Hello, I have the following peculiar setup. There is a main FusionPBX instance where all the phones connect to, this works well. Now in one location the SIP Trunk Provider allows connections only from a physical place, a specific port on a router and with a static IP. So what I thought is setting up another FusionPBX instance there and route the calls to my main PBX.
Below is a little diagram of the setup, PBX A is the main PBX where all the VOIP phones are connected to. PBX B is the one that is in that specific location.
The two PBX-es are connected through a Wireguard VPN directly to make things easier. 10.12.0.1 is PBX A, 10.12.18.200 PBX B
On PBX B I have ext 1000 and PBX A connect to it as a gateway. This way incoming calls on a DID from PBX B to PBX A are routed and working. Now I am trying to make outgoing calls from PBX A to PBX B and I do receive the INVITE calls, but I am not able to catch them in any way.
Here is the INVITE that I receive on PBX B from PBX A through extension 1000:
2026/04/04 21:38:29.093047 10.12.0.1:5060 -> 10.12.18.200:5060
INVITE sip:07XXXXXX0@10.12.18.200 SIP/2.0
Via: SIP/2.0/UDP 10.12.0.1;rport;branch=z9hG4bKHp5jtr0Hc2Fjp
Max-Forwards: 69
From: "210" <sip:1000@pbxb>;tag=Q9pgcmpap3Btj
To: <sip:07XXXXXX0@10.12.18.200>
Call-ID: 50a48e8e-aaf8-123f-5e9b-bc241145b13b
CSeq: 113110162 INVITE
Contact: <sip:gw+29407c55-abe6-44ac-8dfa-e5a643a9f5a1@10.12.0.1:5060;transport=udp;gw=29407c55-abe6-44ac-8dfa-e5a643a9f5a1>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 312
accountcode: ltd.domain.net
X-FS-Support: update_display,send_info
Remote-Party-ID: "210" <sip:3XXXXXX40@pbxb>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1775298949 1775298950 IN IP4 10.12.0.1
s=FreeSWITCH
c=IN IP4 10.12.0.1
t=0 0
m=audio 28960 RTP/AVP 0 8 18 9 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:13 CN/8000
a=ptime:20
Below is a little diagram of the setup, PBX A is the main PBX where all the VOIP phones are connected to. PBX B is the one that is in that specific location.
The two PBX-es are connected through a Wireguard VPN directly to make things easier. 10.12.0.1 is PBX A, 10.12.18.200 PBX B
On PBX B I have ext 1000 and PBX A connect to it as a gateway. This way incoming calls on a DID from PBX B to PBX A are routed and working. Now I am trying to make outgoing calls from PBX A to PBX B and I do receive the INVITE calls, but I am not able to catch them in any way.
Here is the INVITE that I receive on PBX B from PBX A through extension 1000:
2026/04/04 21:38:29.093047 10.12.0.1:5060 -> 10.12.18.200:5060
INVITE sip:07XXXXXX0@10.12.18.200 SIP/2.0
Via: SIP/2.0/UDP 10.12.0.1;rport;branch=z9hG4bKHp5jtr0Hc2Fjp
Max-Forwards: 69
From: "210" <sip:1000@pbxb>;tag=Q9pgcmpap3Btj
To: <sip:07XXXXXX0@10.12.18.200>
Call-ID: 50a48e8e-aaf8-123f-5e9b-bc241145b13b
CSeq: 113110162 INVITE
Contact: <sip:gw+29407c55-abe6-44ac-8dfa-e5a643a9f5a1@10.12.0.1:5060;transport=udp;gw=29407c55-abe6-44ac-8dfa-e5a643a9f5a1>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 312
accountcode: ltd.domain.net
X-FS-Support: update_display,send_info
Remote-Party-ID: "210" <sip:3XXXXXX40@pbxb>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1775298949 1775298950 IN IP4 10.12.0.1
s=FreeSWITCH
c=IN IP4 10.12.0.1
t=0 0
m=audio 28960 RTP/AVP 0 8 18 9 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:13 CN/8000
a=ptime:20
