Which is what I was finding out, but once I set it to the DMZ everything was fine seen as I didn't have to worry about the ports at all.
Now if I could only get video to work again I'll be all set! I'll have to look into the IAX2 protocol also. -blake On 7/7/05, Rich Adamson <[EMAIL PROTECTED]> wrote: > > You can try to open up port for SIP 5060udp and RTP 100000-20000udp... > > (default setting) to your asterisk box. You will also have to specify > > that your extensions are nat=yes & your externip=xxx.xxx.xxx.xxx (in > > SIP.conf) so that the SDP protocol will write the public IP and port > > translations for RTP (voice data). If this doesn't work, switch to > > IAX2 protocol- there are many hard-phones out there that support IAX2 > > protocol- You will only have to open up 4569udp on your firewall to > > your asterisk box and thats it. > > Better be careful with the RTP statement above as its not necessarily > true for all implementations and configurations. > > If asterisk initiates the RTP negotiation, "it" will use udp source > ports from the range shown above. However, each sip phone vendor (hard > or soft) can choose whatever port range they want. > XLite is in the 8,000 range > Cisco 79x0's are in the 16384 to 32766 range > etc. > > If a remote device initiates the RTP negotiation, it may not fall into > the range that you've stated. (E.g., don't bank on your favorite itsp > falling into that range.) > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
