Inbound calls ending after 5s

Hello everyone, I have been struggling with a problem of inbound calls ending after 5s. During these 5s I have full audio. Any ideas would be greatly appreciated!

FreePBX 12.0.76.4
pfSense 2.3.2
Sip Provider: ArcTele

Call Log (UPDATED: 12/16):

U 1.1.1.229:5070 -> 1.1.1.8:5060
INVITE sip:4472@voip.mywebsite.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.229:5070;rport;branch=z9hG4bK5m6QQQ1Dcr2US.
Max-Forwards: 26.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Contact: <sip:mod_sofia@1.1.1.229:5070>.
User-Agent: ArcTele VSS 2.2.2.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Privacy: none.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 290.
P-Asserted-Identity: "8185555555" <sip:8185555555@inbound.v-aci.com>.
.
v=0.
o=ARCTELE_VSS_2_2_2 1750851968 1750851969 IN IP4 1.1.1.229.
s=ARCTELE_VSS_2_2_2.
c=IN IP4 1.1.1.229.
t=0 0.
m=audio 18614 RTP/AVP 18 0 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=maxptime:20.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Length: 0.
.


U 1.1.1.8:5060 -> 1.1.1.147:49680
INVITE sip:4472@1.1.1.147:49680;ob SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.8:5060;branch=z9hG4bK6e32185a;rport.
Max-Forwards: 70.
From: "8185555555" <sip:8185555555@1.1.1.8>;tag=as457f20ca.
To: <sip:4472@1.1.1.147:49680;ob>.
Contact: <sip:8185555555@1.1.1.8:5060>.
Call-ID: 08b0856a5971ce6b318775c626073e06@1.1.1.8:5060.
CSeq: 102 INVITE.
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Date: Fri, 16 Dec 2016 20:37:23 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
P-Asserted-Identity: "8185555555" <sip:8185555555@1.1.1.8>.
Content-Type: application/sdp.
Content-Length: 283.
.
v=0.
o=root 1962633121 1962633121 IN IP4 1.1.1.8.
s=Asterisk PBX 11.21.2.
c=IN IP4 1.1.1.8.
t=0 0.
m=audio 10686 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Length: 0.
.


U 1.1.1.147:49680 -> 1.1.1.8:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 1.1.1.8:5060;rport=5060;received=1.1.1.8;branch=z9hG4bK6e32185a.
Call-ID: 08b0856a5971ce6b318775c626073e06@1.1.1.8:5060.
From: "8185555555" <sip:8185555555@1.1.1.8>;tag=as457f20ca.
To: <sip:4472@1.1.1.147;ob>.
CSeq: 102 INVITE.
Content-Length:  0.
.


U 1.1.1.147:49680 -> 1.1.1.8:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 1.1.1.8:5060;rport=5060;received=1.1.1.8;branch=z9hG4bK6e32185a.
Call-ID: 08b0856a5971ce6b318775c626073e06@1.1.1.8:5060.
From: "8185555555" <sip:8185555555@1.1.1.8>;tag=as457f20ca.
To: <sip:4472@1.1.1.147;ob>;tag=4088b0e3c39c40fb8d183b79a6b24577.
CSeq: 102 INVITE.
Contact: <sip:1.1.1.147:49680>.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS.
Content-Length:  0.
.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Length: 0.
.


U 1.1.1.147:49680 -> 1.1.1.8:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.8:5060;rport=5060;received=1.1.1.8;branch=z9hG4bK6e32185a.
Call-ID: 08b0856a5971ce6b318775c626073e06@1.1.1.8:5060.
From: "8185555555" <sip:8185555555@1.1.1.8>;tag=as457f20ca.
To: <sip:4472@1.1.1.147;ob>;tag=4088b0e3c39c40fb8d183b79a6b24577.
CSeq: 102 INVITE.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS.
Contact: <sip:1.1.1.147:49680>.
Supported: replaces, 100rel, timer, norefersub.
Content-Type: application/sdp.
Content-Length:   280.
.
v=0.
o=- 3690884243 3690884244 IN IP4 1.1.1.147.
s=pjmedia.
b=AS:84.
t=0 0.
a=X-nat:0.
m=audio 4034 RTP/AVP 0 101.
c=IN IP4 1.1.1.147.
b=TIAS:64000.
a=rtcp:4035 IN IP4 1.1.1.147.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.


U 1.1.1.8:5060 -> 1.1.1.147:49680
ACK sip:1.1.1.147:49680 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.8:5060;branch=z9hG4bK68e87c9c;rport.
Max-Forwards: 70.
From: "8185555555" <sip:8185555555@1.1.1.8>;tag=as457f20ca.
To: <sip:4472@1.1.1.147:49680;ob>;tag=4088b0e3c39c40fb8d183b79a6b24577.
Contact: <sip:8185555555@1.1.1.8:5060>.
Call-ID: 08b0856a5971ce6b318775c626073e06@1.1.1.8:5060.
CSeq: 102 ACK.
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Content-Length: 0.
.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 285.
.
v=0.
o=root 1652470227 1652470227 IN IP4 1.1.1.11.
s=Asterisk PBX 11.21.2.
c=IN IP4 1.1.1.11.
t=0 0.
m=audio 36456 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 285.
.
v=0.
o=root 1652470227 1652470227 IN IP4 1.1.1.11.
s=Asterisk PBX 11.21.2.
c=IN IP4 1.1.1.11.
t=0 0.
m=audio 36456 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 285.
.
v=0.
o=root 1652470227 1652470227 IN IP4 1.1.1.11.
s=Asterisk PBX 11.21.2.
c=IN IP4 1.1.1.11.
t=0 0.
m=audio 36456 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 285.
.
v=0.
o=root 1652470227 1652470227 IN IP4 1.1.1.11.
s=Asterisk PBX 11.21.2.
c=IN IP4 1.1.1.11.
t=0 0.
m=audio 36456 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 285.
.
v=0.
o=root 1652470227 1652470227 IN IP4 1.1.1.11.
s=Asterisk PBX 11.21.2.
c=IN IP4 1.1.1.11.
t=0 0.
m=audio 36456 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 285.
.
v=0.
o=root 1652470227 1652470227 IN IP4 1.1.1.11.
s=Asterisk PBX 11.21.2.
c=IN IP4 1.1.1.11.
t=0 0.
m=audio 36456 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 1.1.1.8:5060 -> 1.1.1.229:5070
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.229:5070;branch=z9hG4bK5m6QQQ1Dcr2US;received=1.1.1.229;rport=5070.
From: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
To: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Server: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:4472@1.1.1.11:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 285.
.
v=0.
o=root 1652470227 1652470227 IN IP4 1.1.1.11.
s=Asterisk PBX 11.21.2.
c=IN IP4 1.1.1.11.
t=0 0.
m=audio 36456 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 1.1.1.8:5060 -> 1.1.1.147:49680
BYE sip:1.1.1.147:49680 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.8:5060;branch=z9hG4bK571c5527;rport.
Max-Forwards: 70.
From: "8185555555" <sip:8185555555@1.1.1.8>;tag=as457f20ca.
To: <sip:4472@1.1.1.147:49680;ob>;tag=4088b0e3c39c40fb8d183b79a6b24577.
Call-ID: 08b0856a5971ce6b318775c626073e06@1.1.1.8:5060.
CSeq: 103 BYE.
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.


U 1.1.1.147:49680 -> 1.1.1.8:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.8:5060;rport=5060;received=1.1.1.8;branch=z9hG4bK571c5527.
Call-ID: 08b0856a5971ce6b318775c626073e06@1.1.1.8:5060.
From: "8185555555" <sip:8185555555@1.1.1.8>;tag=as457f20ca.
To: <sip:4472@1.1.1.147;ob>;tag=4088b0e3c39c40fb8d183b79a6b24577.
CSeq: 103 BYE.
Content-Length:  0.
.


U 1.1.1.8:5060 -> 1.1.1.229:5070
BYE sip:mod_sofia@1.1.1.229:5070 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.11:5060;branch=z9hG4bK3f88c697;rport.
Max-Forwards: 70.
From: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
To: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 102 BYE.
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.21.2).
X-Asterisk-HangupCause: No user responding.
X-Asterisk-HangupCauseCode: 18.
Content-Length: 0.
.


U 1.1.1.229:5070 -> 1.1.1.8:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.1.1.11:5060;branch=z9hG4bK3f88c697;rport=5060;received=50.242.142.12.
From: <sip:4472@voip.mywebsite.com:5060>;tag=as754aeaa1.
To: "8185555555" <sip:8185555555@inbound.v-aci.com>;tag=0pcy77Bt0cQyK.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 102 BYE.
User-Agent: ArcTele VSS 2.2.2.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY.
Supported: timer, path, replaces.
Content-Length: 0.

Many thanks for the help!

Your logs shows this as the hangup cause reason

The correspoding ACK after the 200 OK is not received

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