On Wed, Oct 18, 2006 at 01:44:06PM +0200, Giorgio Incantalupo wrote:
> sometimes I have one way calls and noise between sip phones connected to
> the same LAN so no nat/firewall is involved. I tried with different sip
> phone models soft phones and the result is the same. I searched inside
> eve
Hi Klaus,
I tried to set canreinvite=no following wiki advices but nothing
changed. But I do not have a nat because I'm on a LAN.
Is there any kind of program to control rtp packets flow? I think SIP
conversation is established but RTP packets do not pass. I've tried to
open up all ports (from
Do you use canreinvite (sip.conf)?
Change the setting (setting canreinvite=yes may cause nat problems) nad
verify if the problem still appears.
Using htis setting you can find out if the Audio problem occurs only
when media is relayed via Asterisk (->the problem is caused by Asterisk)
or in
Hi,
sometimes I have one way calls and noise between sip phones connected to
the same LAN so no nat/firewall is involved. I tried with different sip
phone models soft phones and the result is the same. I searched inside
every log file but found nothing. I made different PBX with different
hard
