similar to: FireFly Problems

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

My name is Camelia Botez and I'm linux system admin. I try to configure samba server usind ldap authentication and different sambaHomePath from login home directory. For ex. A user user_test has his home (login) directory /home/user_test and I want , using samba map on windows to have /srv/user_test home directory for this user. We included in ldap schema 2 attributes sambaHomeDrive (H: ) And

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

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, 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

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

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 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]:

Hi, I'm curious if anyone is using firefly as SIP client and if qualify=yes works for it. In my case Asterisk just keeps retransmitting of OPTION SIP message and Firefly doesn't seem to respond - but all this could be just wrong settings ? Anyone has working SIP configuration with qualify ? Thanks , Rob.

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>

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

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.

I have installed Firefly, but I cannot figure out how to use it with Asterisk. I have seen the settings in Asterisk, but I do not see any settings in Firefly. I need a light .... bye Ronald

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 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

Is there a way to register with or call the firefly network from an Asterisk server. It would be pretty cool if you could gateway calls onto it. Have a nice day, -- Martijn van Oosterhout

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