Hi,<br><br> I have the same issue. Did you solved it?<br><br><br><div class="gmail_quote">On Thu, Jun 18, 2009 at 5:21 PM, Brad Johnson <span dir="ltr"><<a href="mailto:bjohnson@ecessa.com">bjohnson@ecessa.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">We are trying to configure Asterisk (version 1.6.1.0) with some SIP<br>
phones behind a SIP Proxy/NAT device. The phones register properly to<br>
Asterisk, and to get Asterisk to register properly to the external SIP<br>
registrar we added this to the general section of sip.conf (the address<br>
of the Asterisk system on the LAN is 192.168.30.5):<br>
<br>
outboundproxy=192.168.30.10<br>
register => <a href="http://myname:mysecret@my.provider.com/100" target="_blank">myname:mysecret@my.provider.com/100</a><br>
<br>
The problem we are facing is that it appears that the outboundproxy<br>
value is being treated globally by Asterisk so it sends all SIP traffic,<br>
including traffic to the phones, to the proxy. The behavior we want is<br>
that all outbound traffic is sent to the proxy, but inbound SIP traffic<br>
to the phones should be sent direct to the phones.<br>
The result we see is that an inbound Invite is received by Asterisk and<br>
then the Invite for the phone is sent by Asterisk to the outbound proxy.<br>
This causes much confusion.<br>
Can anyone please tell me how to configure Asterisk properly for working<br>
behind a SIP Proxy?<br>
Below you will find our configuration.<br>
<br>
Thanks,<br>
Brad<br>
<br>
Here is the channel for our SIP provider:<br>
<br>
[my_provider]<br>
type=peer<br>
host=<a href="http://my.provider.com" target="_blank">my.provider.com</a><br>
username=100-phone<br>
secret=mysecret<br>
context=incoming<br>
canreinvite=no<br>
qualify=300<br>
insecure=port,invite<br>
<br>
Here is a sample phone entry in sip.conf:<br>
<br>
[100_phone]<br>
type=friend<br>
username=100-phone<br>
secret=100secret<br>
host=dynamic<br>
context=internal<br>
<br>
Here is the relevant part of extensions.conf:<br>
<br>
[incoming]<br>
exten => 100,1,Dial(SIP/100_phone,30)<br>
exten => 100,n,Hangup()<br>
<br>
[internal]<br>
exten => _X.,1,Dial(SIP/my_provider/${EXTEN})<br>
<br>
<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Best Regards,<br>Giedrius<br>