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Steve wrote:<br>
<blockquote cite="mid200401111924.46880.steve@szmidt.org" type="cite">
<pre wrap="">On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote:
</pre>
<blockquote type="cite">
<pre wrap="">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.
</pre>
</blockquote>
<pre wrap=""><!---->
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...
</pre>
</blockquote>
<br>
I am having problems similar to Owen's. Just for grins, can you tell me
which ports you opened up? I opened the following:<br>
<br>
tcp 4569 192.168.2.212<br>
udp 4569 192.168.2.212<br>
udp 5036 192.168.2.212<br>
udp 5060 192.168.2.212<br>
tcp 20000:21000 192.168.2.212<br>
udp 20000:21000 192.168.2.212<br>
<br>
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.<br>
<br>
<sip.conf><br>
[ringwald]<br>
fromuser=ringwald<br>
disallow=all<br>
host=dynamic<br>
allow=ulaw<br>
type=friend<br>
username=ringwald<br>
secret=MySecret<br>
canreinvite=no<br>
reinvite=no<br>
nat=yes<br>
dtmfmode=inband ; Choices are inband, rfc2833, or info<br>
<br>
Any help that you can provide would be greatly appreciated.<br>
<br>
Steve Ringwald<br>
<br>
<br>
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