There's an update to Firefly on Virbiage
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
lots of bug fixes - see if that helps
-Adam
Paul Redstone wrote:
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
Ivan Meic (Vox Mundi) wrote:
Actually G.729A is a reduced complexity version, and G.729B is a version
with silence suppression. The data rate while sending voice is exactly
the same, although the quality of G.729B should be a little higher.
However the average rate for B can be lower if the
What's your end device? if it's a voip device (eg SIP phone or a soft
phone) then you shouldn't need a jitter buffer.
Also, you don't need bandwidth=low if you specify the codecs (the
disallow=all will override the bandwidth=low) and maxjitterbuffer is the
param you're after with this line
Steve Kann wrote:
Something *proprietary* is something exclusively owned by someone
nobody owns the IAX2 protocol.
Although, Digium have trademarked IAX
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Ethereal on various boxes should help solve the issue (probably firewall)
Jon Walsh wrote:
When I connect to the third party softphone (firefly) I get connected
at my house and at my office where I have the asterisk..but when I
went to my friends house to set him up his firefly showed a gray
you mean IAX isn't a standard :) Also IAX requires your call router /
billing gateway to handle the voice traffic too (or you put your CDR
recording at the end points) With SIP, just the signaling is needed,
allowing more scalability. I recall talking about this at astericon but
it never
Duane wrote:
Adam Hart wrote:
As always, I'm happy to announce a new version of Firefly.
Firefly 1.9.8 has more of what you want and less of what you don't
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
There's a few bug fixes - notably fixed the Reject button and sending
As always, I'm happy to announce a new version of Firefly.
Firefly 1.9.8 has more of what you want and less of what you don't
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
There's a few bug fixes - notably fixed the Reject button and sending of
audio before answering in some
use ethereal or iax2 debug to see what capabilities are been set in your
NEW message
Ernie Ankele wrote:
Hello,
Could someone give me clues where to figure out this problem?
If I call from a Sip client to an Firefly client running IAX, the call
connects fine, no problems.
I can connect to
: 15725 DCall: 3 [xx.xxx.xxx.xxx:20406]
CAUSE : No compatible Codecs
Jan 10 18:50:06 WARNING[1378]: chan_iax2.c:6017 socket_read: Call
rejected by xx.xxx.xxx.xxx: No compatible Codecs
Thanks, Ernie
On Jan 10, 2005, at 6:34 PM, Adam Hart wrote:
use ethereal or iax2 debug to see what
Michael Vogel wrote:
Hi!
The encoding, decoding and recoding cost cpu time, that's sure. But does
this time differs much depending on the used codec?
Is - for example - a G729 faster than a GSM codec?
Try 'show translations' in asterisk's CLI
(GSM is much faster than G.729)
Christopher L. Wade wrote:
Matthew Boehm wrote:
Can you say 'overkill' ? *smiles*
I just recorded a 2min voicemail and the resulting file on the server was
slightly over 200KB in size.
We are only storing 1 format of soundfiles, WAV49.
A 160GB drive is approx 1,677,721,160 KB.
At the rate above
Bastian Schern wrote:
Hello Asterisk friends,
is it possible to avoid plain text passwords in the iax.conf or the
iaxfriends MySQL database table?
Asterisk needs the plain text password to authenicate. You could wrap a
base64 decode when reading the passwords, but this is obsecurity, yet
Bastian Schern wrote:
Adam Hart schrieb:
Bastian Schern wrote:
Hello Asterisk friends,
is it possible to avoid plain text passwords in the iax.conf or the
iaxfriends MySQL database table?
Asterisk needs the plain text password to authenicate. You could wrap
a base64 decode when reading
Bastian Schern wrote:
Adam Hart schrieb:
Bastian Schern wrote:
Adam Hart schrieb:
Bastian Schern wrote:
Hello Asterisk friends,
is it possible to avoid plain text passwords in the iax.conf or the
iaxfriends MySQL database table?
Asterisk needs the plain text password to authenicate. You could
Niels Chr. Sørensen wrote:
Hi,
In constant search for optimization, a friend told us about his experience
with Gentoo Linux-distro. He claimed that he doubled the performance of his
server by changing to Gentoo from Debian.
Does anyone have any experience with running Asterisk on a Gentoo linux?
nkb wrote:
Hi.
I'm really new.
I was just wondering if it is possible at all to do a IP to IP call
without a * server (or as a matter of fact, any other kind of server)?
say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at
hisdomain.com's 192.168.0.3. Is this sort of things
Andrew Kohlsmith wrote:
On November 23, 2004 05:28 pm, Adam Hart wrote:
iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -m state
--state NEW -j DNAT --to-destination ASTIP
iptables -t nat -I POSTROUTING -p udp -d ASTIP --dport 4569 -j MASQUERADE
Any reason why you need both
Run ethereal and look the dump, prehaps A) the SIP invite doesn't match
the correct IP port B)try turning on Asterisk's NAT fix C) send the
dump to me :)
-Adam
Alejandro Gutiérrez wrote:
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
An untested guess
iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -m state
--state NEW -j DNAT --to-destination ASTIP
iptables -t nat -I POSTROUTING -p udp -d ASTIP --dport 4569 -j MASQUERADE
cheers,
Adam
Peter Osborne wrote:
Hello,
With all the talk about Firefly, I decided to check
Chris Olson wrote:
Chris Olson wrote:
Hello,
I have firefly installed and it is somewhat working. It is registering
with my Asterisk server and I can call out, but I receive no audio
coming into Firefly. From the Asterisk end, everything looks OK with
the call, just no audio is being received on
Chris Olson wrote:
Thanks Adam. Can you let us know when the fix is available and
where we can download the fixed 3rd-party from?
A little more info ... this is actually a one-way audio problem as
audio passes from Firefly to Asterisk, but not from Asterisk to
Firefly.
You can grab the new
Chris Olson wrote:
Hello,
I have firefly installed and it is somewhat working. It is registering
with my Asterisk server and I can call out, but I receive no audio
coming into Firefly. From the Asterisk end, everything looks OK with
the call, just no audio is being received on the Firefly end.
try running ethereal, make sure everything looks ok and send me the
result. No firewall? Also, download debugview from www.sysinternals.com
to see Firefly's debug msgs. Could be simply wrong audio device?
Andrew Kohlsmith wrote:
Using Firefly 1.9.5 (thirdparty) on Win2k
Using Asterisk CVS HEAD
Andrew Kohlsmith wrote:
On October 31, 2004 05:36 pm, Bastian Schern wrote:
I've got no success to get a friend in Bogota (Colombia) connected to my
Asterisk. He has got a ISDN Internet connection and the UDP packets will
be fragmented. It seems that the MTU of this connection is round about
400
[EMAIL PROTECTED] wrote:
Hello
I would say,
First of all, for users who are authenticated, so really can make calls,
just configure asterisk to limit the number of calls users can make
concurrently
Next, put a firewall in front of your asterisk box which rate limits the
number of connection
Remember the requirements of GPL is regarding distribution, not use, you
can do what ever you like with it internally, with no requirement to
publish it. Config files being GPL doesn't really make sense as you
would only ever be distributing them as they are anyway (not compiling them)
GPL in
Ronald Wiplinger wrote:
On Tuesday 26 October 2004 12:33, Adam Hart wrote:
Remember the requirements of GPL is regarding distribution, not use, you
can do what ever you like with it internally, with no requirement to
publish it. Config files being GPL doesn't really make sense as you
would only
The best way for me or yourself to debug it is using ethereal (google
for it) and debugview from www.sysinternals.com. I'm happy to help, so
send the logs, the native transfer might be the issue.
-Adam
Willem de Groot wrote:
Is anyone succesfully using FireFly-Thirdparty in SIP modus with
We'll add that to next version, should be out next week
Deon Rodden wrote:
I put FireFly on my moms computer, but ran into a problem. She went
home and was able to place calls from it (using her headset and such).
But, she could not receive calls. I figured out the problem was with the
Seems to be alot of these questions on the mailing list recently. AUSTEL
is the old name for the ACA, A-tick is the correct term for certification.
It's only illegal if you connect to a carrier network without A-tick
(you can get consent from them to connect without A-tick).
The ACA has plently
register on gateway.freshtel.net, not cts-au
-for firefly numbers, call gateway.freshtel.net
-for PSTN termination, call cts-au.freshtel.net
don't think you need the @freshtel at the end of your dial
good luck,
Adam
Shaun Dwyer wrote:
Hi,
I'm having some problems getting calls to go out via
Dave Cotton wrote:
On Sat, 2004-08-28 at 23:36 -0700, Muiz Motani wrote:
I'm using the firefly third-party softphone. However, the same thing happened
when I used IAXphone 2.0.
I can't offer any real solution because I was only testing the
connection with Firefly, but I got exactly the same
try installing mysql-devel
-Adam
DIPAK PAUL wrote:
Hi Every one and Lerale Erwan
I have briefly describe my problem and I have provide the steps as follows:
I have intalled redhat properly and from the konsole I checked with mysql.
rpm -qa | grep mysql and the konsole provide me the message:
Daniel Niasoff wrote:
Hi Everyone,
Is G729 more sensitive to packet loss or delays due to its higher
compression. If Ive generally got the bandwidth available, am I best
sticking to ulaw.
G.729 has lost packet concealment, G.711 doesn't. G.711 will sound
better otherwise if you can afford
Steve Underwood wrote:
Adam Hart wrote:
Daniel Niasoff wrote:
Is G729 more sensitive to packet loss or delays due to its higher
compression. If Ive generally got the bandwidth available, am I best
sticking to ulaw.
G.729 has lost packet concealment, G.711 doesn't. G.711 will sound
better
just send me your key and I'll help :p just kidding
try ftp://ftp.digium.com/pub/asterisk/g729/ the README and the needed
files are in there
Jean-Yves Avenard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dear all
Last week I purchased 10 G729 codec licenses from Digium. The only thing
I
try sip debug and see what each side is offering in codecs (make sure yo
u have allow=g729
Walter Klomp wrote:
Hi,
I am trying to post this again as I am getting no answers and the
[EMAIL PROTECTED] bounces...
(I have searched the whole list and can't find the answer either)
I have installed a
Daniel Daley wrote:
I'm hoping someone might have seen this before because I'm just about at
a loss of what to do. I have an asterisk system setup in a call center
environment with multiple queues. After a random uptime asterisk will
suddenly come to a partial halt where I can connect to the
Chris Foster wrote:
The Register is carrying a article written by Kevin Poulsen of
Securtiy Focus, calling asterisk ..the most powerful tool for
manipulating and accessing CPN data..
http://www.theregister.co.uk/2004/07/07/hackers_gut_voip/
I hope NuFone doesn't drop asterisk-set-able
Bradley D. Thornton wrote:
snip
i don't need nats, nat traversal, nat anything. if i did, iax
might well be one of the technologies i would consider. but i
don't.
snip
Watch out for this man Bush! He is a professional espionage troll
and hides his agent status behind his condescending
[EMAIL PROTECTED] wrote:
I have no idea who Randy Bush is but I found it funny the first article
I found on him was a presentation on why NAT is evil espically for
voice. Now he asserts that NAT traversal is not needed.
Jay Milk wrote:
I'm guessing it's too expensive -- looks like my friends took a
reference design and barely modified the sample firmware. I was
surprised to even find g729 in there (licensing cost), but I'll take it.
I'd be glad to get CID name and MWI working, and wouldn't even mind if
they
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
I may be wrong but prehaps the answer is in your email
-- Executing DigitTimeout(SIP/avenardj-acfc, 5) in new stack
-- Set Digit Timeout to 5
-Adam
Jean-Yves Avenard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dear all.
I'm new to this so please forgive my ignorance if I missed
try tcpdump -i lo port 5432 or icmp
(or tethereal if you have it)
Prehaps it's trying a UNIX socket connection?
also, please change your database password as you've now supplied
ip,user,pass to the mailing list :) Hopefully, you've got it restricted
to localhost
Caleb Kow wrote:
Here we go:
Andrew Kohlsmith wrote:
On Wednesday 23 June 2004 04:46, GIBERT Frédéric wrote:
Has someone know a good call generator for asterisk including SIP protocol
(freeware if possible)?
I need to stress a plateform and I don't find any.
Are there any IAX2 call generators?
Regards,
You can use asterisk
check under your network settings that you have all the codecs selected
and obviously type IAX
Jason Penton wrote:
Hi All
I have a strange problem using IAX2. When placing a call to my IAX clients
(firefly) via the Asterisk dialplan all works great. However trying to
initiate a call via the
] On Behalf Of Adam Hart
Sent: 17 June 2004 08:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX2 no compatible codecs
check under your network settings that you have all the
codecs selected
and obviously type IAX
Jason Penton wrote:
Hi All
I have a strange problem using IAX2. When placing
Kevin P. Fleming wrote:
Adam Hart wrote:
I've also added support for SIP via TCP and the ability to change the
SIP port
It complains every time you click OK in the Options page about Changing
SIP port requires restart, even if you never looked at the SIP page
(and don't even have any SIP
With all the talk of SRV support in Asterisk, I thought I'd add support
in Firefly so enjoy. Thanks to Olle for helping me with it, explaining
the wonderful world of SIP and SRV to me. There's also an option to
disable it (seems to take quite a few DNS lookups for SRV) - warning
Duane may hunt
Jason A. Pattie wrote:
|
| One workaround is to use Firefly, but that may not be for everyone?
True. I almost got it working under Wine, though. Kept dumping files
into C:\. Probably just means I don't have the necessary dependencies
or Wine doesn't have the capabilities needed to run this
Tor Houghton wrote:
I have the same problem. IAXCOMM works fine with * 0.7.2, but not 0.9.
However, you can make calls fine, just not pick up inbound calls.
One workaround is to use Firefly, but that may not be for everyone?
To the Firefly maintainer: why does the contacts list fill up with
There's a new version out with some bugs fixed
major ones fixed: deadlock on call end, iax thread getting locked out,
few contact group list bugs, one on exit crash bug fixed
I'd highly recommend upgrading to it
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
-Adam
Adam Hart
:
On Wed, 2 Jun 2004, Adam Hart wrote:
I'd highly recommend upgrading to it
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
Can I recommend you label files with version numbering - this must be
about the third ? fourth ? firefly-thirdparty you've released
: deadlock on call end, iax thread getting locked out,
few contact group list bugs, one on exit crash bug fixed
I'd highly recommend upgrading to it
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
-Adam
Adam Hart wrote:
As Promised, I've released a new version of Firefly (ver 1.8
there is no chance to influence the RTP/RTCP
Portrange for the audio channel.
Please correct me if I'm wrong.
jo
Adam Hart wrote:
I just put up another version - fixed that issue and also added to
ability to disable registration to a network. Why it's needed? If you
will only be making outgoing calls but still
the log looks legit except why does asterisk have a different IP in the
contact compared to the 'to' address.
I can connect successfully to my asterisk server and FWD - can anyone
give me sip access to a asterisk server that firefly doesn't work on?
[EMAIL PROTECTED] wrote:
Why all the time
PROTECTED]
31/05/2004 09:19 PM
Please respond to
asterisk-users
To
[EMAIL PROTECTED]
cc
Subject
Re: [Asterisk-Users] New Firefly version
Thanks Adam,
no crash after installing over 1.5 B3388. However changing the SIP RTP
Port is still not accepted.
jo
Adam Hart
I'll look at it tomorrow
jo wrote:
Thanks Adam,
no crash after installing over 1.5 B3388. However changing the SIP RTP
Port is still not accepted.
jo
Adam Hart wrote:
As Promised, I've released a new version of Firefly (ver 1.8) with IAX
SIP support back in.
Get it from Virbiage site
It's the standard LibIAX2, the nice features are implemented using text
messages. I'd recommend you use the standard LibIAX2 as it's more upto
date (Something I've been needing to do too)
Reto Stauss wrote:
Hi
Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features
PROTECTED]/extension
jo wrote:
Thanks Adam,
no crash after installing over 1.5 B3388. However changing the SIP RTP
Port is still not accepted.
jo
Adam Hart wrote:
As Promised, I've released a new version of Firefly (ver 1.8) with IAX
SIP support back in.
Get it from Virbiage site or here's
As Promised, I've released a new version of Firefly (ver 1.8) with IAX
SIP support back in.
Get it from Virbiage site or here's the direct link
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
If it crashes on startup, export your Firefly tree from the registry
(current user -
Duane wrote:
Adam Hart wrote:
As Promised, I've released a new version of Firefly (ver 1.8) with IAX
SIP support back in.
STUN support doesn't seem to work... Keeps saying unable to contact stun
server, and when I did a packet dump and closed and reopened the prog
several times I couldn't see
this has been a cause of many crashes,
people having Xten running in the background. Thanks to Karl for the
dump file on that one.
keep the bugs coming,
Adam
PS hope you're enjoying the new contact groups :)
Adam Hart wrote:
Duane wrote:
Adam Hart wrote:
As Promised, I've released a new version
I'm going to have to go against this statement, there's one bug that I
need to fix so unfortunately it will have to be Monday now.
For those after the IAX/SIP firefly (albeit an old version) get
http://www.virbiage.com/firefly/download/firefly-dev.exe
apologies,
Adam
Adam Hart wrote:
They'll
They'll be a new version at the end of the day (it's 9:25am now) - The
reason it was like that was to cope with overlap for the firefly network
going to Freshtel. Freshtel will have the Firefly Network and special
version of Firefly (no IAX and SIP) while Virbiage will have a standard
IAX and
Adam Goryachev wrote:
On Fri, 2004-05-28 at 09:28, Adam Hart wrote:
If anyone's after Australian IAX termination (or Australians wishing to
call overseas), try www.freshtel.net - iax server is ctsau.freshtel.net
Except I get:
[EMAIL PROTECTED]: ~$ mtr ctsau.freshtel.net
mtr: Unknown host
Adam Goryachev wrote:
I suppose I could do QoS on outbound, which should improve things
somewhat for the remote caller, but that doesn't help inbound packets.
Does anyone have any comments on what this would mean for VoIP calls
with the above variables?
I think the biggest problem is the
Tony Hoyle wrote:
Scott Brooks wrote:
Has anyone ported the ztdummy module to 2.6? I don't really want to
dive into it that far if someone already has.
http://www.nodomain.org/asterisk/ztdummy.diff
:)
Tony
Forgive me for prehaps a stupid question but does the 2.6 kernel have
accurate timers
Tony Hoyle wrote:
Adam Hart wrote:
Forgive me for prehaps a stupid question but does the 2.6 kernel have
accurate timers built in now? as I see your code just wraps their timer.
The HZ value in 2.6 is now 1000 to support realtime scheduling etc.
It's certainly accurate enough. I'm not sure
Andrew Yager wrote:
Hi,
Last weekend I was planning to buy a physical PBX system, but instead I
have been blown away by the fact that VoIP really works, that Asterisk
is so easy to set up and use... and free!
We're in Australia, so as I understand it, we aren't allowed to use the
Zaptel cards.
Kevin Walsh wrote:
brian [EMAIL PROTECTED] wrote:
I've seen that licenses are purchased on a per-channel basis. Could we
make some sort of agreement on having a no-limit channel license? Even,
we would like to have the possibility of installing it on how many
machines we wish to do.
No you MUST
compared to? My P4/Xeon 2.8 does SLINR - iLBC in 12ms so a 2.4ghz
should take 14 (?)
Andrew Kohlsmith wrote:
-fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse
-msse2 -mfpmath=sse
Made no difference whatsoever on my P4/Xeon 2.4GHz for iLBC (I don't use
speex).
-A.
Actually it encodes a second of data, which with a 20ms codec would be
50 frames. The timing shows better than expected results due to caching.
-Adam
brian wrote:
http://asterisk.bkw.org/diff/translate.patch.txt
If you try that patch out it adds a nice feature...
show translation recalc [xx]
You
:
Yes I realized my error in my wording but it was early :P It doesn't
improve alot but does give you some ways to get a better idea of translation
times if your box is loaded up with calls.
bkw
PS this patch was added to CVS-HEAD
- Original Message -
From: Adam Hart [EMAIL PROTECTED
John Todd wrote:
At 8:23 PM -0600 on 5/12/04, Rich Adamson wrote:
Current dev cvs install on two systems. System A is behind a SonicWall
firewall, and system B is on a registered IP address. (System B has
multiple iax links that are fully functional to multiple locations.)
System A is correctly
We're waiting on the processor chip to be made for our first production
run, there's currently no stock and they're in the process of making
more. It's completely out of our hands and, trust me, I'm as frustrated
as you guys are.
As soon as our manufactures tell us the completion date, I'll
apply the openh323 patch (it's in the root of ast-oh323), recompile
openh323 and it should work fine
David Hindmarsh wrote:
Hi
I have recently updates to the latest cvs of asterisk, openh323 and pwlib as recommended.
The OPenh323 and pwlib compile fine.
When compiling the Asterisk-oh323 I
rr80 wrote:
Is there is support for G.723 codec in Asterisk 0.7.2+Astrisk-OH323 0.5.10 or it should be bought separately like G.729?
-
Pavel Riko
___
Neither, you can't get asterisk to en/decode G.723.1 - only proxy it
Duane wrote:
William Suffill wrote:
They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who use hardware phones but still want to keep TXT
chat and presence and perhaps integrated
WipeOut wrote:
Doesn't NuFone use SER in front of Asterisk? so using asterisk purely
as the PSTN gateway..
Later
Nufone offers IAX termination, SER is SIP - or am I missing something here?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Ken DeMaria wrote:
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.
Aaron Martin wrote:
Has anyone heard any more info about the Virbiage FT201 VoIP phones?
About 3 months ago I was told they were 6 weeks away, about 3 weeks
ago I was told they were 2 weeks away, and now I am told they are 2
months away again! Are they EVER going to arrive? Can anyone shed
Read README under channels/h323, it should point you in the right direction
Terence Parker wrote:
I have posted before but didn't get any replies so i'll ask again in a
more simple way :
Does H323 work on asterisk out of the box? I notice there is already a
channels/chan_h323.c file, but
Carlos Chavez wrote:
I see that I can purchase G.729 licenses for my Asterisk server, but I
have seen that many phones support a G.729 variant like A or B. Are these
suppoted by the same G.729 codec in Asterisk?
B is just the fixed point version of A (from memory) - so it works the
same
I've put up a new dev version of Firefly
(http://www.virbiage.com/firefly/download/firefly-dev.exe)
Notable Changes:
DTMF now works with SIP
Speex codec has been added
1 crash bug fixed - 2 more to go (if you can crash Firefly, send me the
Hex address - probably stored in event viewer under
Olle E. Johansson wrote:
An informational RFC documenting the protocol would be a good start,
it would
make it more open but not an IETF product. Security specialists would
get something
to read and analyze. A VOIP protocol with RSA authentication,
implemented today.
Is there any IAX2
Robert Hajime Lanning wrote:
quote who=Adam Hart
I also like to see two
people behind the same nat being able to communicate directly (without
requiring pin-wheeling). Ie The client attaches their private ip to the
register packet, which is used when client A B's public ips match
James H. Thompson wrote:
No guarantee then when public IPs match that clients are both on same NAT LAN.
Client A 192.168.0.1 - NAT Router A - NAT Router X with Public IP 123.123.123.123 ---
Internet
Client B 192.168.0.1 - NAT Router B -|
Jim
James H. Thompson
[EMAIL
Comment below...
Steve wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 24 March 2004 08:45 pm, James H. Thompson wrote:
No guarantee then when public IPs match that clients are both on same NAT
LAN.
Client A 192.168.0.1 - NAT Router A - NAT Router X with
Public
Simon Brown wrote:
I had exactly the same problem. I tried removing and reinstalling several
times but it always crashed. I sent an email to verbiage asking for help and
all I got in response was Have you got it working yet? from them. I have
been unable to get a reply since.
Simon Brown
Dave Cotton wrote:
On Wed, 2004-03-17 at 04:43, Adam Hart wrote:
Eric Wieling wrote:
6) are there USA resellers
Yes, many USA resellers have expressed interest. Virbiage won't be
selling directly.
And the 255 million people in Europe? Please not the usual, 75US
hank smith wrote:
hello I am not sure where to ask this question at so please except my
apologise if this is the wrong list.
I need to ask if any one has got firefly sip version to work with fre
world dialup?
if so what info did they use to connect?
once again if this is the wrong list if the
Jim Flagg wrote:
Firefly's Protocol Support now is:
Voip Protocols: SIP, IAX
Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL)
Sounds good.
Any plans for Speex codec support?
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Adding it this week, along with some bug fixes
Eric Wieling wrote:
The FT201 is currently being manufactured and will be available shortly!
The retail price will be $129.95 USD...
http://www.virbiage.com/products/lanphones.php
The web page does not say:
1) how many call appearances does the phone has
It can present 5 calls and you can
Matthew Marlowe wrote:
(reposted to be in text format, sorry. :))
The FT201 is currently being manufactured and will be available shortly!
The retail price will be $129.95 USD...
http://www.virbiage.com/products/lanphones.php
Let me clarify the FT 201 situation, the current ETA is 8 weeks.
Just a quick update, there's was a problem with SIP - if you were
getting SIP registration failed, grab the new version.
(http://www.virbiage.com/firefly/download/firefly-dev.exe)
thanks for the feedback about this bug,
Adam
Adam Hart wrote:
I've been sitting on this release for a week so
, there's was a problem with SIP - if you were
getting SIP registration failed, grab the new version.
(http://www.virbiage.com/firefly/download/firefly-dev.exe)
thanks for the feedback about this bug,
Adam
Adam Hart wrote:
I've been sitting on this release for a week so I thought I'd better
Protocol Support now is:
Voip Protocols: SIP, IAX
Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL)
Next major feature will be conferencing.
feel free to email me,
Adam Hart
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