Hi all,
I am trying to connect to a friend via sipphone using linphone or
phonegaim.
In either case the phone rings but I can't hear him and he can't hear
me.
I found what are the probable causes from the linphone docs, but I
can't figure out what to do with this information and am asking some
help.
I am connected to the net by cable and actcom
Audio problems
Linphone seems to connect to the remote sip url, it rings, but
when the callee answers, nothing happens and we can't hear each
other.
Most people get problems because they don't choose the correct network
interface in the property box, section
network. For a dialup connection, it should be "ppp0". Note also that the "lo"
interface SHOULD ONLY be used
for testing with sipomatic. In other cases, it will fail.
So the choices are ethernet, adsl,cable, and various modems 56
and 96 I think. But with Isreal we connect via ppp, so which
is right adsl, ethernet or a modem?
First rise up playback and recording level.
This didn't help
If the voice is sometines cutted, you can modify parameter RTP->jitter
compensation in the property box to
greater values to avoid this. But it increases the delay transmission.
this isn't the ca*
If linphone cannot open the audio device, check if it has the permission to
open /dev/dsp, close all programs able to use audio
device (xmms, kaiman...).
*
Here is my main question how do I make sure that linphone and
phone gaim can write to my audio device /dev/dsp.
I also want a way to see if other devices are calling /dev/dsp
especially arts
Anyone tried this and succeeded?
Thanks
Aaron
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