Displaying 20 results from an estimated 11000 matches similar to: "slightly OT: firefly won't hang up!"
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 Jan 29
1
re: help with voicepulse connect IAX2
hello,
after playing with an asterisk configuration for voip for a few weeks i'm
trying to get outbound dialing with voicepulse going - i've cut down the
asterisk to a very minimal install (1 SIP client) to try to localize the
problem. The SIP client works fine (SIP and * on the same NAT) and could
access the demo from samples before i removed it, and can call itself - so
i am
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 Jul 20
3
Firefly 3rd party - it hangs on "Initialising" and exits with error
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
2004 Dec 02
1
firefly and caller id
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.
2005 Jul 02
3
call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is
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 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
2004 Apr 29
0
Queues and IAX2
I'm running Asterisk CVS-04/28/04-13:22:35 (fairly current)
Today when I setup queues for the first time (with one member in my
default queue), I got some really strange behaviour, aside from my
hysterical laughing after hearing the default MOH =)
I only have one SIP hardphone I'm testing with right now, so I tested
using DIAX, Firefly(IAX) and XLite(SIP). My hardphone is an analog
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
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
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>
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 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
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.