Domain: fwdnet.net
Stories and comments across the archive that link to fwdnet.net.
Comments · 7
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Not the first time it was done
"it's enough to be able to make a phone call in a Web page for the first time."
Not true. FWD (FreeWorldDialup) had an activex implementation that allowed you to do the same. Here it is http://account.fwdnet.net/fwdtalk/
I have used it before and it works fine.
Another thing - this wengovisio looks suspiciously similar to meebo styling... -
Re:Hmm
This is already being done. Checkout Free world dialup: http://account.fwdnet.net/index.php
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Corrected links...The above article forgets to link to the most important and popular Asterisk site. Specifically, voip-info - a wiki where you'll find documentation on everything you'd like to know about Asterisk and various ways of administering it.
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!
--Rob -
The article fails to link to....the most important and popular Asterisk site. Specifically, voip-info - a wiki where you'll find documentation on everything you'd like to know about Asterisk and various ways of administering it.
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!
--Rob -
Re:From the article
Any US Domestic calls for free?! It's not what the FAQ says... maybe I should sign up?
http://www.fwdnet.net/fom/serve/cache/28.html
"Is this a PC-to-Phone or Phone-to-phone service?"
No, We are an Internet Telephony Network Service. We only provide connections between IP Phones, PC Phones, and IP Services. We do not provide any toll bypass. -
How?
This is all nice and all but how the hell are they going to regulate this exactly? Sure it might be easy to target companies like Vonage but what do you do with all the free services out there like Skype or Free World Dialup?
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Re:But who has the directory?
Replying to myself.. Too bad you can't edit your own posts.. But then again, the goatse guy might crop up a lot!
;-)
Free World Dialup is already running a directory that voip services can hook up to for free. For example "Dial 1010333number to reach iConnectHere subscribers" and "Dial **478number to reach any iptel subscriber."
Not as good as using DNS (you could just dial number@iptel.com if that were implemented correctly) or a global standard so that voip services would just have their own LD/country code for example (though the PSTN telco's are actually moving toward SIP rather than the other way around - using enum.)