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:[email protected]:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.229:5070;rport;branch=z9hG4bK5m6QQQ1Dcr2US.
Max-Forwards: 26.
From: "8185555555" <sip:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]:5060>.
Call-ID: 72af2146-c3cf-11e6-b5fc-0b4e5249c7de.
CSeq: 100624257 INVITE.
Contact: <sip:[email protected]: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:[email protected]>.
.
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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]:5060>.
Content-Length: 0.
.


U 1.1.1.8:5060 -> 1.1.1.147:49680
INVITE sip:[email protected]:49680;ob SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.8:5060;branch=z9hG4bK6e32185a;rport.
Max-Forwards: 70.
From: "8185555555" <sip:[email protected]>;tag=as457f20ca.
To: <sip:[email protected]:49680;ob>.
Contact: <sip:[email protected]:5060>.
Call-ID: [email protected]: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:[email protected]>.
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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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: [email protected]:5060.
From: "8185555555" <sip:[email protected]>;tag=as457f20ca.
To: <sip:[email protected];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: [email protected]:5060.
From: "8185555555" <sip:[email protected]>;tag=as457f20ca.
To: <sip:[email protected];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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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: [email protected]:5060.
From: "8185555555" <sip:[email protected]>;tag=as457f20ca.
To: <sip:[email protected];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:[email protected]>;tag=as457f20ca.
To: <sip:[email protected]:49680;ob>;tag=4088b0e3c39c40fb8d183b79a6b24577.
Contact: <sip:[email protected]:5060>.
Call-ID: [email protected]: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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=0pcy77Bt0cQyK.
To: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=as457f20ca.
To: <sip:[email protected]:49680;ob>;tag=4088b0e3c39c40fb8d183b79a6b24577.
Call-ID: [email protected]: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: [email protected]:5060.
From: "8185555555" <sip:[email protected]>;tag=as457f20ca.
To: <sip:[email protected];ob>;tag=4088b0e3c39c40fb8d183b79a6b24577.
CSeq: 103 BYE.
Content-Length:  0.
.


U 1.1.1.8:5060 -> 1.1.1.229:5070
BYE sip:[email protected]:5070 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.11:5060;branch=z9hG4bK3f88c697;rport.
Max-Forwards: 70.
From: <sip:[email protected]:5060>;tag=as754aeaa1.
To: "8185555555" <sip:[email protected]>;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:[email protected]:5060>;tag=as754aeaa1.
To: "8185555555" <sip:[email protected]>;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!

I do not see any FreePBX issue here.
Your provider is not responding to your 200-INVITE with ACK. Compare with Bria to Asterisk message exchange where all the required messages are present.
Speak with provider.

1 Like

Thank you for the prompt response, I too notice the SIP transactions appear to be normal. Can you provide any insight as to what you think may be happening with the communication to the SIP provider?

Edit: I updated the logs after making several changes and they still appear to be normal. Just odd I can hear full audio from both sides but the call ends after 5s every time…

Your provider is not responding to your 200-INVITE with ACK as he should.
Please refer to rfc3665 Session Initiation Protocol (SIP) Basic Call Flow Examples, section 3.1. Successful Session Establishment.
Speak with provider.

Not getting an ACK is typically the result of incorrect NAT or firewall configuration, such that either the ACK or the OK gets blocked or mis-routed.

I would note that you are sending from 1.1.1.8, but have a contact address of 1.1.1.11. Applying the principles of minimum obfuscation and not obfuscating the distinction between networks, particularly rouable an unrotable ones, all the addresses are public, and there is something seriously wrong with the configuration which is resulting in your providing the wrong address for yourself.

If you haven’t followed those principles, please re-obfuscate obeying them.

The fact that you get the OK to the BYE suggests that the problem is that the peer cannot send to 1.1.1.11, rather than that you cannot send to 1.1.1.229, as responses are generally send to the actual address from which the request was received, whereas BYE is sent to the contact address.

Note this thread has been muli-posted to community.asterisk.org, although most of the useful responses are here. Please do not multi-post. It wastes people’s time. If you must, please cross-link the threads.

@datasplice, please make sure that you’re following pfSense guidelines for VoIP:

https://doc.pfsense.org/index.php/VoIP_Configuration
https://doc.pfsense.org/index.php/PBX_VoIP_NAT_How-to

Do not use siproxd though.