Hello all, I need to setup a new asterisk system with the following requirements:
1. Will be moving from chan_sccp to sip (7960's), but I want to support the sccp phones until everyone has been migrated. 2. Need to maintain current portability of the 7960's. (ie a user can unplug his phone from the internal LAN, take it home or wherever, and plugin and have the phone register and work just like in the office. I have tried the following: 1. Asterisk server on LAN behind NAT, LAN phones on the same net, tested a phone with a public IP. 2. Asterisk on Public IP, LAN phones on LAN behind NAT, didn't even get to testing remote phones. I am having trouble with calls completing but not passing the audio stream. I have done fixup SIP/opened 5060/tried many settings in SIP.conf/set 7960's to NAT=YES etc. I can not NAT each phone individually and allow RTP to it, as I saw one person did. I really don't want to run asterisk on a public IP and a LAN IP going around my firewall. Do I need to put the phones on a separate LAN network and run asterisk on a public ip and private? Do I need to run a SIP proxy. I looked at SER/OPENSER, but it seems to break some things. (Need to be able to record all calls need MWI) Should I run 2 asterisk boxes connected with maybe TDMoE? Would that work? Any suggestions would be greatly appreciated. Thanks, Andy Hester _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
