Displaying 20 results from an estimated 11000 matches similar to: "Connections, but no voice paths except by console"
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
2003 Jun 24
2
Asterisk ALSA module not working
Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The
module chan_alsa.so won't load even if the oss module, chan_oss.so,
isn't loaded. There are no error messages.
I've been chasing ALSA/Asterisk/client problems in one form or another
for some time now. In previous versions of Asterisk and ALSA -- i.e.,
last week -- I could load either chan_oss.so or
2003 Jun 13
5
Applications, dialplan not loading
I've built the latest CVS of asterisk -- not the zaptel or libpri
directories, just the asterisk directory. asterisk installs
successfully, but there are severe problems. I built this system in the
past and ran it, but now building it again fails. This is the CVS as of
this morning, 2003-06-13, but I had problems on 06-11/12 as well.
After make; make install; make samples; make config, I
2003 Jun 15
5
.gsm files
Hi guys,
Being a true Linux geek, I've never been too much into sounds or sound
files other than a few .mp3 songs I got. My question is pretty
straightforward and simple. I see that the music format of choice for
asterisk is .gsm. What can I use to listen to files in .gsm format and
what is the most effective way of recording files into .gsm format?
The last part of the question is
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
2003 Jun 13
5
Disabled echo canceller because of tone (rx)
Does anyone know what this means? It is in DMESG, and we have
people complaining about echo.
Disabled echo canceller because of tone (rx)
John
2004 Dec 07
1
asterisk and kphone (sip soft phone for linux) on same machine
Hi, i just installed latest asterisk on fedora rc2 and on the same
machine i installed a sip soft phone called kphone. Kphone complains
about /dev/dsp being used and can't place/answer calls (/dev/dsp is
obviously used by asterisk) . how can "share" my sound card with these
two programs?
or
can i disable the sound card in asterisk so i can use kphone to
place/answer calls?
BTW kphone
2004 Sep 14
1
Requested device 'ttyI1' does not exist
Hello List!
I finally got asterisk with capi working, and its already answering my
call as well! :)
Now i would like to call a number from my shoft phone (kphone).
This is my extentions.conf:
---
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
2005 May 04
4
Problem with realtime SIP
Hi Guys,
We have just set up Asterisk (CVS Head) for a realtime enviorment using
MySQL & Asterisk Addons.
I have populated the "sip_buddies" table with the same information that
is came from our sip.conf, however registration seems to fail for the
softphone we have set up.
Does anyone have any idea as to what I should be looking for here? I'm
not getting any error messages
2003 Jul 15
5
Text to Speech - Someone needs to do this
Why hasn't someone found 50 people who sound alike, put them in sound
studios and record the 10,000 most commonly used words. You would all
differnent forms of the 1,000 most words, i.e. leading, trailing, question
etc.
You can synthesize the other 0.05% when you run into them. With hard drives
so big, processors so fast and EXT3 that can handle 30,000+ files in a
single directory that
2005 Jun 17
6
Console ALSA Sound
Hi
... probably one of those RTFM kind of questions (while I'd be happy to know
where a good reference "FM" is :-) )
Has anyone an idea on how to disable the console sound driver. My problem is
that a running asterisk is muting my speakers.
Thank you in advance for your help
Conrad
2005 Mar 16
2
Basical question to asterisk
Hello!
I'm new to asterisk and because I try to configure the package for my
needs the last days without success, I'd like to ask a basical qestion.
I need asterisk to work together with the German VoIP provider sipgate
(http://www.sipgate.de). Asterisk should act as a softphone, I want to
recive and make calls only with the software under linux, no softphone
should be used. Is this
2004 Jun 23
5
Really basic stuff :(
Hi :)
I've had all this working before, but I'm revisiting it, and in short, I
currently have huge problems receiving incoming calls. I've been trying
with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel
and libpri as of yesterday afternoon.
Would someone mind helping? :)
My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set
as the 'DMZ
2003 Jun 13
1
strace shows that files are not accessed
strace on file access in asterisk shows that * is not even attempting to
access the voice files.
If I *manually* load app_playback.so, app_macro.so, and then
pbx_config.so, I they will load and I get a dialplan. Ok, that's a
problem -- autoconf is clearly not working, or there's some other
related issue.
So I try to use the demo and do "dial 500". This should connect and
2003 Mar 09
2
How to play sound AND run asterisk?
Hi,
I'm a new asterisk user developing an AGI application. As part of my
application I'd like to play sounds on the server's speakers, but it seems
that I can't do this while asterisk is running.
When I try to play sounds using the play or aplay command, it blocks until
I stop asterisk. My guess is that asterisk is using the sound device and
this means that other programs
2008 Mar 04
1
console dsp
I am trying to get a console/dsp application going with
1.4.18 and not hearing any audio. In the CLI I see the call coming in,
I see the Dial(Console/dsp)
I see <auto answered>
I see ALSA default
but I hear no audio.
What can I do to tell what is happening here.
I have in modules.conf:
noload chan_oss.so
load chan_alsa.so
For kicks I tried it the other way to noload chan_alsa.so and load
2004 Aug 16
2
disable console channels
I have a Digium TDM400 in my system and I'm using my main system as my
asterisk box at home (very light load).
When I start up *, though, it grabs my sound card and I cannot play other
music through it (e.g. x/ XMMS). I have moved the alsa.conf and oss.conf
files so that there is no configuration for them (though those files seemed to
do little), but still the sound card is grabbed.
How can
2016 Aug 16
2
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 04:38, George Joseph wrote:
>
>
> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> Hello
>
> using pjproject 2.5.5
> using asterisk-certified-13.8-cert1
>
>
> IIRC there were API changes in pjproject 2.5 that aren't accounted for
> in
2004 Jul 15
1
Fedora Core 2 softphone
Hello all,
I am in the process of converting our company over to * to replace our
ancient executone system. As part of the testing process my boss wants
us to all run softphones on our desktops until he gets the phones
ordered. Quite a few of us run fedora core 2, and I haven't had any luck
getting a soft phone to work. Kphone works the best out of all I have
tried but I get no sound out of