[Asterisk-Users] voice problems on console

[Juha Heinanen]

i did some more testing with console sip ua.  if i call from * to kphone
using gsm codec, voice works fine both ways, but if the call comes from
kphone, voice works only from kphone to * (although gsm packets fly ok
both directions).

i also tried with g711u codec and when i call from kphone to *, there is
no voice from kphone to * (not even ringing tone).  when i call from *
to kphone, there is a huge delay from kphone to * direction and other
direction works fine.

i don't know what is going on.  sometimes i saw on console an error
message

WARNING[14350]: File chan_oss.c, Line 675 (oss_indicate): Don't know how to display condition -1 on OSS/dsp

may be the problems are sound card/module dependent.  i did my tests in
via/opensound.com environment.

-- juha