Displaying 20 results from an estimated 6000 matches similar to: "Calling the firefly network?"
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
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
2005 Feb 25
3
How does the g.729 registration program work?
I'm asking because I'm planning to install multiple machines from the
same image and I need to know what file(s) I need to backup/restore to
make sure I don't lose my licences in the process. The only options I
can think of are:
- There's a config file, though I've seen no mention of it
- The actual binary shared library is modified
- The system contacts Digium every time you
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
2005 Jan 27
1
Digium and Intel Chipset compatability
Hi,
I'm going to be setting up some machines with 4 port E1 cards from
Digium and I'm being told that TE410 is incompatable with several Intel
chipsets including the ones in a lot of Dell server systems.
Is this true? I can't find any confirmed details on the mailing list
about it. Also, the email seems to imply that the TE405P will be fine,
though it doesn't say that explicitly.
2005 Mar 09
1
Providing a dialtone
Hi,
I see applications for signalling busy, congested, ringing, progress
etc, which I understand can be provided either in or out of band. But
all I want to do is generate a dialtone. This obviously can only be
done in band.
There is code for generating the tones when you have a physical line,
like the alsa channel, or a zap channel. But I'm just thinking of if
they've selected an option
2005 Mar 21
2
Permission issue with outgoing calling
I have created a call file which has been moved into the outgoing directory.
However the log file displays the following message: Unable to open
/var/spool/asterisk/outgoing/1.call: Permission denied, deleting
I have executed chmod 777 1.call on the file prior to moving it to the
outgoing directory but is there something else I need to do before the file
can be used by Asterisk?
Any help
2005 Feb 24
4
What is an E400P-SS7??
Hi,
Is this card the same as the T410P, after all, it's made by Digium.
There's one prior reference on the mailint list[1] but it didn't answer
the question.
There was also an SS7 status report[2] last June but it's doesn't seem to
have lead anywhere either. There was post saying an SS7 release was
immenent last September[3], but then silence.
Any info anyone would like to
2004 Jun 16
1
asterisk/netmeeting works, asterisk/ohphone doesn't?
I've been banging my head on this one for a few days and am quite stuck.
I've got a gatekeeper running and everything works there. Netmeeting works
calling other netmeeting clients. Netmeeting calling asterisk connects, but
netmeeting can't generate the signals to make the demo do anything other
than talk.
But connection from ohphone always disconnects straight away. I can't seem
2004 Nov 17
2
Firefly 1.9.5 and 20041117 CVS HEAD -- IAX2 one way audio
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
2004 Jun 17
4
Problems with PRI with T410 messages
Hi all,
I have a box running asterisk with T410 connected to a Nortel DMS 100 switch
and another box running SER with grandstream phones on it
So if there is a call from the pstn it goes from the Nortel to the asterisk
and then to the SER box and finally to the phones.if the phone is busy or
the number is invalid the * box will first send an ALERT message to the
Nortel and say the call is going on
2004 Jan 27
4
Introducing 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 -
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
2004 Dec 12
1
Totally LOST with dialplan and Extensions.
Ok I have spent the last week working on getting my small PBX to work.
I will in the end only have 4 SIP extensions being either softphones of
IP phones. Currently only 1 SIP config for testing.
And at the this point it should be all fairly easy with all inbound and
outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via
IAX. Inbound does work in it's current basic state.
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 Nov 03
5
FireFly Problems
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
2005 Mar 02
4
timing/clock problem
Hi all,
We have been fighting with telco for a entire week.
Today they came here with a LITE3000 to analyze what is going on.
When I configure zaptel with no external clock, E1 gets aligned/synchronized
with bit rate in 2048000 bps, both me and telco.
span=4,0,0,ccs,hdb3,crc4
But when I configure span4 to get clock source from telco they become
unsynchronized. TElco bit rate stays in
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 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 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