similar to: FireFly - no sound after first call

Displaying 20 results from an estimated 40000 matches similar to: "FireFly - no sound after first call"

2004 Apr 21
0
FWD <> SIP <> Asterisk <> IAX <> Firefly
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
2004 Aug 10
1
Firefly and *... Argh!
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
2006 Feb 09
0
Firefly & iaxLite dont stop ringing when answering incoming call
Hi Everyone, I've got a weird problem with both Firefly & iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read
2004 Feb 01
1
Configuring Firefly Network in *
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:
2005 Jan 12
1
linphone -> NAT -> * -> NAT -> firefly woes.
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>
2005 Jan 26
0
Firefly reject problem - it just keeps ringing
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.
2005 Jan 18
1
Asterisk and IAX softphone (firefly) problem/question
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
2005 Jun 09
1
IAX2 Max Retries dropped calls Firefly
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
2005 Oct 04
1
Firefly 2 third-party version?
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
2006 Oct 10
0
Cubix / Firefly softphone and Asterisk
Hi All Has anyone used Cubix / Firefly successfully with Asterisk? When someone calls a Cubix softphone, Cubix never seems to answer the call correctly. The other person just hears ringing even though it has been answered. I am using IAX as the SIP support doesn't seem to 100% either. Idefisk works 100% on the same setup. Kind Regards Garth
2004 Oct 05
0
Re: Firefly 1.9.5 released (gARetH baBB)
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:
2004 Oct 05
1
Firefly 1.9.5 released
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
2004 Jun 16
3
X-Lite/Firefly behind NAT connecting to Asterisk not receiving RTP
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
2004 Apr 03
2
FireFly Problem
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
2004 Jun 28
2
New Firefly release - 1.9.3
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
2005 May 24
0
IAX Firefly config
hello all... newbie question: I have FireFly setup on my laptop and I would like to test this out using IAX in this scenario: FireFly Softphone > Asterisk > TDM Gateway i do not wish to use this on the firefly network, but simply within my own "3rd party" network as the website and setup of FireFly defines it... does anyone have a sample iax.conf and extensions.conf i
2005 Jul 25
0
slightly OT: firefly won't hang up!
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,
2004 Dec 06
0
Firefly prescence + Asterisk
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
2004 Apr 02
1
Firefly Client can't receive incoming calls
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
2004 Nov 23
1
Firefly:Canreinvite problem
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.