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 all cases (the problem is not caused by Asterisk)
regards
klaus
Giorgio Incantalupo wrote:
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
hardware but the result is always the same.
Is there anybody experiencing this terrible problem?
Considering to monitor a remote PBX via ssh, which text-only
application could I use?
TIA
Giorgio Incantalupo
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