Displaying 20 results from an estimated 2000 matches similar to: "Fedora Core 2 softphone"
2004 May 25
4
Sip/IAX Clients for Linux
Hi There,
i think all VOIP clients for Linux are unusable!
i got testet:
Linphone + Linphonec all in version 12.2
Kphone
gophone
and other...
the only programm that is usable is gnomemeeting...
does anybody knew some other tools?
Best Regards,
Mark
2003 Jun 24
1
Working Clients for Linux?
All the clients that I'm aware of for IP telephony have drawbacks. Some
won't work at all.
KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to
dial tones during the middle of the call, so the demo that * comes with
can't be run. Kphone (3.1, the latest) also has a habit of crashing if
you do something even mildly stressful, such as hang up while Kphone is
2004 Aug 06
1
Asterisk Dry Run
Hi everyone,
I just installed asterisk on my system with the purpose of rerouting calls
on sip channels.
I don't think i need any hardware for that.
I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both
of them and to call one from the other on the same machine, however could
not.
I 1-) I could connect sjphone in isolation to freeworld dialup howver i got
no sounds
2005 Jan 14
5
Softphone for Linux recommendation
Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.
So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back from the X-Lite beta folks.
--
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
------------------------------------------
"When you just
2004 Sep 10
1
(Resend) Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs
Got no responses to this, but the list seemed to be down for a while, so
here it is again. Sorry for the extra bandwidth!
John
Hi, I've been messing with getting SIP working for days now, with
limited success. I've got Asterisk set up on a remote server with the
echo test. Please try it out to verify I've got the server working
right:
sip:robot at nixon.butchwax.com
2009 Mar 03
2
Asterisk analog DID with Adit 600
Hello All,
I'm trying to connect Asterisk to an Executone phone system with an
analog DID card and I'm hoping someone can help me figure out what I'm
doing wrong. The Executone DID card provides battery to the telco, when
the telco wishes to dial a DID it goes off-hook, waits for a wink from
the Executone and then dials the last three digits on the number with
pulse (as opposed
2005 Aug 04
1
Asterisk Voice Mail Server and older Executone PBX..can it be done?
Does anyone have experience with melding Asterisk with an older Executone PBX?
I have a client whose existing voicemail server(repartee) has become
bonkers and we need to stick a VM system in there asap. I thought
asterisk would be a good thing to use.
Does anyone have experience with the older Executone PBX's and asterisk?
Any caveats, any tips, any things I should be aware of?
Thanks,
2004 Dec 18
1
Setting up asterisk for one user in private ip NAT.
Hi.
I've just bought SIP telephony service from a Swedish telco.
I've managed to make and receive calls with kphone.
Now I want to set up asterisk to be able to add fancy features like
voice mail and recording conversations. But first I
have to get the basic setup right. I'm running asterisk and kphone on
the same machine, behind at NAT-router.
When I make a call (from my regular
2005 Feb 14
6
Linphone / Kphone
Hi,
I have * working with X-Lite and Sipura adapters, but I have one person
who is linux based, and is trying to use Linphone and Kphone. His end
works, but I get very bad echo on my end. Have any of you folks been
able to get linux based soft phones working well with *?
I'd appreciate links to howtos/docs if you have them, and/or samples of
working configs for * and the linux
2003 May 15
3
Linux SIP/IX clients
DOes anyone have any good suggestions as to good SIP or IAX clients for
linux? I have set up and am currently testing asterisk in a controlled
environment. I have gnophone running on one of my boxes but the gnophone
site has been down. So I can't seem to fing the IAX and Ix-devel rpms or
the gsm and gsm-devel rpms. So that prevents me from setting up gnophone
on another box. I am
2005 Feb 01
2
Soft phones that _actually_ work under Linux?
Surely there has to be one soft phone that works under Linux.
I've tried:
kphone - it sometimes complains about the need to release the sound
device
linphone - sssssssslllllllllooooooooowwwwwww
iaxcomm - needs some strange widgets
various others - either only supplied as binaries, or just plain don't
work, or won't compile.
Is there just one out there that is guaranteed to work with
2004 May 29
4
PlayTones problem
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi!
I am having problems with the PlayTones application and VoIP softphones.
I have the following in my extensions.conf:
exten => 123,1,Answer
exten => 123,2,PlayTones(Busy)
exten => 123,3,Hangup
But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call
just hangs up immediately.
I get the following on the console:
--
2003 Apr 24
7
Outgoing SIP Call to unregistered Users
Hi!
I'm using asterisk with a few kphone SIP-Clients. The registration process
seems quite OK. But there are some problems:
Calling other registered users is possible, but the rtp-stream is not reaching
the right port, so you can hear nothing. In ethereal you can see, that the
SIP/SDP fields addresses different ports at each client, so client A sends to
port 32000 but client B listens on
2003 Jul 01
2
Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Today's "frustrated programmer" award goes to Linphone, which has the
following debug output:
> (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer!
I get this message when I connect to linphone using a softphone, or when
I try to use linphone to connect to asterisk and listen to an
announcement. I suspect that
2004 Apr 26
3
dtmf tone clamping in calls to external ivr
Hello,
I'm having trouble working out how to send DTMF tones to an external
IVR. My system has an analog phone connected to a TDM400P card, a SIP
software phone (Zultys LIPZ4) and is connected to a BRI in Australia
with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched
with the ISDN audio patch from Traverse (which allows the card to do
voice).
DTMF works fine between
2004 Sep 30
2
OT: Kphone installation problem
Hello,
I know that my Kphone question may be a bit off topic, but I have been
busy with this again and again for about one month now, sent three
mails to kphone@wirlab.net (the contact address mentioned on
http://www.wirlab.net/kphone/index.html), asked for a solution in a
german ip phone forum and tryed many things by myself.
I try to compile KPhone 4.0.3 (tryed CVS Version as well) but
2004 May 25
2
sip phone problem
Hi all.
I have 2 ip phones (Grandstream Budgetone):
-budgetone1
-budgetone2
All two are connected to an Asterisk server.
When I make a call from budgetone1 to budgetone2, I
can speak with budgetone2 whith no problem. But when
budgetone2 hangs up, budgetone1 does not play any tone
(like busy tone). Budgetone1 seems to be still in
conversation, but what conversation!
Has anyone had a problem
2004 May 25
1
Troubles with Kphone]
-------- Original Message --------
Subject: Re: [Asterisk-Users] Troubles with Kphone
Date: Tue, 25 May 2004 15:44:15 +0530
From: Murali Krishnan <murali@bksys.co.in>
Reply-To: ismk@myrealbox.com
Organization: bk SYSTEMS (P) LTD.,
To: asterisk-users@lists.digium.com
References: <200405250652.46370.klky3@fibertel.com.ar>
enano wrote:
>Hi ,
>
>
>
>I'm triying to use
2004 Aug 24
0
Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs
Hi, I've been messing with getting SIP working for days now, with
limited success. I've got Asterisk set up on a remote server with the
echo test. Please try it out to verify I've got the server working
right:
sip:robot@nixon.butchwax.com
Running FC1, ThinkPad T22, headset thru the soundcard. Asterisk is
asterisk-1.0_RC1. No NAT. The phones I've tried so far are as
2003 Apr 16
4
iLBC
i tried asterisk ilbc codec against kphone. when the call got
connected, i started to immediately get these kind of message to the
console:
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)?
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of