You will need to setup 2 sip profiles for this setup, one for each interface. Mike On Oct 2, 2009, at 4:10 AM, Timur Irmatov wrote:
Hi.
We have a local network 192.168.1.0/24, where all the users are. Out FreeSWITCH server is connected to this network, and has ip address 192.168.1.242. Through different network card it is connected to external gateway, and has address 172.16.12.11 in this network.
I set up a test client with softphone. When incoming call is deliviered to this client, call is set up normally, but client can't hang it up. It sends BYE to external address - 172.16.12.11 - which is not reachable from the client. It seems this address is coming from Contact: field in INVITE that FreeSWITCH sends:
U 192.168.1.242:5060 -> 192.168.1.34:37169 INVITE sip:100@192.168.1.34:37169 SIP/2.0. Via: SIP/2.0/UDP 172.16.12.11;rport;branch=z9hG4bKrvp6jm3myyaaF. Max-Forwards: 70. From: "FreeSWITCH" <sip:0000000000@172.16.12.11>;tag=v817pS9c6v6Fe. To: <sip:100@192.168.1.34:37169>. Call-ID: 797bd088-29cd-122d-9b93-0060979d54c5. CSeq: 121117089 INVITE. Contact: <sip:mod_sofia@172.16.12.11:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14898. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 267. Remote-Party-ID: "FreeSWITCH" <sip:0000000000@172.16.12.11>;party=calling;screen=yes;privacy=off.
What should I tweak in freeswitch to change this behaviour?