[Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Steven Ringwald]

Steve wrote:

>On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote:
>  
>
>>I'm using Asterisk on a open server (no firewall or NAT) and trying to
>>communicate with a Grandstream BudgeTone 102 SIP phone which is behind
>>NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS
>>about a week ago.  My problem is that I'm only getting half-duplex
>>communication -- I can hear voice from the Asterisk server but the server
>>does not understand any voice from me.  From the console "sip debug" shows
>>that the SIP part is working fine and DTMF via SIP INFO works.
>>    
>>
>
>
>I use OpenBSD firewalls with NAT and redirect and it works just as it's 
>supposed to. 
>
>That's not even half duplex. In half duplex each side Can talk, but only one 
>at a time. It seems to be an error with configuring your firewall. (One 
>common error is to only turn on redirect. But you also need to Allow the 
>traffic to flow...
>  
>

I am having problems similar to Owen's. Just for grins, can you tell me 
which ports you opened up? I opened the following:

tcp 4569 192.168.2.212
udp 4569 192.168.2.212
udp 5036 192.168.2.212
udp 5060 192.168.2.212
tcp 20000:21000 192.168.2.212
udp 20000:21000 192.168.2.212

192.168.2.212 is the IP of the Asterisk box within my firewall. I have 
no trouble connecting to it on the local LAN, but if I go remote, it 
always wants to connect via a bridge connection to the other BudgeTone 
phone.

<sip.conf>
[ringwald]
fromuser=ringwald
disallow=all
host=dynamic
allow=ulaw
type=friend
username=ringwald
secret=MySecret
canreinvite=no
reinvite=no
nat=yes
dtmfmode=inband         ; Choices are inband, rfc2833, or info

Any help that you can provide would be greatly appreciated.

Steve Ringwald


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