I’m using three NIC cards
eth0 192.168.0.0/24 Intranet
eth1 188.8.131.52/24 TRUNK-1 (default route)
eth2 184.108.40.206/24 TRUNK-2 dedicated
The issue is that each vendor requires their own network to send/receive their SIP communication.
Dedicated TRUNK-2 network is not for general use, but TRUNK-1 is.
General users register their devices using eth0. and eth1.
Routing table shows default to GW 220.127.116.11 on eth1
When placing a call routed to TRUNK-2, the system incorrectly identify the call as coming from TRUNK-1 as originator, however correctly send the packets through TRUNK-2 as expected, the call is placed but a) no signaling (no hangup) b) no audio. Also, incoming call entering through TRUNK-2 are received, you can answer, but no voice nor signaling
Looks like when creating SIP headers, asterisk is not aware of the multiple interfaces and use TRUNK-1 (IP address) as default identification for all the calls.
Yes, I have static IPs and static routes (including each vendor ‘sip’ multiple addresses), and routing, looks ok,however, the asterisk INVITE package, still send the ‘main’ IP address (18.104.22.168 instead of 22.214.171.124) as a the connecting user.
Basically, every reference to 1.1.1.* is incorrect on this package.
Note: 126.96.36.199 and 188.8.131.52 are the PBX IP address.
Retransmitting #4 (no NAT) to 184.108.40.206:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 220.127.116.11:5060;branch=z9hG4bK4747f56e Max-Forwards: 70 From: "USERNAME" <sip:[email protected]>;tag=as02e3c75e To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.25.0) Date: Tue, 21 Aug 2018 16:15:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 227 v=0 o=root 1940137437 1940137437 IN IP4 18.104.22.168 s=Asterisk PBX 11.25.0 c=IN IP4 22.214.171.124 t=0 0 m=audio 17782 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv ---