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

[Holger Latz]

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