[Asterisk-Users] Introducing Firefly

[Andy Powell]

Hi,

I downloaded this the other day and finally got it to stop crashing. It appears that any response from asterisk
that implies an error (for example dialing a non-existant number, using the wrong password, selecting a codec
that you've configured a local * not to use etc) resulted in a crash. I've only tested the IAX proto not sip or your
own network. running XP with uptodate patches on a local lan.

When it works it works really well, although I don;t particularly like in initial beep and end beep when i make 
a call (I haven't played with all the options so it may be that I can turn this off).. sound quality is good. All in all
a nice little app. Are you planning on allowing other people to run your server side (like Jabber does) in their
environments?

If you need any further debugging info on the crashes, let me know...

HTH

Andy


*********** REPLY SEPARATOR  ***********

On 28/01/2004 at 12:11 Adam Hart wrote:

>After many months of development, I'm pleased to announced Firefly - an
>IAX soft phone and network.
>
>The firefly softphone - free, runs under windows, allows connection to
>multiple networks, skinable interface, connection to firefly network, IAX2
>protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw,
>GSM. - contact lists, selectable ringtones.
>
>download from here - http://www.virbiage.com/firefly/
>
>The firefly network - also free, runs on an enhanced version of IAX2
>(simply uses IAX2 text messages for customised part), voicemail, text
>messaging, online presence, ability to indicate status (available, away,
>NA). I believe you can connect using a standard asterisk box but you'll
>miss out on the extended features. The network runs on iLBC so
>unforunately it won't work with most IAX2 clients (unless you get * to
>translate)
>
>Thousands of people have used it but it's still regarded in beta, as we
>are still in heavy development (so expect a few bugs). It doesn't use
>iaxcomm as we needed our own framework to support sip, including our own
>jitterbuffer. If you don't wish to connect to the firefly network, click
>cancel when it asks you.
>
>Coming soon features
>SIP - in alpha, few bugs outstanding
>music onhold - playing mp3s while the other party is onhold
>fast audio - will reduce the latency by 40-50ms
>speex - (if anyone wants it?)
>
>Feel free to contact me on or off the list to report bugs and suggestions.
>I'll post everytime we release a new version (probably every week),
>including fixed bugs and new features
>
>Our website is http://www.virbiage.com/
>
>
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