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/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040128/1a8ff0ed/attachment.htm
Hi, ----- Original Message ----- From: Adam Hart> I believe you can connect using a standard asterisk box but you'll missout on the extended features. It is a standard IAX client or a softphone using a modified/non-standard version of IAX2? I have tried to register with my own * box without success.> The network runs on iLBC so unforunately it won't work with most IAX2clients (unless you get * to translate) There is any specific/technical reason not to use GSM as another possible codec?> If you don't wish to connect to the firefly network, click cancel when itasks you. When you have setup multiple servers and try to dial a number which of the servers is used to place that call? I have defined just one (my box) and then tried to call an existing extension, but it doesn't work. Best regards, Dan
Nice!! Have just tried it a bit, seems cool... Congrats!!! Will test it against my * box and will provide some feedback. Thanks! Sam\\\ ----- Original Message ----- From: Adam Hart To: iaxclient-devel@lists.sourceforge.net ; asterisk-users@lists.digium.com Sent: Tuesday, January 27, 2004 7:11 PM Subject: [Asterisk-Users] Introducing Firefly 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/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040129/b9c8c273/attachment.htm
When will the IP phone be available and would you have an idea about the price? --- Samuel Jimenez <jimenezchava@racsa.co.cr> wrote:> Nice!! > Have just tried it a bit, seems cool... Congrats!!! > Will test it against my * box and will provide some feedback. > Thanks! > > Sam\\\ >====Chris Albertson Home: 310-376-1029 chrisalbertson90278@yahoo.com Cell: 310-990-7550 Office: 310-336-5189 Christopher.J.Albertson@aero.org KG6OMK __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/
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/ > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users