I’ve just rolled out all of our remote extensions that are now connecting and registering with our PBX. We have no audio in either direction.
- Our PBX is hosted in our main office, i have the in built firewall enabled and then it goes through a USG Pro which is forwarding all ports to our PBX.
- On the remote extensions, they have all ports allowed to the external PBX.
- I have NAT enabled at both ends with the external IP of the PBX being used.
- I’ve double checked the correct ports are open.
I’m not sure what next steps to take. Here is a snapshot of a call;
[2020-02-10 10:02:54] VERBOSE chan_sip.c: --- (9 headers 0 lines) --- [2020-02-10 10:02:54] VERBOSE chan_sip.c: Really destroying SIP dialog 'firstname.lastname@example.org:5160' Method: OPTIONS [2020-02-10 10:02:54] VERBOSE chan_sip.c: Retransmitting #2 (NAT) to 85.00.00.19:65476: OPTIONS sip:email@example.com:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 00.00.47.35:5160;branch=z9hG4bK430ca9d9;rport Max-Forwards: 70 From: "Unknown" <sip:Unknown@212.00.00.35:5160>;tag=as10607e18 To: <sip:firstname.lastname@example.org:5060;transport=UDP> Contact: <sip:Unknown@212.00.00.35:5160> Call-ID: email@example.com:5160 CSeq: 102 OPTIONS User-Agent: FPBX-220.127.116.11(16.6.2) Date: Mon, 10 Feb 2020 10:02:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0
Using sngrep (it’s a great tool, wish i knew about it before). i have the following;
[ ] 426 INVITE firstname.lastname@example.org email@example.com 8 REMOTE_EXT_IP:26571 Internal_PBX_IP:5160 IN CALL
I’m guessing i shouldnt have the internal PBX IP in the destination and it should be the pbx external IP?