[asterisk-users] canreinvite problems

[Luki]

Stefan,

> When I have 2 SIP endpoints that both aren't configured with
> "canreinvite=no" then I get no sound.

The Sipura 3102 definitely works fine with canreinvite=yes and I never
really had a problem with any of the Sipura devices in this respect,
especially when there is no NAT involved. However, the default "Auto
NetService Private IP Ranges:" includes 192.168.0.0-192.168.255.255,
so your 192.168.254.0/24 network would be considered a LAN address by
the 3102 and hence the traffic would go out the LAN interface (not
WAN). Change this setting by removing this range. It's on the Admin >
Advanced > LAN Setup tab.

If that doesn't help, then you need to check what traffic is being
sent. Since all devices are on the same internal network I assume they
can see each other. You need to look at the Invite (and ReInvite)
messages sent and received and see if the IP addresses for RTP listed
there make sense. Then I suggest you use tcpdump to see what traffic
is sent by each device, and where. If you have a switched network
environment this will be a bit tricky as your * box won't see this
traffic, so you may want to use a hub for this test (just temporarily)
or if available set up port mirroring to sniff the traffic.

Good luck and keep us posted.

--Luki