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On Mon, 2008-02-18 at 08:35 +0100, Mikael Aleksander Bjerkeland wrote:
> 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.
If 4571 doesn't work and 4215 does, a regression test would be useful
(ie. binary search versions in between until you find the commit which
broke it).
Also, nat=no doesn't completely disable NAT, use nat=never.
>
> Mikael
>
> _______________________________________________
> Callweaver-users mailing list
> [email protected]
> http://lists.callweaver.org/mailman/listinfo/callweaver-users
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