ekiga sounds like a nice starting point, though I don't know how secure it is. It even supports calls to normal phones, so it seems I was wrong about that being a massive barrier.

Personally I don't care about trustworthiness or security in voice/video chat, since I've only ever used it for chatting to friends. For business use then being assured of confidentiality is more important of course.

Sure, for SIP to SIP, but not for SIP to PSTN, which is where most calls terminate. ( http://www.voip-info.org/wiki/view/Asterisk+DTMF )

I've used Gizmo5, Google Voice, Broadvoice and Junction Networks for US based calls.

There's an example of LCR using extensions.conf here:

http://www.voip-info.org/tiki-index.php?page=Asterisk%20least%20cost%20routing%20using%20broadvoice

It demonstrates how to test for carrier availability and failover if necessary.

I don't know about you. But as soon as I realize it is a call from an autodialer, I hangup.

One trick if you don't recognize the caller ID is to pick up the phone and just listen. If it's complete silence on the other end, it's an autodialer and it will hang up after five seconds or so. Bonus points if you play the "number not in service" tone -- download that from here and play the "ss-noservice" file.

You cannot use the Skype client as a SIP softphone without going via Skype themselves

No, but you could use Skype's Skype for SIP to do it, one of the SIP Skype gateway software combos out there or even a SIP service provider that I am too lazy to Google right now.

Being that Skype is providing this functionality directly, I honestly don't see the issue.

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

Have an Android phone? I think there are SIP clients for Android as well.

Don't have either, just want something that's a WIFI SIP phone? Check out VoipSupply.com. They have a WIFI phone section. I'd either go cheap with the QuickPhones GA-342 or spend a little more for the Hitachi IP3000.

You'll need a SIP VOIP service. Check out Voip-Info.

Of course, test before you commit to something! There are free "toll-free-only" sip providers, which will allow you to test to see if it really works.

--Pathway

Be interesting to get more detail on the phone server. Our local provider actually had a pretty good system and the price was right. Google Apps was very popular.

I run asterisk in a VPS. For $15 a month I get a private PBX that I can tap from anywhere in the world. Takes about 2-3 hours of work to get it up and running with extensions & voice-mail, if you're following the O'Rieley book. I pay $1/month for my phone number and 0.4 cents per minute incoming. Since I pay by minute, I don't have a limit on the number of calls. For about $9-15 a month you can get unlimited incoming and outgoing minutes, but it's usually limited to 2 simultaneous calls.

The practical limit is about 48 simultaneous calls or 96 active connections - above that and I would need to upgrade my server. For me, I get a business phone line, with the ability to do a conference call with more people than I care to talk to, for about $18/month including usage.

For people worried about backups etc, the VPS company has 4 locations with auto roll over and I get a backup stored on their servers and I can keep as many backups as I can be bothered with on my own. If they went out of business tomorrow, I could upload my image to any Parallels VPS provider and be back in business with an IP address change on the server and at my DID provider. Your mileage will of course vary depending on your VPS supplier.

If you want more info on the server itself, check out either Asterisk or Voip-Info.

andy zebrowitz

The majority of many telco's backbones are already converted to packet switched (IP) networks vs circuit switched (POTS) networks. Packet switching has a huge cost saving vs circuit switching. And yes, it works with any sort of data that is already being sent over the lines. We aren't talking Skype or SIP here, we are talking lower level type of hardware/interfacing.

What the real question is (the one the FCC is asking), what sort of measures should be taken to ensure that as the network goes full IP (and potentially to full VOIP) the quality of service isn't degraded. Do they need to demand a certain level of latency, lack of jitter, vocal quality (i.e. mandate a specific codec or bit rate).

VOIP at the same level of quality as what we consider POTS to have is quite doable, the reason it isn't done is that part of the cost savings in VOIP is the 'doing more with less' mantra it's currently being implemented under by most people. But is it necessary to mandate that quality and if so, how?

PS For specific information concerning your actual concern, you can start by reading this article.

Ok, I'll bite. Does this seem like a business opportunity to anyone?

Anyone who is worried about this kind of thing should already have an asterisk server which could do this for all phones, not just the cordless ones. And yes, its a huge business opportunity.

Yes. The "new user wizard" allows you to sign up for a VOIP-to-telephone account. You can also get incoming phone numbers.

The benefit to ekiga is that it uses a standard protocol (SIP) so there are literally hundreds of alternatives in various countries. I use Acanac here in Canada. I pay ~10 bucks a month for unlimited incoming/outgoing calling in North America- and that's a regular price, not a short-term bait-and-switch deal. Oh, and I use a cheap ATA box in my house so we can use a regular phone at home. Beat that Skype.

Interesting SIP sites for phone geeks:
http://www.voxalot.com/
http://www.sipbroker.com/
http://www.voip-info.org/

Yes you can. Unlike Skype, you're not bound to a central service provider with a dictating price model. You can choose one of the numerous SIP (or IAX, i don't know if Ekiga even supports it) service providers. Many of those that I've looked at are local providers, but with decent rates for long distance too. A lot of them are offering pre-paid plans, so it's easy and cheap to try, and you can later upgrade to a flatrate model if you so wish.

Or, you could even set up a gateway service yourself, if you want to afford the hardware and/or tinker with open software. Or why not a full-blown telephony server like Asterisk while you're at it?

True to the free software ideas, you have all the choice you want, the burden is just to review it all and to find something to fit your needs.

Currently, it's all rather open from a security viewpoint as well - but the technology is still young, and hey, it's probably not less secure than skype :)

Yes you can. Unlike Skype, you're not bound to a central service provider with a dictating price model. You can choose one of the numerous SIP (or IAX, i don't know if Ekiga even supports it) service providers. Many of those that I've looked at are local providers, but with decent rates for long distance too. A lot of them are offering pre-paid plans, so it's easy and cheap to try, and you can later upgrade to a flatrate model if you so wish.

Or, you could even set up a gateway service yourself, if you want to afford the hardware and/or tinker with open software. Or why not a full-blown telephony server like Asterisk while you're at it?

True to the free software ideas, you have all the choice you want, the burden is just to review it all and to find something to fit your needs.

Currently, it's all rather open from a security viewpoint as well - but the technology is still young, and hey, it's probably not less secure than skype :)

No you don't, you only need to meet the license conditions if you distribute code yourself. The GPL "is a distribution license, not a use license." From the same link "Acceptance of the GPL is NOT required to download and use Asterisk". From GNU: "If I only make copies of a GPL-covered program and run them, without distributing or conveying them to others, what does the license require of me?" "Nothing. The GPL does not place any conditions on this activity." "The GPL only requires that you make the source available to the same people that you distribute the software to. It only requires public "release" under the GPL if you're distributing the software to the general public."

Falcon

Check out the Asterisk software, and specifically the "chan_mobile" extension. It allows you to use a cellphone (with bluetooth) as an "incoming" channel for a phone system, or to use the cellphone as an extension on the phone system. I believe that chan_mobile is included by default in the newest (1.6.x) version of the asterisk software.

Asterisk has a fairly steep learning curve, so it will likely be a time consuming adventure to get it all working, but assuming the bluetooth on your phone supports it, it should allow you to do what you want. You will need to have a Linux computer that has bluetooth, runs the asterisk software, and you will also need an "FXS" port (can be a $15 internal card, or a $30 IP based one) that connects your home phones to the computer.

The voip-info.org site and the asterisk-users mailing list are both invaluable if you are just starting out with asterisk.

If diving into setting up your own asterisk server from scratch is too daunting, it may be easiest to try a prebuilt setup (such as Trixbox CE) and then following one of the guides for adding chan_mobile support to it. I can't personally say how involved this would be, since I've never used any of the pre-setup Asterisk systems.

Good luck!

VOIP is not illegal in India since 2008. See this press release for more details.

I do that on my home phone line (actually even simpler than that -- "Press 1 to continue in English"), and it works quite well.

Could you please provide a link that could explain how one would go about doing this themselves?

I'm using a Gumstix box running Asterisk with a SPA-3102 for the connectivity to the actual phone line proper, and a compact flash adapter (on the Gumstix) for storing voicemail. It also routes outgoing international calls to my SIP account with the Gizmo Project folks (much cheaper than AT&T, the local landline provider), and feeds incoming SIP calls into the house phone.

This was set up as a hobby project, so I wasn't going for a lowest-cost solution. If I were doing it again, I'd probably see about using my home router in place of the Gumstix box (I'm waiting for stable OpenWRT support for the WRT610N, with its USB host interface and 64MB of RAM -- more than powerful enough to run Asterisk in addition to its normal workload, with the voicemail storage and software that won't fit in 8MB flash kept on an attached external drive), or at least get one of the newer Gumstix motherboards with an FPU onboard to be able to receive and send faxes with iaxmodem (as the SpanDSP library it uses hasn't yet been ported to fixed-point, and so doesn't run acceptably on FPUless embedded hardware).

Once the hardware is set up, the actual Asterisk configuration is embarrassingly trivial, at least until I get around to implementing all the wishlist features I've been putting off. Should you decide to go the same route, drop me an email and I'd be glad to lend some assistance.

I do that on my home phone line (actually even simpler than that -- "Press 1 to continue in English"), and it works quite well.

Could you please provide a link that could explain how one would go about doing this themselves?

I'm using a Gumstix box running Asterisk with a SPA-3102 for the connectivity to the actual phone line proper, and a compact flash adapter (on the Gumstix) for storing voicemail. It also routes outgoing international calls to my SIP account with the Gizmo Project folks (much cheaper than AT&T, the local landline provider), and feeds incoming SIP calls into the house phone.

This was set up as a hobby project, so I wasn't going for a lowest-cost solution. If I were doing it again, I'd probably see about using my home router in place of the Gumstix box (I'm waiting for stable OpenWRT support for the WRT610N, with its USB host interface and 64MB of RAM -- more than powerful enough to run Asterisk in addition to its normal workload, with the voicemail storage and software that won't fit in 8MB flash kept on an attached external drive), or at least get one of the newer Gumstix motherboards with an FPU onboard to be able to receive and send faxes with iaxmodem (as the SpanDSP library it uses hasn't yet been ported to fixed-point, and so doesn't run acceptably on FPUless embedded hardware).

Once the hardware is set up, the actual Asterisk configuration is embarrassingly trivial, at least until I get around to implementing all the wishlist features I've been putting off. Should you decide to go the same route, drop me an email and I'd be glad to lend some assistance.

There's a reason DISA exists. Yes you need to take steps to secure it, but it serves a useful purpose.

You know, all I need to know is the IP of my machine to wake it up remotely.

Behind a NAT, all I need to do is set a port forwarding rule.

No, you don't just need the IP -- you really do need the MAC address (at least for earlier versions of WoL; newer ones have some alternate keying mechanisms). If you're on the same network segment, the software you're using will look it up for you; if you aren't, it can't.

And you want Joe and Jane Average to have to set up port forwarding rules for the voip/IM/whatever providers they're using? Yeah, right.

You can certainly use MAC-based wake up packets even when not on the same LAN. There is such a thing as a static ARP entry you know.

The lack of an ARP table entry isn't the problem.

Your first line of defense, blocking people who block caller-ID, is redundant. The phone company offers this. Calls will never even get to your house if you enable it. No need to power up for the spam phone calls.

The phone company offers that... and charges for it. Fuck 'em.

You can also handle blacklisting at the phone company end, and much more powerfully. (This will cost you, though).

Not as powerful as what I can do locally.

You do all that to keep people from spamming you? Wow, I'd hate to be a family member who had to legitimately call you.

They're whitelisted... the ones I actually want to talk to, anyhow.

Voicemail. Great. So if I call John and he's not there, I get his voicemail. I have to call again and press 3 for Sarah, to find out she's not there. I call again and press 4 to find out Bob is there, thankfully. I just wanted to know what time the BBQ was, fucker.

Fine theory. Moot in practice. We don't do family outings -- we're antisocial like that. Plus, if you were someone we cared about, you'd be on the whitelist to get the forward-to-cell-phone option instead of going to voicemail. (Incoming is POTS [for faxing support], outgoing is cheap VoIP, so we can do that with only one POTS line).

The only person who ever calls with the intent to talk to whichever arbitrary person happens to be here is my father-in-law, and if I ever care enough I'll write a custom rule for him.

Telemarketers routinely spoof caller ID. You're blocking and blacklisting is ineffective.

Nope -- it works just fine; we got most of our spam from the same 3 caller IDs. Further, most telemarketers won't bother to go through the who-are-you-calling-for? menu, so even for those that aren't blacklisted the phones never ring.

Evidence against them? Now you're talking out of your ass. I defy you to litigate against any of them.

I file FCC complaints on a regular basis. Having actual records to back up something I'm swearing to on the paperwork gives me warm-and-fuzzies.

Faxes, eh? Who the hell still receives faxes? And why touch sourceforge when a simple Dell printer/copier/fax supports pdf conversion and e-mailing/forwarding/blocking/whatever of faxes out of the box?

Our Dell printer/copier/fax is out in the living room, and that part of the house isn't wired. Faxing is too sensitive to jitter to reliably connect it over voip with one of the SPAs, the model we own doesn't do the PDF thing and, ya know, I might actually do this shit because I think it's fun. Beats wasting my time playing World of Warcraft or something.

Oh -- and as far as "who the hell receives faxes" -- I did (in a work-related function) until I changed jobs a few months ago.

I hope I'm wrong, otherwise we're going to have pages of posts helping a telemarketer do his or her job :)

That's a risk that we have to take. But don't forget that there are other people reading this too.

Just take a look at the Asterisk Telemarketer Torture if you have problems with Telemarketers.

And even if this is a single telemarketer, the catch is to make it impossibly ineconomical to run a telemarketing company. Many that are employed may have had to chose between being unemployed or be a telemarketer.

What keeps me with Skype is that I can have US telephone number. So no matter where I am my friends and family can call me.
        If there was another service which allowed me to have a US telephone number for incoming calls and let me call any other POTS number I'd use it.

Ummmm, one of of any number of several hundred VOIP providers (or Vonage) with a PC softphone, give you exactly that. In fact, I'm pretty sure it's possible to get free DID's (phone numbers) in major cities. Even here in Canada, LES.NET gives you local VOIP numbers for $8.88/mo (unlimited incoming) and 1.5c/minute North American outgoing. It's a very generic (and open) way to do things. Skype is a just one proprietarized VOIP solution, that happens to be a bit easier to set up.

VoIP/SIP is open.
You only need a client and an account with any of the free SIP providers. Or you setup asterisk (or another free PBX software) and become your own provider.

The problem with SIP is that few people actually use it whereas skype is everywhere.

There are quite a number of alternatives based on the open SIP protocol. Have a look at the list: http://www.voip-info.org/wiki-Open+Source+VOIP+Software

Does VOIP really use TCP usually? Shouldn't it be using UDP?...

Yeh... the RTP protocol to transmit voice is UDP. This causes a lot of the problems with sip and nat/firewalls. SIP is also generally UDP, though it can use TCP. SIP is used to setup the session, do the auth, figure out where the streams need to go and how to translate addresses and even do billing stuff, etc, then it lets RTP do the grunt work of transporting the media. MGCP and other VoIP protocols also use RTP for the transport.

What most people dont understand about VoIP (even just SIP) HAS filled many books, and websites ;)

Tm

Does VOIP really use TCP usually? Shouldn't it be using UDP?...

Yeh... the RTP protocol to transmit voice is UDP. This causes a lot of the problems with sip and nat/firewalls. SIP is also generally UDP, though it can use TCP. SIP is used to setup the session, do the auth, figure out where the streams need to go and how to translate addresses and even do billing stuff, etc, then it lets RTP do the grunt work of transporting the media. MGCP and other VoIP protocols also use RTP for the transport.

What most people dont understand about VoIP (even just SIP) HAS filled many books, and websites ;)

Tm

You are lowballing the figure somewhat, but are more or less correct. VOIP is not bandwidth-intensive. It is, however, extremely vulnerably to latency and jitter. Considering the telcos will use arguments such as this patently absurd one to, presumably, slow down other vendors implementations of VOIP and provide facsimile arguments for deep stateful packet-inspection, VOIP is a concern as it is an area the traditional telcos will branch into and promptly begin to shoot each other, and the consumer, in the foot.

you can save bandwidth if you set up your cam to only send pictures when your motion detection software trips an alarm.

http://www.voip-info.org/wiki/view/How+to+implement+an+alarm+system+with+Asterisk+and+a+webcam

Another one that seems to be gaining a lot of traction and backed by a lot of online help, is PBX in a Flash - it is created and supported by Nerd Vittles.

They have loads of info on their site, including the obvious requests, like how to setup a new system quick & easy, what phones to look for, what hardware cards or peripherals to use to interface with POTS lines, as well as a list of VoIP providers that they have reviewed and recommend (or don't,) which you can read at Providers - The Best of Nerd Vittles

They also have some stuff you might not think of, such as setting up services on your Asterisk box to tell you the weather report for a zipcode you enter on your phone's keypad, or "MailCall" which allows you to get your e-mail via the phone, or a telephone reminder system... there are lots of options, and you can find out about most of them at Applications - The Best of Nerd Vittles

There are also forums for lots of this stuff:
Nerd Vittles Forum
Voxilla VoIP Forum
PBX in a Flash Script Site

And, of course, the venerable VoIP Wiki at voip-info.org

Encrypt your streams, encrypt your data, encrypt your voice.

Few, but more than you would think, devices and providers understand Secure Real-time Transport Protocol (SRTP) for SIP channels.

It's important that we get this working in the free software world as well:
http://www.voip-info.org/wiki/view/Asterisk+encryption

Blowfish or not, any encryption is better than no encryption.

I've done a lot of work with VOIP both for myself and for a few clients. In a business situation look at setting up an Asterisk server (very easy with TrixBox). For home use there are lots of similar providers. You can look up some information at http://www.voip-info.org/ and the forums at http://www.trixbox.org./ I've had good experience with Vitelity and VoicePulse. I use their services to connect to my Asterisk box, so that doesn't directly transfer over. Based on what I've seen a few of my friends go through I can recommend that you NOT go with SunRocket.

andy zebrowitz

There are many, many alternatives out there to Vonage. One list is here: http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential

I personally have been with a small provider for the last 2 years that provides excellent service, personal support (no hold queues) and is actually cheaper than Vonage. So yes, Vonage isn't the only VoIP game in town.

SIP with WiFi would make it totally worth it. There are many VoIP providers which support the open SIP protocol. I use Broadvoice using their BYOD plan with a VoIP hard phone and love it.

SIP with WiFi would make it totally worth it. There are many VoIP providers which support the open SIP protocol. I use Broadvoice using their BYOD plan with a VoIP hard phone and love it.

Set up a small little Asterisk server in your place and transfer them into hell http://www.voip-info.org/wiki/view/Asterisk+Telema rketer+Torture

It looks like a firm in Germany already offers a AES-256 bit encrypted mobile and POTS phone, as well as a softphone. Although their hard phones aren't cheap, the softphone is free to give to your contacts. http://www.cryptophone.de They alse include source code for "full independent review" with their products.

Similarly, Phil Zimmermann, the creator of PGP has released his Zphone to make encrypted VoIP calls. Also, the Asterisk project offers an encrypted IAX channel.