7 Mar
2006
7 Mar
'06
5:25 p.m.
----- Mike Taht <mike.taht(a)gmail.com> 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. 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.
Even though I have done it before, I still have the belief that installing Asterisk
locally is overkill for a single person to make calls. I'm not even sure how an
Asterisk jack channel would function for RTP input to Asterisk. What would do the
signalling?
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....
I'll work on setting up a conference room with a limit of 15 users (that seems high
but what the hell) running on SIP and IAX2 channels. Give me a few weeks.
-lee
7 Mar
7 Mar
9:08 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
On 3/7/06, Lee A. Azzarello <lee(a)rockingtiger.com> wrote:
Even though I have done it before, I still have the
belief that installing
Asterisk locally is overkill for a single person to make calls.
In wanting to bridge the world of Linux audio and asterisk telephony I
had desires far beyond
making a single phone call. I would argue, first, that a local setup
of "single user" asterisk could be made a lot easier (freeswitch is
veering in that direction) and the flexibility of having a pbx on your
laptop or wherever (local voicemail. call presence information.
Text-to-speech support. Etc)
In particular I wanted to make it easier to do a call in radio station
(see rivendell) and integrating the voice to mp3 function and music to
mp3 function struck me as asterisk with jack as a natural bridge...
If ladspa plugins could be run through asterisk or a jack compliant
sip phone you could give your outgoing voice calls a little bass boost
for that "voice of god" effect...
surround sound conferencing becomes feasible.
stuff like that. It doesn't make sense to me that these two worlds -
telephony and professional audio - should be seperated.
I haven't had much spare time recently to work on these ideas, but
freeswitch seems to be a bit more hackable than asterisk has become,
so I've been looking at that...
I'm not even
sure how an Asterisk jack channel would function for RTP input to Asterisk.
What would do the signalling?
Mentally to me, a jack port is a inband telephone connection, no real
signalling save perhaps silence suppression need be used... DTMF, etc,
generated in band...
--
Mike Taht
PostCards From the Bleeding Edge
http://the-edge.blogspot.com
9:25 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
On 8 Mar 2006, at 01:08, Mike Taht wrote:
That would be a pair of jack ports, and to be at all usable in a
radio context it needs
to support at least "ring indicator" and ideally call termination
detection.
Being able to set up (and terminate) calls would also be kind of nice.
Perhaps a daemon that could connect multiple sip "lines" to jackd and
provided a couple of fifos
to communicate line status and to handle dialing?
This is something that has also been on my todo list for a while
(with exactly the same intended use).....
Regards, Dan.
In particular I wanted to make it easier to do a call in radio station
(see rivendell) and integrating the voice to mp3 function and music to
mp3 function struck me as asterisk with jack as a natural bridge...
If ladspa plugins could be run through asterisk or a jack compliant
sip phone you could give your outgoing voice calls a little bass boost
for that "voice of god" effect...
A sip <-> jack "hybrid" would be way cool, but while that covers the
audio
side of the problem, it leaves the call setup and control side.
<Snip>
I haven't had much spare time recently to work on these ideas, but
freeswitch seems to be a bit more hackable than asterisk has become,
so I've been looking at that...
So many potential programs, so little time....
I know that one.
I'm not
even
sure how an Asterisk jack channel would function for RTP input to
Asterisk.
What would do the signalling?
Mentally to me, a jack port is a inband telephone connection, no real
signalling save perhaps silence suppression need be used... DTMF, etc,
generated in band... 
9:32 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
On Wed, 2006-03-08 at 01:25 +0000, Dan Mills wrote:
That would be a pair of jack ports, and to be at all
usable in a
radio context it needs
to support at least "ring indicator" and ideally call termination
detection.
Being able to set up (and terminate) calls would also be kind of nice.
Perhaps a daemon that could connect multiple sip "lines" to jackd
and
provided a couple of fifos
to communicate line status and to handle dialing?
SIP over MIDI?
Lee
8 Mar
8 Mar
12:20 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
Dan Mills <dmills(a)spamblock.demon.co.uk> writes:
A sip <-> jack "hybrid" would be way
cool, but while that covers the
audio side of the problem, it leaves the call setup and control
side.
The way this now with freeswitch is that you send messages to the
server. No, not using OSC, though I have mentioned it to them;). They
are using a simple text protocol over Jabber. With this you can
transfer your call to a conference room, then call other participants
and raise and lower their volumes, f.ex, just by sending these
messages. (bind them to a midi controller)
Perhaps a daemon that could connect multiple sip
"lines" to jackd
and provided a couple of fifos to communicate line status and to
handle dialing?
This is not really necessary as you really only need one bus to send
and one to receive, using ardour;) covering the mic and different
feeds with a midi controller (like feed0, feed1) if you do all
call management on the server, by sending these messages.
--
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
3:42 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
On 8 Mar 2006, at 18:10, Esben Stien wrote:
<Snip>
How do you handle "Talk to caller" for say line 2, while the remote
sports
report is going to air fed from line one?
IMHO You actually do need separate pairs of ports for each line into
the "desk" to make this work in all but trivial cases....
Regards, Dan.
Perhaps a
daemon that could connect multiple sip "lines" to jackd
and provided a couple of fifos to communicate line status and to
handle dialing?
This is not really necessary as you really only need one bus to send
and one to receive, using ardour;) covering the mic and different
feeds with a midi controller (like feed0, feed1) if you do all
call management on the server, by sending these messages.
4:42 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
Dan Mills <dmills(a)spamblock.demon.co.uk> writes:
How do you handle "Talk to caller" for say
line 2, while the remote
sports report is going to air fed from line one?
Well, air is just another output on ardour, but I see what you are
saying. If you got some feed coming in on line 2 from VOIP while you
want to talk to someone on line 1, then you really need such a daemon.
You're absolutely right and I was too quick.
--
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
5:03 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
On 8 Mar 2006, at 22:31, Esben Stien wrote:
Dan Mills <dmills(a)spamblock.demon.co.uk>
writes:
Your using ardour as a broadcast mixer??
Shudder!
Not that ardour is not a very very cool program, but it strikes me as so
wrong for this application (no automated monitor ducking, no fader
start,
insufficient PFL routing flexibility, no talk to caller.....).
IMHO There is a reason recording studio desks and radio production
desks are different!
Regards, Dan (Now contemplating writing software radio desk).
How do you handle "Talk to caller" for
say line 2, while the remote
sports report is going to air fed from line one?
Well, air is just another output on ardour, but I see what you are
saying. If you got some feed coming in on line 2 from VOIP while you
want to talk to someone on line 1, then you really need such a daemon. 
3:52 a.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
Hi,
On Tue, 7 Mar 2006, Mike Taht wrote:
In particular I wanted to make it easier to do a call
in radio station
(see rivendell) and integrating the voice to mp3 function and music to
mp3 function struck me as asterisk with jack as a natural bridge...
If ladspa plugins could be run through asterisk or a jack compliant
sip phone you could give your outgoing voice calls a little bass boost
for that "voice of god" effect...
I'm currently working with the sofia-sip project and we have a couple of
test clients that use gstreamer for media and rtp. Gstreamer's JACK
support is unfortunately not quite up to date, but at least it has LADSPA
support, and an architecture that makes it easy to add new processing
elements to the voice send/receive chains.
- http://sofia-sip.sourceforge.net/
- http://sofia-sip.sourceforge.net/cgi-bin/twiki-bin/view/SofiaSIP/SofiaAppli…
.. see the Gaim SIP plugin, Farsight/Telepathy and Tapioca apps
These are so far just clients of course, but you can use them to connect
to asterisk or to conference bridges (or use the libraries to build
your own and/or enhance asterisk).
Btw, in true ecasound-fashion ;), I'm working on a state-of-art,
console-mode SIP VoIP/IM client that uses gstreamer for RTP and media (and
hopefully JACK support at some point):
- http://sofia-sip.sourceforge.net/cgi-bin/twiki-bin/view/SofiaSIP/SofSipCli
stuff like that. It doesn't make sense to me that
these two worlds -
telephony and professional audio - should be seperated.
True, lot of the audio routing and processing functionality can be used in
telephony apps, while transport of media over wide area networks is of
interest to the audio crowd as well.
--
http://www.eca.cx
Audio software for Linux!
12:05 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
"Mike Taht" <mike.taht(a)gmail.com> writes:
a call in radio station
This is my idea as well. I want listeners to call in using VOIP.
If ladspa plugins could be run through asterisk or a
jack compliant
sip phone you could give your outgoing voice calls a little bass boost
for that "voice of god" effect...
This is fully possible and it's working without problems for me using
oss2jack.
--
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
12:01 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
"Lee A. Azzarello" <lee(a)rockingtiger.com> writes:
not (..) sure how an Asterisk jack channel would
function for RTP
input to Asterisk. What would do the signalling?
Not sure what you mean. You don't stream anything directly to asterisk
without going through a stack.
I'll work on setting up a conference room with a
limit of 15 users
Excellent.
--
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
6:47 p.m.
New subject: [linux-audio-user] Re: GNU Audio Community Conference Room
"Lee A. Azzarello" <lee(a)rockingtiger.com> writes:
Give me a few weeks.
Situation Report?;).
--
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
6816
days inactive
6837
days old
linux-audio-user@lists.linuxaudio.org
11 comments
6 participants
participants (6)
-
Dan Mills -
Esben Stien -
Kai Vehmanen -
Lee A. Azzarello -
Lee Revell -
Mike Taht