On Tue, 1 Feb 2005 08:08:50 -0600 (CST), [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > I'd like to open up my firwall so that I can connect my SIP phones to a > test server behind or firewall. I can configure an outside addtess to pass > traffic to the internal address of the Asterisk server. I'm not sure what > other ports need to be opened. My SIP phone will either be at my home > behind a linksys cable/dsl router or perhaps at a hotel when I'm on the > road. What am I missing in this configuration? Is STUN needed? > > Pat >
looks to me that it would be easier to prchase a vpn lic...... I've tried some of the suggestions on the list to no avail. Until version * 1.03 came out... I still had echo/cutouts through the vpn. Now so far <knock on wood repeatadly> it is going ok. Here is what I found on the list: It works fine for me. I have a handful of Cisco 7960's behind a PIX firewall and they register to a Asterisk server outside of the PIX with no trouble at all. I didn't do anything special to the PIX (i.e. no access list entries). The tricks I found to make it work generally apply to any setup where the clients are behind NAT. I also run the tftp server for the phones to get configs inside the firewall, and the SIPDefault.cnf file specifies the proxy address outside of the firewall. In the Cisco phone config I have these NAT settings: nat_enable: 1 ; 0-Disabled (default), 1-Enabled nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled And the sip.conf entry for this peer is: [7000] type=friend nat=yes qualify=yes context=xxxx secret=xxxx callerid=xxxx host=dynamic canreinvite=no dtmfmode=rfc2833 timer_register_expires: 120 Setting the registry timer to 120 seconds causes the phone to send out a packet at least every 2 minutes which will open a UDP xlate on the PIX for the session. Then the trick is to use both 'nat=yes' and 'qualify=yes' so Asterisk chats with the phone pretty often. The interval of OPTIONS or REGISTER messages between Asterisk and phone definitely needs to be shorter than the PIX's UDP xlate timeout or the PIX will close the xlate and you won't be able to pass packets into the phone for an incoming call. Note that you can put a numeric value after qualify= instead of "yes" to fine-tine the interval at which it sends a OPTIONS message. Good Luck! t o n y _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users