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