Hi,

I just upgraded our CallWeaver box to revision 4571 and encountered
problems with calls on the local network. I have externip and localnet
set, but when I place calls to an Asterisk box (this worked before) in
the same subnet the call goes on for 20 seconds and is disconnected due
to rtptimeout. I can see from SIP debug that it believes it is "reliably
transmitting (NAT)", but the peers are set as nat=no and there is no NAT
between the two servers. If I issue rtp debug ip 10.100.4.190 I see "Got
RTP packet from 10.100.4.190:17430 (type 8, seq 4573, ts 231520, len
160)" and the screen scrolling, however there are no "Sent RTP packet to
10.100.4.190..." What has happened here? Is the networking support in
the SIP channel broken again?

My last revision was 4215 and crashed randomly, but users on the local
network as well as external users could place calls to the other server.
The configuration is the same.


Mikael

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