similar to: IAX Firefly config

Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com jeromy@voipempire.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/e668136a/attachment.htm

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

I did get it to work, and can place and receive calls through the Firefly network via *. Compared to iaxtel or FWD, there is a significantly higher amount of latency, but it is workable. For some reason, this needed to be the last entry in my iax.conf or it would try to authenticate with a different user ID when receiving calls (and obviously would fail. Relevant section from my iax.conf:

I found version 2.0.0 of Firefly on the Freshtel site, but it only has the network setup options for the Freshtel network, despite the final statement on the page http://www.freshtel.net/firefly/download/ that says: ----------------- Standalone SIP / IAX mode: If you want to use Firefly on our network (with your own voicemail etc.) you will need to register a Firefly number. However, you can

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

Hello, I am having horrible echo problems when using the FireFly client on both the caller and callee sides of the call. When I use another IAX soft client like IAXcomm or IAXPhone I do not have the same echo problems. Has anyone else experienced this and do you know what might be the problem? Thanks, dj -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Ganeral --> Language correct from "portugese" to "portuguese". Kind regards, Miguel Date: Tue, 5 Oct 2004 09:47:08 +0100 (BST) From: gARetH baBB <hick.asterisk@gink.org> Subject: Re: [Asterisk-Users] Firefly 1.9.5 released To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Just a quick announcement for Firefly users that Firefly 1.9.5 is out. Mainly just a bug fix release as we get ready for Firefly 2.0. One notable feature added is DTMF via SIP INFO. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe is the URL As always, send me any bugs, features or suggestions. -Adam

I have an asterisk server up and running, using Firefly in IAX mode works great, even with Firefly behind a NAT (as expected, since IAX works really well with NAT). Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from behind the NAT, and I can't seem to get there. At this point, the phone will successfully register with Asterisk, and the Asterisk qualify messages get

Hello, In my sip.conf I have: ;Register and forward FWD numbers to internal extensions register => FWDNUMBER:PASSWORD@fwd.pulver.com/9500 Which should register Asterisk at FWD and then when any calls are made to FWDNUMBER those calls should be forwarded to extension 9500 as specified in the extensions.conf. What I am getting is it is trying to dial the 9500 (IAX Firefly) client twice when

Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid="Foo" <100> qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar <10>

There's a new firefly release out for those who are using firefly with your lovely asterisk / SIP server. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe the main changes are improved GUI fixes (mouse wheel works now :) ), few url parsing fixes, mic volume control and improved compatibility with SIP servers (namely SER). Send me all bugs, problems and suggestions (even

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hello all, i have a strange problem....i am running SER in front of asterisk, and am testing softphones. x-lite works fine...i can dial, hang up, DTMF, all good. Firefly looks really cool and i'm very impressed with the IM-like interface and the skinning ability, but something strange is happening...when i call from the firefly and run something on the server and press hangup on the client,

Does anyone know if there's a way to get the FireFly presence stuff to work with Asterisk? Details: 1. I would like to be able to see other people's status in my firefly client. This one I think may be able to be done without asterisk. I.E. maybe the clients need to be registered against the firefly network and just not send calls through? 2. I would like (if possible) to use the

I'm having a problem configuring asterisk to send incoming calls to Firefly. I can make outgoing calls from firefly through asterisk without any problems at all. The firefly client does this when it's on the same IP subnet without a firewall, or from a NAT'd environment. Can anyone tell me where I'm going wrong? Here is output from iax2 show peers: Name/Username Host

Hi!. I am testing firefly and I can say it's a great program, but I have a problem. When I use Sip and I activate the "canreinvite" option in Asterisk, I can't hear anything. My network is the following: -Two Firefly clients with SIP. Each firefly is in different networks behind NAT. -One Asterisk server with a public IP. First, I tested my network with canreinvite=no.

With all the talk of SRV support in Asterisk, I thought I'd add support in Firefly so enjoy. Thanks to Olle for helping me with it, explaining the wonderful world of SIP and SRV to me. There's also an option to disable it (seems to take quite a few DNS lookups for SRV) - warning Duane may hunt you down if you do disable it though :) I've also added support for SIP via TCP and the

Hi, when I press reject on Firefly, another end hears congestion, but I'm still listening to ringing... Also how you determine which version of Firefly you have? I also don't understand licence completely and would like to get this answer - can I distribute Firefly to commercial users if I don't charge anything ? Regards, Rob.

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using