[asterisk-users] random one way audio and noise between SIP phones on same LAN

[Giorgio Incantalupo]

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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