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

[Moshe Yudkowsky]

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