iaxclient-devel

[Iaxclient-devel] Introducing Firefly

[Adam H. <ad...@te...>]

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/

Re: [Iaxclient-devel] Introducing Firefly

[Dan <dt...@fx...>]

Hi,

----- Original Message ----- 
From: Adam Hart

> I believe you can connect using a standard asterisk box but you'll miss
out 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 IAX2
clients (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 it
asks 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

Re: [Iaxclient-devel] Introducing Firefly

[Adam H. <ad...@te...>]

From: "Dan" <dt...@fx...>
> Hi,
>
> ----- Original Message ----- 
> From: Adam Hart
>
> > I believe you can connect using a standard asterisk box but you'll miss
> out 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.

It should work, others have reported success. In networks, ensure you select
IAX. Firefly protocol runs over FRAME_TEXT in the iax2 protocol but only if
you select Firefly as the type of network.
Just a quick note, whenever I say IAX i mean iax2, iax1 isn't used at all.
>
> > The network runs on iLBC so unforunately it won't work with most IAX2
> clients (unless you get * to translate)
> There is any specific/technical  reason not to use GSM as another possible
> codec?

This was mainly a quality decision, only allowing iLBC ensured compability
and quality. Beats asking 'which codec are you using?', 'do you have iLBC
enabled?'. Also native transfers don't work between different codecs.

>
> > If you don't wish to connect to the firefly network, click cancel when
it
> asks 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.

dialing a number without an @ suffix will use your internal network. eg
12345678 (To make a network your internal network, tick route internal calls
for that network - ditto for external)
dialing a number with a + at the start (and no @ suffix) will use your
external network. eg +413123135
you can dial a number with @<network name> to explicitly use a certain
network, eg if you have a network named iaxtel and you dial 1234@iaxtel, it
will use your iaxtel network details to make the call.
Otherwise, urls will work eg iax://123@my*server.com/123 - next release will
add more support for urls. - push F2 to enter a url


Additional - new release out

Music on Hold is now working, play mp3s to other party while they are on
hold :)
you can now store urls in your contacts (used to limit to 20 chars, ta JTodd
for finding that)