Asterisk on Freerunner was: voip on Debian
nicola.mfb at gmail.com
Sat Apr 18 17:49:05 CEST 2009
2008/9/6 TL Mieszkowski <mieszkowski at gmail.com>:
> I've had a lot of success running both twinkle and asterisk and I thought I'd
> share my experiences.
> Twinkle works well, but the gui is limiting on the touchscreen. I think
> once configured properly
> asterisk will make an excellent voip backend for the neo. You can control
> it through asterisk
> manager commands by writing text strings to a socket, and which has hooks
> for most languages I'm sure.
Let's survive this interesting topic.
I will be happy to write an AMI gui but now I'm hold having problems
with the alsa channel. Using the pcm default is not compatible with
the default shipped /etc/asound.conf, so I just tried to use
plughw:dnsoop and plughw:dmix, the result is that there freerunner
does not ring on incoming call (and you cannot hear the other peer),
while audio transmitting is perfect. Using plughw:0,0 for input/output
works but I have stuttered audio (from freerunner to peer). I tried
the mentioned asound.conf from koolu too, the same, If i move out from
plughw there is no sound in fr with asterisk. If I use dnsoop form
input and plughw for output, the input is stuttered again. I'm using
shr-testing and asterisk 1.4.17-r1 from the same branch.
As in the old thread there was success story may someone share some hint?
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