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 1020 to 65535) without success.
These random one way calls ALWAYS happen about 10 times a day on EVERY
PBX I have (on different LANs) and customers complain.
I'm still searching for a monitoring application to catch packets on
remote machines. Hope to find a good one.
TIA
Giorgio Incantalupo
Klaus Darilion wrote:
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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