asterisk users - Jun 2003 - Connections, but no voice paths except by console

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?

-- 
  Moshe Yudkowsky * http://www.Disaggregate.com

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@basesys.com>

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@basesys.com>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
  Moshe Yudkowsky
  Disaggregate
  2952 W Fargo
  Chicago, IL 60645 USA

  <http://www.Disaggregate.com>

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@basesys.com>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
  Moshe Yudkowsky
  Disaggregate
  2952 W Fargo
  Chicago, IL 60645 USA

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