This isn't an Asterisk issue, it's a routing issue. Take a look at iproute2 and routing policies.
Another way to view it is that Asterisk hands the communications over to Linux, where the network route takes over. (The * bind statement just tells * what IP to listen on) If you have 3 nic's on the same subnet, you have a routing challenge. Either setup static routes to the subnets/hosts you want (via certain NIC's), or use iproute2 to force traffic out a certain NIC based on port, policies, etc. Michelle ________________________________________ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Mike [l...@virtutel.ca] Sent: Monday, May 31, 2010 10:01 AM To: Asterisk Users List Subject: Re: [asterisk-users] How to use one single IP as origination Sorry, that made no sense, just re-read your problem. I believe Asterisk simply takes the default IP, which would in this case be eth0/first IP (not the virtual IPs) as outgoing IP. Is this a problem? It is for me, I would like to define the IP used per peer, but that’s the way it is, at least on 1.4. I read somewhere (can`t find the page) that 1.6 works differently. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, May 31, 2010 9:55 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to use one single IP as origination See bindaddr here: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf That should do exactly what you want. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR Sent: Sunday, May 30, 2010 10:06 To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to use one single IP as origination I have an Asterisk with multiple IP's, on the same subnet. When a call comes in, I need to send it back out via SIP, but need that only one IP is used as originating IP for all calls. For example machines has 192.168.50.3 192.168.50.4 192.168.50.5 .... but when I originate the second leg of a call, the IP address that is supposed to be read as source IP must be 192.168.50.5, regardless of how the call arrived. How do I do that? -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
