Do you have a range of registration ports configured and forwarded through the firewall on the server end? Ie. 5060-5065 for example.
On the Phone side you should forward 5060 to phone1 and 5061 to phone 2 etc. and configure the phones to use that port for registration. You may need to forward ports for the actual voice as well. 2 ports per phone so 10000-10001 for phone1 and 10002-10003 for phone2. It's either that or mess around with STUN or Proxy servers or whatever. SIP+NAT=headache -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, February 02, 2008 8:23 PM To: [email protected] Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall The server is at a remote datacenter - no nat, no firewall, pure public IP. The phones are at home offices (i.e. DSL or Cable with Linksys-type firewall/routers). My initial testing was with a single SIP phone at the home office - and everything worked fine. But when I have two SIP phones at the home office, things start behaving badly. I understand the issue of phone-to-phone, where both phones are behind a nat at the home office - but that is not the issue I am having. My main problem is when I have two phones at the home office, the second phone cant register, and/or, you cant here the voicemail greeting when you try to check messages. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
