>I am using Polycom SP300 phones. You have to separate 'user' and 'peer' part of it to >>>get it working. Search the wiki for description of the problem.
Nice to know ... I don't own any of those but its good general knowledge. >You have to forward port 5060 so that phone from outside can register and call. And >>>>>ports 10000-20000 do that voice can go through. Actual port ranfge is isn filr >>>>>>rtp.conf.> 10000-20000 is the default range Ive done this on the firewall infront of our * box. >Yes, only port 5060. If you do not forward 5060, you can not call this phone >from outside. Seem to work OK without other ports being forwarded. You mean on the remote sip phone firewall? What if there arem ore than 1 sip phone on that network behidn that firewall? Don't you need to forward ports 10000-20000 for voice? Or does the sip phones just open up the ports from inside (by doing the in to out calls and keep alives)? _______________________________________________ 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