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

[Gondar Monn]

I have no problem doing that by adding the information you have under
[general] to /etc/asterisk/sip_nat.conf

On Thu, Jan 29, 2009 at 7:30 AM, Holger Latz <tech at globalview.de> wrote:

>
> 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<http://0.0.0.0/0.0.0.0disallow=alltype=friendsecret=XXXqualify=yesport=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<http://0.0.0.0/0.0.0.0nat=yesmailbox=42@devicehost=dynamicdtmfmode=rfc2833dial=SIP/42context=from-internalcanreinvite=no>
> callgroup=
> callerid=device <42>
> allow=alaw
> accountcode=
> call-limit=50
>
>
> Regards
> Holger
>
>
>
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