be great if a real jack driver were implemented in
asterisk, but few developers have shown interest.
To get help with these issues, please try #lad and the
linux-audio-user mailinglist.
There is also a oss2jack mailinglist being set up.
We finally made it;).
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Esben Stien is [EMAIL PROTECTED
things I can
try. It would also be nice if others who experience this, screams out.
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Tom Rymes [EMAIL PROTECTED] writes:
forward port 5060
Yup, configurable in sip.conf
ports 1-2
Yup, configurable in rtp.conf
it could be more complicated than that.
Nope.
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jonny hashem [EMAIL PROTECTED] writes:
i have
You really don't say much about what you have.
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is a difficult, if possible at all
chrt(1)
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stack;)
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netfilter;). Using netfilter is way more powerful, in my
opinion.
Some dedicated filter rack modules ship with linux/netfilter now,
though.
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[sip
Moises Silva [EMAIL PROTECTED] writes:
make sure you have the next line in /etc/asterisk/modules.conf
load = app_dial.so
Not only that, but be sure to have a sound system loaded in the
modules.conf files.
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ever;). Doing speex at ultra wideband
32kHz would be awesome.
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month. I regularly build cvs
and have run into this issue before. Now it's been like this for some
weeks.
Anyone experienced this?
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[sip|iax
David Brodbeck [EMAIL PROTECTED] writes:
You can't follow the religion of free software and still run a
company that pays the bills.
I believe that you can.
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). The original
authors code would be as free as they choose to make it.
You're just repeating yourself and I'm doing the same in saying this
is what we want.
You may disagree, but I certainly see things this way.
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http://www
support on what you want, pre compiled versions or not, but this whole
idea of dual licensing is hurting us, in my opinion.
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to release
it as non free software, something which is unacceptable to me. The
whole idea of free software is to ensure the freedom of the software,
wherever it may go.
As a side note, I have nothing against giving the fsf the copyright of
my code, which is an organization I trust.
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commercial software is very important.
One can use ``copyleft'' to protect these freedoms legally for everyone.
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; no problem.
Digium also sells hardware.
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that protects it's
freedom.
When you own the entire copyright on a project, you can easily change
it. The code that has been released as free software will however
always be free, but as you have the copyright, you can also release
this code as non free (in addition).
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have missed this blasting.
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Matthew T. O'Connor matthew@zeut.net writes:
I have looked at all the info on voip-info.org
It would be nice if this was a public wiki, meaning requiring no
registration to edit. I think we would get more activity there, then.
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in the manager interface or is
there something wrong with my setup?
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I'm getting close to 90% CPU usage when doing SPEEX calls. When using
GSM everything is fine. This has only happened in the last months with
CVS HEAD. I'm running now CVS as of yesterday on
linux-2.6.12-rc3-RT-V0.7.46-02.
Anyone else experienced this?
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Rod Bacon [EMAIL PROTECTED] writes:
find firefly one of the best
Be aware that firefly is not free software.
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Any plans for asterisk to support jack for realtime audio?,
http://jackit.sf.net
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Asterisk
channel gets me nowhere. I've tried
many many times formulating it different ways, but few seems to
understand I'm only using the interactive shell mode of asterisk
dialing the echo application on the same server.
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the
latency on this.
Sure, when you use it
What exactly did you want realtime audio for (bearing in mind that
the sound subsystem is not normally used by Asterisk)?
Just for when I sit at the asterisk box.
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tried both alsa and oss and get about the same result. From the
source, chan_alsa.c and chan_oss.c, I see that the buffer is set to
60ms; way lower than what I'm experiencing.
I'm really interesting in what other people get when running the echo
test. Is this just me or is it a bug?.
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