asterisk users - Oct 2006 - random one way audio and noise between SIP phones on same LAN

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

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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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 
> every log file but found nothing.
Plug everything into a hub, not a switch, and then run tcpdump to look at
the traffic going back and forth:

# tcpdump -i eth0 -n -s0 -v

Look at the RTP packets, in particular source and destination IP addresses.
Do they go phone1->phone2 and phone2->phone1 ? Or do they go
phone1->PBX,
PBX->phone2, phone2->PBX and PBX->Phone1 ?

In the latter case, it's not the phones which are at fault, it's your
PBX.

If the PBX is Asterisk, and you want help solving the problem, then you'd
better give full details of your system (hardware and O/S platform, asterisk
version, how it's configured)

If you want the streams to go direct between phones, and the PBX is
Asterisk, then set "canreinvite=yes" in sip.conf
> Considering to monitor a remote PBX via ssh, which  text-only 
> application  could  I use?
Depends what the PBX is. If it's something with a serial console, then
minicom, tip or cu will let you talk to it. If it's asterisk, then use the
asterisk console (asterisk -r)

HTH,

Brian.