7 Mar
2006
7 Mar
'06
4:12 p.m.
----- Esben Stien <b0ef(a)esben-stien.name> wrote:
To get a SIP soft phone, you can check out the various free software
projects at voip-info[2], specifically the SIP soft phone clients
page[3]. Be aware that not all projects on that page is free
software. If you need a terminal client, check out asterisk[4]. You
can also run your sip phone through JACK;). A GUI client might be
ekiga[5].
Could you elaborate on this? I have tried to connect soft phones to a jack server and none
worked. There appear to be a limited selection of SIP soft phones which are free software
and one IAX soft phone that I have used with OSS drivers only. Specific, tested program
recommendations would be super cool.
FWIW, I have an Asterisk server running that has /plenty/ of bandwidth to go around for
conference rooms.
-lee
7 Mar
7 Mar
4:31 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
Only kphone has jack support built in, and it doesn't upsample or
downsample, so you have to run your jack server at 8khz. Even then it
has trouble if your hardware can't do 8khz. It would be nice to have a
jack driver and jack codec for asterisk.
So far as I know the cvs version of gnomemeeting (now, akiga) has
iaxclient support.
I love the idea of a standard conference room server for linux-audio
users. Freeswitch's #freeswitch irc channel is integrated with their
audio conference room, and it's a gas. Useful would be a weekly
concall for developers of the various linux audio projects, or just a
place to hang out sharing helpful tips and information, even music....
Like everything in this world such a concall would only get more
useful as metcalfe's law kicks in. If anyone needs help on how to
setup an asterisk server or needs help getting voip (SIP or IAX)
working in general, feel free to drop me a line. At the moment I have
several asterisk servers operational, (a podcast -> iax/sip gateway
for thespaceshow is publicly available) but few have tons of bandwidth
available, so perhap's lee's server could become a std meeting
place....
On 3/7/06, Lee A. Azzarello <lee(a)rockingtiger.com> wrote:
----- Esben Stien <b0ef(a)esben-stien.name> wrote:
--
Mike Taht
PostCards From the Bleeding Edge
http://the-edge.blogspot.com
To get a SIP soft phone, you can check out the various free software
projects at voip-info[2], specifically the SIP soft phone clients
page[3]. Be aware that not all projects on that page is free
software. If you need a terminal client, check out asterisk[4]. You
can also run your sip phone through JACK;). A GUI client might be
ekiga[5].
Could you elaborate on this? I have tried to connect soft phones to a jack
server and none worked. There appear to be a limited selection of SIP soft
phones which are free software and one IAX soft phone that I have used with
OSS drivers only. Specific, tested program recommendations would be super
cool.
FWIW, I have an Asterisk server running that has /plenty/ of bandwidth to go
around for conference rooms.
-lee
4:49 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
On Tue, 2006-03-07 at 12:31 -0800, Mike Taht wrote:
Only kphone has jack support built in, and it
doesn't upsample or
downsample, so you have to run your jack server at 8khz.
Heh, that's pretty funny. Seems like it should be easy to fix if JACK
is running at the standard 48000Hz or 96000Hz.
Lee
6:30 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
On Tue, 7 Mar 2006 at 15:48 -0500, Lee Revell wrote:
On Tue, 2006-03-07 at 12:31 -0800, Mike Taht wrote:
It's probably easier to fix kphone's jack support, but if anyone has the
itch I once upon a time started a jack IAX client for talking to
asterisk. It's not done, but what I did do is available. It just needs a
UI and a jitter buffer, if my memory serves correctly. A simplistic UI
wouldn't be hard, but realizing I needed a jitter buffer is where it
stopped being fun. ;-)
http://hans.fugal.net/src/alex
At the time I was not impressed with any softphones for linux, so I
started my own. Now I really like Twinkle, so if I really wanted JACK
I'd see if I could add it to Twinkle.
http://twinklephone.com
--
Hans Fugal ; http://hans.fugal.net
There's nothing remarkable about it. All one has to do is hit the
right keys at the right time and the instrument plays itself.
-- Johann Sebastian Bach
Only kphone has jack support built in, and it
doesn't upsample or
downsample, so you have to run your jack server at 8khz.
Heh, that's pretty funny. Seems like it should be easy to fix if JACK
is running at the standard 48000Hz or 96000Hz.
8:42 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
This is mostly a note for Esben...
I setup an asterisk extension on one of my boxes to dial the
1222GNUAUDIO extension
it's very straightforward, for you asterisk newbies out there, it's a
one liner in your dialplan:
exten => 333,1,Dial(SIP/1222GNUAUDIO(a)proxy01.sipphone.com);
Judging from your setup description, however, it's not clear if the
"-"s in 1-222-GNU-AUDIO are not required and in fact is invalid syntax
for dialing via some clients.
sip:1222GNUAUDIO@proxy01.sipphone.com
sip:122246828346@proxy01.sipphone.com
Secondly, these two numbers are not equivalent in your conference
setup map. I just dialed both the ascii version and the number and
they dump you into distinct conference rooms.
The "hold" sample is quite nice. I would argue in favor of a greeting
message "welcome to the lad conference room", and quiet crickets, over
the repeated "you are the first participant" message (from a "I'll
just hang out here and talk to anybody" perspective)
I'm sore tempted to open up a whole bunch of sip/iax/h323 extensions
on my own box with gateways to free world dialup, firefly, sip guest,
iax guest, etc...
Is there anything in sipphone.com's terms of service against using
their conferencing service in this way?
On 3/7/06, Hans Fugal <hans(a)fugal.net> wrote:
On Tue, 7 Mar 2006 at 15:48 -0500, Lee Revell wrote:
--
Mike Taht
PostCards From the Bleeding Edge
http://the-edge.blogspot.com
On Tue, 2006-03-07 at 12:31 -0800, Mike Taht
wrote:
It's probably easier to fix kphone's jack support, but if anyone has the
itch I once upon a time started a jack IAX client for talking to
asterisk. It's not done, but what I did do is available. It just needs a
UI and a jitter buffer, if my memory serves correctly. A simplistic UI
wouldn't be hard, but realizing I needed a jitter buffer is where it
stopped being fun. ;-)
http://hans.fugal.net/src/alex
At the time I was not impressed with any softphones for linux, so I
started my own. Now I really like Twinkle, so if I really wanted JACK
I'd see if I could add it to Twinkle.
http://twinklephone.com
--
Hans Fugal ; http://hans.fugal.net
There's nothing remarkable about it. All one has to do is hit the
right keys at the right time and the instrument plays itself.
-- Johann Sebastian Bach
Only kphone has jack support built in, and it
doesn't upsample or
downsample, so you have to run your jack server at 8khz.
Heh, that's pretty funny. Seems like it should be easy to fix if JACK
is running at the standard 48000Hz or 96000Hz. 
8 Mar
8 Mar
11:57 a.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
"Mike Taht" <mike.taht(a)gmail.com> writes:
invalid syntax
Damn, /me hides. I knew this too well;). The SIP URI is just like any
URL and can contain any character of course.
welcome to (..)
Yeah, would be great with a greeting, too
anything in sipphone.com's terms of service
against using their
conferencing service in this way?
Well, if Lee setups a room on his PBX, then we don't need sipphone.com
and that's great because it seems they kick idlers.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
11:51 a.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
Hans Fugal <hans(a)fugal.net> writes:
I once upon a time started a jack IAX client
Woah, this is really great. I hope you continue with this project. We
badly need a terminal client. I really don't need a GUI for this.
Why not use a simple existing SIP stack?.
An extremely simple client to just connect to the PBX is really enough
for me. Anything else I can do by sending messages to the PBX.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
11:42 a.m.
New subject: [linux-audio-user] GNU Audio Community Conference Room
"Lee A. Azzarello" <lee(a)rockingtiger.com> writes:
I have tried to connect soft phones to a jack server
and none
worked.
You can use oss2jack. It will work with iaxcomm and kphone with no
problems. Been using it for many months. It also works to some degree
with asterisk, but got some chopping sounds. I badly need a terminal
client, but I've decided to leave asterisk in favor of freeswitch.
I have an Asterisk server running that has /plenty/ of
bandwidth to
go around for conference rooms.
This is really great cause sipphone.com has a problem with users
idling.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
1:33 p.m.
New subject: [linux-audio-user] GNU Audio Community Conference Room
Esben Stien <b0ef(a)esben-stien.name> writes:
It also works to some degree with asterisk
Of course, this is only when using the built in VOIP phone
functionality of asterisk. I connect all my soft phones which use jack
to asterisk and it works great.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
2:04 p.m.
New subject: [linux-audio-user] GNU Audio Community Conference Room
On Wed, 2006-03-08 at 18:32 +0100, Esben Stien wrote:
"Lee A. Azzarello"
<lee(a)rockingtiger.com> writes:
Please, please bug the authors of these apps to add support for a modern
audio API. It's crazy that they all think OSS is acceptable. Is it
because that's what Skype does?
Lee
I have tried to connect soft phones to a jack
server and none
worked.
You can use oss2jack. It will work with iaxcomm and kphone with no
problems. Been using it for many months. It also works to some degree
with asterisk, but got some chopping sounds. I badly need a terminal
client, but I've decided to leave asterisk in favor of freeswitch. 
3:19 p.m.
New subject: [linux-audio-user] GNU Audio Community Conference Room
On Wed, 2006-03-08 at 13:04 -0500, Lee Revell wrote:
On Wed, 2006-03-08 at 18:32 +0100, Esben Stien wrote:
more likely because OSS exists on solaris too, so they kill two
platforms with one stone
"Lee A. Azzarello"
<lee(a)rockingtiger.com> writes:
Please, please bug the authors of these apps to add support for a modern
audio API. It's crazy that they all think OSS is acceptable. Is it
because that's what Skype does? I have tried to connect soft phones to a jack
server and none
worked.
You can use oss2jack. It will work with iaxcomm and kphone with no
problems. Been using it for many months. It also works to some degree
with asterisk, but got some chopping sounds. I badly need a terminal
client, but I've decided to leave asterisk in favor of freeswitch.
4:48 p.m.
New subject: [linux-audio-user] GNU Audio Community Conference Room
Lee Revell <rlrevell(a)joe-job.com> writes:
please bug the authors
I have. I've sent mail to the kphone-, ekiga- and asterisk- list. The
iaxcomm client is using portaudio.
The JACK issue with glitch free connection is also a hot issue for
something like an inet show.
Some of these clients are ALSA capable, so it should be possible to
use the plug:jack plugin, but I'm having some issues with that and I
have more to test before I can provide a useful report for my HDSP
Multiface II and this plugin.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
5:05 p.m.
New subject: [linux-audio-user] GNU Audio Community Conference Room
On Wed, 2006-03-08 at 23:38 +0100, Esben Stien wrote:
I have. I've sent mail to the kphone-, ekiga- and
asterisk- list. The
iaxcomm client is using portaudio.
The JACK issue with glitch free connection is also a hot issue for
something like an inet show.
Some of these clients are ALSA capable, so it should be possible to
use the plug:jack plugin, but I'm having some issues with that and I
have more to test before I can provide a useful report for my HDSP
Multiface II and this plugin.
Yeah what would be perfect is a cross platform VOIP app written in a
callback model with ALSA, JACK, OSS, CoreAudio, etc backends. It seems
like most of the existing apps focus too heavily on the network and GUI
aspects and just hack together the audio part as fast as possible...
even the flagship VOIP app (Skype) has crappy audio support.
I guess the closest thing is something with a gstreamer backend. What
are the showstoppers with ekiga (formerly gnomemeeting) + JACK?
Lee
5:35 p.m.
New subject: [linux-audio-user] GNU Audio Community Conference Room
Lee Revell <rlrevell(a)joe-job.com> writes:
what would be perfect is a cross platform VOIP app
written in a
callback model with ALSA, JACK, OSS, CoreAudio, etc backends.
Isn't PortAudio supposed to be preferred option?. Maybe we need to
start a thread on this?. It doesn't help much if it don't support JACK
soon, though;). I don't even know if PortAudio can output through
gstreamer and can gstreamer even be RT safe even though it got a JACK
output plugin?.
What are the showstoppers with ekiga (formerly
gnomemeeting) + JACK?
They don't seem to care at the moment judging from the reply I got:
http://mail.gnome.org/archives/gnomemeeting-list/2005-July/msg00137.html
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
28 Mar
28 Mar
6:46 p.m.
New subject: [linux-audio-user] GNU Audio Community Conference Room
Esben Stien <b0ef(a)esben-stien.name> writes:
I [..] need a (SIP) terminal client
Linphone is a SIP daemon with a terminal interface as well as a GUI. I
got it to compile and it has native JACK support. It has some issues
that I'm ironing out in linphone-users. It's in heavy flux as it's
just replacing a core library.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
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14 comments
6 participants
participants (6)
-
Esben Stien -
Hans Fugal -
Lee A. Azzarello -
Lee Revell -
Mike Taht -
Paul Davis