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

[Moshe Yudkowsky]

So, as usual, about 30 seconds after I send off the message I come to the 
realization that I've done a Dumb User Trick (aka "operator error").

* noload of chan_oss.so & chan_alsa.so does disable the console.

* I can call from a softphone on the PBX to the PBX's voicemail and 
transmit audio!

* But -- I cannot transmit or receive audio from a softphone on a different 
machine. I can connect, but there's no audio sent or recevied to the PC.

They're all behind the same firewall, the debug output shows that the RTP 
port is correct. Any ideas for something else simple I've missed?>



At 12:16 2003-06-30 -0500, Steven Critchfield 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

  www.Disaggregate.com
  speech at pobox.com
  +1 773 764 8727