similar to: Firefly as Asterisk SIP client - qualify works ?

Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. bkw

Hi, I spot weird behaviour of latest Firefly 3rd party on my laptop. Sometimes it comes to state that it won't start (hangs on "Initializing" ) and it again works after system restart... Didn't yet figured out how to recreate it..... Any similar experience ? Also - how can I force Firefly to make outgoing calls (also sip or iax calls) through Asterisk ? I'd like to

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

Hi, I'm looking for adressbook that could easily integrate into Asterisk, so it should: - have agi script to search for caller id name - have fields for notes on previous contacts (for CRM spawning of FOP) - have web interface to edit entries easily ... Any advice, pointers ? What is your favourite address book to use with Asterisk ? Regards, Rob.

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

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

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

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

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

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>

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

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 -

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

G'Day, I have a bit of FireFly problem that hopefully someone has seen before. What happens is if I make to or receive a call from the FireFly network the call will connect successfully. However, around 10 seconds after I answer the call I am disconnected. The weird thing is same thing happens if I make a call. I've had a look at the * console and I can't see that my * PBX drops

Using Firefly 1.9.5 (thirdparty) on Win2k Using Asterisk CVS HEAD 20041117 (also tried with 20040806 and 200410-something) IAX2, no NAT. Firefly->Asterisk audio works, but I can't hear anything from the other side. Using GSM codec, also tried ulaw. Any ideas? -A. relevant bits of iax.conf: [andrew-bt] type=peer host=dynamic trunk=no [andrew-bt] type=user context=fxs secret=12345

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

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.

Hello List, On my cvs-head (29-Dec) asterisk, my sip phones can use *78 for DND and *79 to turn it off. With my firefly (iax) client, I am getting the following errors if I try these feature codes: Jan 10 13:26:18 NOTICE[11702]: chan_iax2.c:5792 socket_read: Rejected connect attempt from xx.xxx.xxx.xxx, request '*78@internal' does not exist Jan 10 13:26:23 NOTICE[11702]:

Is there a bug in Firefly (3rdparty) wherein it does not show caller ID? I am using SetCIDNum(12345) before I dial my firefly (IAX2) phone... no caller ID. CallerID is passed properly to other clients. -A.