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Very Cool =)

If they're ever stuck for which project they should contribute to, send em on over to have a looksie at asterisk, The Wishlists and bounties on the asterisk wiki, and the asterisk bug tracker! [/shameless plug]

Seriously though....

Their codec is an open-source standard called iLBC.

You can read more about it on the VoIP Info Wiki.

Asterisk sucks for newbies. Sorry. http://www.voip-info.org/ (site seems to be currently down) has a lot of information, but it's hard to hash out as a newbie. Took me a half a day just to get internal extensions to call each other. Once you "get" it, it's pretty easy.

When I have time, I'm planning on writing a tutorial.

The site mentioned above has a lot of examples listed. Read them, and it will help you understand.

More information than you'll ever want on Asterisk can be found here.

There are services out there that will take incoming sip/iax connections and route to a POTS #. Very cheap alternatives too, $.02/minutes to UK from anywhere in the world is an example rate. The asterisk wiki lists PTSN to/from services

But another reply in this thread is also right, colocate an asterisk box to have it outdial the local calling area.

http://www.voip-info.org/wiki-SJphone
is what you may like.. (turn off skin and it's better :) )

/msg nickserv register mypassword

/msg nickserv identify mypassword
then /join #asterisk

Come on in and say hi!
Some links

The Wiki
The Asterisk Documentation Project
Andy's Getting Started With Asterisk Guide (it's written for a old version of asterisk, but still useful)
ManxPower's site
For some advanced examples see John Todd's site
Also read all files in ./asterisk/doc after you download Asterisk.

more links (look at the "Unnoficial Links")

Mod me up! :)...

There are un-locked linksys Sipura's out there, look for -NA on the model # PAP2-NA and RT31P2-NA are the two models available according to VoIP Wiki

Check
this out

GSM typically takes about 13k per/call. Not to mention there are other protocols besides SIP. For example, IAX2 is wonderful. You can also "trunk" the calls to lower the TCP/IP overhead.

G.711 (ULAW) typically takes about 64Kbps, which would be comparable single channel on a DS1/T1. With GSM, I can now fit over four calls in that same channel. How is this worse? I run SIP everyday, and did does work....

http://www.telephreak.org [VoIP hackers]

Welp, as many have pointed out ANI != CID. I'm a big, big fan of VoIP and is anything but knew. Whoopy. If you're interested in what you can do with VoIP and asterisk, check out: http://www.telephreak.org and of course a wonderful reference is http://www.voip-info.org . Normal DID lines usually aren't lax enought to let outbound CID go through. However, DS1, etc. circuits, it's not completely uncommon. I think it's sort of cool the Nuphone does this (though, I will have to check it out for myself). When a call via SIP, for example, is made, the CID information is sent - just as normal data. So, it shouldn't be terribly supprising that if your machine is sending the data, you can alter the outbound data. This isn't exactly something ground breaking with asterisk.

For more info on SIP and NAT issues, see this and this and other related pages at voip-info.org. 1:1 nat is atypical. Most people use a much more simplistic nat (cisco calls it PAT) where you only have 1 real external IP address. I'm well aware of the tricks asterisk uses, but again, this is atypical of sip clients.

For more info on SIP and NAT issues, see this and this and other related pages at voip-info.org. 1:1 nat is atypical. Most people use a much more simplistic nat (cisco calls it PAT) where you only have 1 real external IP address. I'm well aware of the tricks asterisk uses, but again, this is atypical of sip clients.

some links:

http://www.myjabber.net/
http://www.voip-info.org/wiki-jabber
http://groups.yahoo.com/group/Jabber-VoIP_Client/
http://www.jabber.org/pipermail/standards-jig/2003 -January/002541.html

The beauty of Jabber/XMPP tho is that there is the possibility of gatewaying to things such as SIP, so you can have the best of both worlds while maintaining a single protocol on the Jabber/XMPP side, so there is no need to worry too much about what will become the dominant voice protocol since there is the possibility of interoperability.

Not right now. SRTP (secure realtime transport protocol) is the protocol you're looking for. Very few clients are capable of using it.

There are the "telezapper" products that may help, but they are kinda lame. I use a linux-based asterisk phone system. If your caller ID is not on the (mysql based) white list, you need to navigate the menu. Numbers I dial are automatically added to the white list, and I also have a web-based management tool for it.

There is also a Telemarketer Torture script for asterisk someone came up with... :-)

Set up Asterisk to try an EnumLookup first, then fall back to NuFone or your home landline using a $16 X100P WinModem from DigitNetworks.

Get all your friends to register their phone numbers with E164.org too, it's a free ENUM service that also verifies people's numbers.

Then if you're really feeling groovy, help a local Community Wireless Network deploy an 802.11a backbone with 11g hotspots all over the place ;) Works great with Asterisk and serexpress. :)

A handy list of VOIP service providers also check out the site for other usefull voip information.

... and try asterisk. This program truely is amazing. Check out VoIP Info for a big wiki all about voip and asterisk. Next!

There's a good article by John Todd at o'Reilly here.

Here's a Guide to Asterisk.

There's also a Wiki

Most of my experience with * is via trial-and-error, reading the newsgroup postings, posing questions for help etc a-la the normal open source way of doing things, and it should be noted that the article says "Zealous fans of the terrific open source PBX-plus software develop easy-to-follow installation and configuration menus". There exists some documentation on the web, with war stories and sample configuration. But, as noted in the Asterisk mailing list, the docs can be somewhat sparse and technically oriented, especially for users just wanting to "try it out".

I basically think the article about Asterisk should be read as "Zealous fans... need to develop easy-to-follow installation and configuration menus, and Asterisk takes off big-time". This, to me, is a key point in making Asterisk viable to the masses. People (the users anyway) tend to expect a PBX system to be something of a "plug-and-play" type of thing, but without concrete documentation, helpfiles etc it would be hard for the n00b installer to get things working in a short amount of time.

Why didn't you just use Asterisk ?

Asterisk is an AWSOME PBX system that doesn't get mentioned enough on /.

It's supported features are equivalent to a PBX costing several thousand bucks. Including support for VOIP and T-1(E-1)'s

Some of the other features include Voicemail, Conference calling, Caller ID, an Auto Attendant (press 1 for sales, 2 for support,), Call Queuing (for call centers), Call Detail Records, more

The documentation is a little sparse but they are currently working on the
Asterisk Handbook Project (warning PDF).

I also found the Getting Started With Asterisk Guide by Andy Powell very useful.

And there's a IRC channel #Asterisk on FreeNode (try irc.debian.org) (argh /. mangles the irc link)

There's some more links to support pages including a Wiki at the bottom of this page