[Asterisk-Users] Connections, but no voice paths except by console

Moshe Yudkowsky speech at pobox.com
Mon Jun 30 12:32:00 MST 2003


Steven,

As per your advice, I have tried to not load the console -- which in this 
case, if I understand correctly, would be the chan_alsa.so and chan_oss.so 
modules.

* I never load chan_alsa.so, because chan_alsa doesn't work for me and I 
haven't figured out what the problem is just yet.

* If I unload chan_oss.so from a running system, I crash on the first call. 
The system shouldn't do that, I expect.

* If I noload chan_oss.so, then I can call from a softphone to, e.g., 
voicemail -- but again I cannot transmit audio, only hear it on the softphone.

A softphone call from the PC to voicemail connects, but cannot transmit or 
receive audio.

I will guess that I'm missing something obvious -- that the console is some 
other application?


At 12:16 2003-06-30 -0500, you wrote:
>If you have the console working, then you can't use a softphone on the
>same machine. If you want to test with a softphone, set the console
>driver to noload and try again. This would probably be the same for your
>dialing to other SIP phones.
>
>I'm guessing that ALSA doesn't allow more than one connection to the mic
>and asterisk already has it. Alsa I think allows multiple connections to
>the speakers though.
>
>On Mon, 2003-06-30 at 11:28, Moshe Yudkowsky wrote:
> > I have a software-only PBX set up. I can register various softphones and
> > they will call each other -- but I've never succeeded in getting any
> > voice routed from any of the softphones. Only the console will transmit
> > audio.
> >
> > I am writing to ask if I have missed some obvious step in configuring
> > the system.
> >
> > Conditions:
> >
> > (1) Softphones running on the same machine as the PBX: Only Kphone seems
> > to work reliably. Kphone will register and connect, but if I dial a
> > different softphone on the same machine and get routed to voice mail, I
> > hear the voice mail announcement but I am unable to leave a voice mail
> > messsage. The reason is explicit on the debug output (I nearly used the
> > phrase "ROP"!), which says that the voicemail SIP connection is dropped
> > because there's been no input.
> >
> > If I reach the other softphone, I cannot transmit from one to the other.
> >
> > Is the inability to transmit part of the continuing problems between
> > Asterisk and ALSA?
> >
> > (2) If I dial from the console into voicemail, I can hear and transmit.
> > If I dial to/from the console to a softphone, then I can transmit audio
> > from the console to the softphone (apparently -- there's no way I can
> > see to debug this to determine who is getting audio from whom).
> >
> > (3) Softphones on a different PC: Using X-Lite from my (*shudder*)
> > Windows box, I can connect to the console on my PBX, or be routed to
> > voicemail. In either case, I cannot transmit audio in *either*
> > direction, from the PC to the console, or the console to the PC. E.g.,
> > the softphone does not hear the audio output of the voicemail announcement.
> >
> > If I let the PC's softphone go to voicemaill, the debug output shows
> > that * drops the call because there's no audio.
> >
> > I'm deducing that for some reason I an not routing *any* audio via SIP.
> > Is there some configuration issue I'm missing?
>--
>Steven Critchfield  <critch at basesys.com>
>
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-- 
  Moshe Yudkowsky
  Disaggregate
  2952 W Fargo
  Chicago, IL 60645 USA

  <http://www.Disaggregate.com>




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