[asterisk-users] RTP/NAT Traffic to private IP [SOLVED]

[Holger Latz]

Ok, I found the problem. I suggested that I disabled completely my 
shorewall-firewall, because there were no rules loaded. But I were 
mistaken... shorewall loads some kernel-modules, especially ip_nat_sip 
and ip_conntrack_sip, and these modules interfere with asterisk!

http://www.mail-archive.com/shorewall-users@lists.sourceforge.net/msg03968.html

Regards
Holger


Holger Latz schrieb:
> Hi all,
> 
> I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone 
> in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the 
> phone is ringing, but when I pickup the call, there's no audio on both 
> sides.
> 
> I debugged the rtp-traffic at home. As long as the phone is ringing, 
> everything is fine. But after the pickup, asterisk sends a SIP/SDP 
> package with its private address (192.168.100.10). After the softphone 
> received this package, it tries to send RTP data to this address! 
> Obviously those packages never reach asterisk...
> 
> Does 'externip' just works for SIP and not for RTP?
> Where does the the internal IP-address come from and how can I set the 
> right one?
> 
> 
> My configuration:
> 
> [general]
> externip = 85.XXX.XXX.XXX
> nat = yes
> localnet = 192.168.100.0/24
> 
> [42]
> deny=0.0.0.0/0.0.0.0
> disallow=all
> type=friend
> secret=XXX
> qualify=yes
> port=5060
> pickupgroup=
> permit=0.0.0.0/0.0.0.0
> nat=yes
> mailbox=42 at device
> host=dynamic
> dtmfmode=rfc2833
> dial=SIP/42
> context=from-internal
> canreinvite=no
> callgroup=
> callerid=device <42>
> allow=alaw
> accountcode=
> call-limit=50
> 
> 
> Regards
> Holger
> 
> 
> 
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