Query regarding call disconnection after 32s

Hi,

Can someone help me in finding the reason for call disconnection after exact 32 s?

-

== Using SIP RTP CoS mark 5
Audio is at 13904
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.5:5060:
INVITE sip:011917001626618@10.0.1.5 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK01bb05a3
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as724380ef
To: sip:011917001626618@10.0.1.5
Contact: sip:3038727681@10.0.3.3:5060
Call-ID: 6589d7104bf7c3d26f7900ae2bad582d@10.0.3.3:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.6.0
Date: Wed, 29 May 2019 15:27:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-LocalTouch: 0
X-UniqueId: 1559143620.430
X-LeadTransitID:
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 1077121793 1077121793 IN IP4 10.0.3.3
s=Asterisk PBX 13.6.0
c=IN IP4 10.0.3.3
t=0 0
m=audio 13904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


-- Called sip/Xo_main/011917001626618

<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 100 trying – your call is important to us
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK01bb05a3;rport=5060
From: sip:3038727681@10.0.3.3;tag=as724380ef
To: sip:011917001626618@10.0.1.5
Call-ID: 6589d7104bf7c3d26f7900ae2bad582d@10.0.3.3:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
– Called 000752874-011917001626618@engine-outbound-1/n

== Using SIP RTP CoS mark 5
Audio is at 11490
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.5:5060:
INVITE sip:011917001626618@10.0.1.5 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK05b8cea3
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5
Contact: sip:3038727681@10.0.3.3:5060
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.6.0
Date: Wed, 29 May 2019 15:27:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-LocalTouch: 0
X-UniqueId: 1559143620.433
X-LeadTransitID:
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 1579542340 1579542340 IN IP4 10.0.3.3
s=Asterisk PBX 13.6.0
c=IN IP4 10.0.3.3
t=0 0
m=audio 11490 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


-- Called sip/Xo_main/011917001626618

<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 100 trying – your call is important to us
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK05b8cea3;rport=5060
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 183 Session progress
CSeq: 102 INVITE
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Via: SIP/2.0/UDP 10.0.3.3:5060;rport=5060;branch=z9hG4bK05b8cea3
Record-Route: sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Record-Route: sip:10.0.1.5:5060;r2=on;lr=on
Record-Route: sip:35.185.249.169:5060;r2=on;lr=on
Server: Twilio
Contact: sip:172.18.11.110:5060
Content-Type: application/sdp
X-Twilio-CallSid: CAb5405c30b42da5121f79cfb017880aba
Content-Length: 236

v=0
o=root 18329777 18329777 IN IP4 34.203.250.169
s=Twilio Media Gateway
c=IN IP4 34.203.250.169
t=0 0
m=audio 13326 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (14 headers 11 lines) —
sip_route_dump: route/path hop: sip:35.185.249.169:5060;r2=on;lr=on
sip_route_dump: route/path hop: sip:10.0.1.5:5060;r2=on;lr=on
sip_route_dump: route/path hop: sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 34.203.250.169:13326
– SIP/Xo_main-0000004d is making progress passing it to Local/000752874-011917001626618@engine-outbound-1-0000006a;2
– Local/000752874-011917001626618@engine-outbound-1-0000006a;1 is making progress

<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 183 Session progress
CSeq: 102 INVITE
Call-ID: 6589d7104bf7c3d26f7900ae2bad582d@10.0.3.3:5060
From: sip:3038727681@10.0.3.3;tag=as724380ef
To: sip:011917001626618@10.0.1.5;tag=96647085_6772d868_0fc34a01-633c-4544-bdd7-a5e2ad08539a
Via: SIP/2.0/UDP 10.0.3.3:5060;rport=5060;branch=z9hG4bK01bb05a3
Record-Route: sip:54.172.60.3:5060;lr;ftag=as724380ef;twnat=sip:35.185.249.169:5060
Record-Route: sip:10.0.1.5:5060;r2=on;lr=on
Record-Route: sip:35.185.249.169:5060;r2=on;lr=on
Server: Twilio
Contact: sip:172.18.6.143:5060
Content-Type: application/sdp
X-Twilio-CallSid: CA4c789e77c2e3178d4da50b19fcb84f73
Content-Length: 238

v=0
o=root 1085716686 1085716686 IN IP4 34.203.250.60
s=Twilio Media Gateway
c=IN IP4 34.203.250.60
t=0 0
m=audio 11488 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (14 headers 11 lines) —
sip_route_dump: route/path hop: sip:35.185.249.169:5060;r2=on;lr=on
sip_route_dump: route/path hop: sip:10.0.1.5:5060;r2=on;lr=on
sip_route_dump: route/path hop: sip:54.172.60.3:5060;lr;ftag=as724380ef;twnat=sip:35.185.249.169:5060
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 34.203.250.60:11488
– SIP/Xo_main-0000004c is making progress passing it to Local/000752874-011917001626618@engine-outbound-1-00000069;2
– Local/000752874-011917001626618@engine-outbound-1-00000069;1 is making progress

<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 200 OK
CSeq: 102 INVITE
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Via: SIP/2.0/UDP 10.0.3.3:5060;rport=5060;branch=z9hG4bK05b8cea3
Record-Route: sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Record-Route: sip:10.0.1.5:5060;r2=on;lr=on
Record-Route: sip:35.185.249.169:5060;r2=on;lr=on
Server: Twilio
Contact: sip:172.18.11.110:5060
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY
Content-Type: application/sdp
X-Twilio-CallSid: CAb5405c30b42da5121f79cfb017880aba
Content-Length: 236

v=0
o=root 18329777 18329777 IN IP4 34.203.250.169
s=Twilio Media Gateway
c=IN IP4 34.203.250.169
t=0 0
m=audio 13326 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (15 headers 11 lines) —
sip_route_dump: route/path hop: sip:35.185.249.169:5060;r2=on;lr=on
sip_route_dump: route/path hop: sip:10.0.1.5:5060;r2=on;lr=on
sip_route_dump: route/path hop: sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
set_destination: Parsing sip:35.185.249.169:5060;r2=on;lr=on for address/port to send to
set_destination: set destination to 35.185.249.169:5060
Transmitting (no NAT) to 35.185.249.169:5060:
ACK sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK4aafcdcf
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Contact: sip:3038727681@10.0.3.3:5060
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


-- SIP/Xo_main-0000004d answered Local/000752874-011917001626618@engine-outbound-1-0000006a;2
-- Local/000752874-011917001626618@engine-outbound-1-0000006a;1 answered
-- Executing [752874@login-agents:1] Wait("Local/000752874-011917001626618@engine-outbound-1-0000006a;1", "0.1") in new stack
-- Channel SIP/Xo_main-0000004d joined 'simple_bridge' basic-bridge <b2c56575-ee76-4189-a544-4536dea1de27>
-- Channel Local/000752874-011917001626618@engine-outbound-1-0000006a;2 joined 'simple_bridge' basic-bridge <b2c56575-ee76-4189-a544-4536dea1de27>
-- Executing [752874@login-agents:2] Set("Local/000752874-011917001626618@engine-outbound-1-0000006a;1", "CASAGENT=752874") in new stack
-- Executing [752874@login-agents:3] WaitExten("Local/000752874-011917001626618@engine-outbound-1-0000006a;1", "5") in new stack
   > 0x7fab20244c20 -- Probation passed - setting RTP source address to 34.203.250.169:13326

<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 200 OK
CSeq: 102 INVITE
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Via: SIP/2.0/UDP 10.0.3.3:5060;rport=5060;branch=z9hG4bK05b8cea3
Record-Route: sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Record-Route: sip:10.0.1.5:5060;r2=on;lr=on
Record-Route: sip:35.185.249.169:5060;r2=on;lr=on
Server: Twilio
Contact: sip:172.18.11.110:5060
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY
Content-Type: application/sdp
X-Twilio-CallSid: CAb5405c30b42da5121f79cfb017880aba
Content-Length: 236

v=0
o=root 18329777 18329777 IN IP4 34.203.250.169
s=Twilio Media Gateway
c=IN IP4 34.203.250.169
t=0 0
m=audio 13326 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (15 headers 11 lines) —
set_destination: Parsing sip:35.185.249.169:5060;r2=on;lr=on for address/port to send to
set_destination: set destination to 35.185.249.169:5060
Transmitting (no NAT) to 35.185.249.169:5060:
ACK sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK2b5198ad
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Contact: sip:3038727681@10.0.3.3:5060
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 200 OK
CSeq: 102 INVITE
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Via: SIP/2.0/UDP 10.0.3.3:5060;rport=5060;branch=z9hG4bK05b8cea3
Record-Route: sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Record-Route: sip:10.0.1.5:5060;r2=on;lr=on
Record-Route: sip:35.185.249.169:5060;r2=on;lr=on
Server: Twilio
Contact: sip:172.18.11.110:5060
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY
Content-Type: application/sdp
X-Twilio-CallSid: CAb5405c30b42da5121f79cfb017880aba
Content-Length: 236

v=0
o=root 18329777 18329777 IN IP4 34.203.250.169
s=Twilio Media Gateway
c=IN IP4 34.203.250.169
t=0 0
m=audio 13326 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (15 headers 11 lines) —
set_destination: Parsing sip:35.185.249.169:5060;r2=on;lr=on for address/port to send to
set_destination: set destination to 35.185.249.169:5060
Transmitting (no NAT) to 35.185.249.169:5060:
ACK sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK58cc10c8
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Contact: sip:3038727681@10.0.3.3:5060
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 200 OK
CSeq: 102 INVITE
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Via: SIP/2.0/UDP 10.0.3.3:5060;rport=5060;branch=z9hG4bK05b8cea3
Record-Route: sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Record-Route: sip:10.0.1.5:5060;r2=on;lr=on
Record-Route: sip:35.185.249.169:5060;r2=on;lr=on
Server: Twilio
Contact: sip:172.18.11.110:5060
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY
Content-Type: application/sdp
X-Twilio-CallSid: CAb5405c30b42da5121f79cfb017880aba
Content-Length: 236

v=0
o=root 18329777 18329777 IN IP4 34.203.250.169
s=Twilio Media Gateway
c=IN IP4 34.203.250.169
t=0 0
m=audio 13326 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (15 headers 11 lines) —
set_destination: Parsing sip:35.185.249.169:5060;r2=on;lr=on for address/port to send to
set_destination: set destination to 35.185.249.169:5060
Transmitting (no NAT) to 35.185.249.169:5060:
ACK sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK0bd8e385
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Contact: sip:3038727681@10.0.3.3:5060
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


-- Timeout on Local/000752874-011917001626618@engine-outbound-1-0000006a;1, continuing...
-- Executing [752874@login-agents:4] Verbose("Local/000752874-011917001626618@engine-outbound-1-0000006a;1", "2630902") in new stack

2630902
– Executing [752874@login-agents:5] Authenticate(“Local/000752874-011917001626618@engine-outbound-1-0000006a;1”, “2630902”) in new stack
– <Local/000752874-011917001626618@engine-outbound-1-0000006a;1> Playing ‘agent-pass.slin’ (language ‘en’)
– <Local/000752874-011917001626618@engine-outbound-1-0000006a;1> Playing ‘auth-thankyou.slin’ (language ‘en’)
– Executing [752874@login-agents:6] AgentLogin(“Local/000752874-011917001626618@engine-outbound-1-0000006a;1”, “752874”) in new stack
– <Local/000752874-011917001626618@engine-outbound-1-0000006a;1> Playing ‘agent-loginok.slin’ (language ‘en’)
== Agent ‘752874’ logged in (format slin/slin)
– Channel Local/000752874-011917001626618@engine-outbound-1-0000006a;1 joined ‘holding_bridge’ agent_hold-bridge <4caf1320-b78c-4e4e-a7c3-1f149fb4c2d6>

<— SIP read from UDP:10.0.1.5:5060 —>
SIP/2.0 183 Session progress
CSeq: 102 INVITE
Call-ID: 6589d7104bf7c3d26f7900ae2bad582d@10.0.3.3:5060
From: sip:3038727681@10.0.3.3;tag=as724380ef
To: sip:011917001626618@10.0.1.5;tag=96647085_6772d868_0fc34a01-633c-4544-bdd7-a5e2ad08539a
Via: SIP/2.0/UDP 10.0.3.3:5060;rport=5060;branch=z9hG4bK01bb05a3
Record-Route: sip:54.172.60.3:5060;lr;ftag=as724380ef;twnat=sip:35.185.249.169:5060
Record-Route: sip:10.0.1.5:5060;r2=on;lr=on
Record-Route: sip:35.185.249.169:5060;r2=on;lr=on
Server: Twilio
Contact: sip:172.18.6.143:5060
Content-Type: application/sdp
X-Twilio-CallSid: CA4c789e77c2e3178d4da50b19fcb84f73
Content-Length: 238

v=0
o=root 1085716686 1085716686 IN IP4 34.203.250.60
s=Twilio Media Gateway
c=IN IP4 34.203.250.60
t=0 0
m=audio 11488 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (14 headers 11 lines) —
sip_route_dump: route/path hop: sip:35.185.249.169:5060;r2=on;lr=on
sip_route_dump: route/path hop: sip:10.0.1.5:5060;r2=on;lr=on
sip_route_dump: route/path hop: sip:54.172.60.3:5060;lr;ftag=as724380ef;twnat=sip:35.185.249.169:5060
– SIP/Xo_main-0000004c is making progress passing it to Local/000752874-011917001626618@engine-outbound-1-00000069;2
– Local/000752874-011917001626618@engine-outbound-1-00000069;1 is making progress
– Called 752874@agent-request/n
– Executing [752874@agent-request:1] GotoIf(“Local/752874@agent-request-0000006b;2”, “0?spyhangup:request”) in new stack
– Goto (agent-request,752874,5)
– Executing [752874@agent-request:5] AgentRequest(“Local/752874@agent-request-0000006b;2”, “752874”) in new stack
– Channel Local/752874@agent-request-0000006b;2 joined ‘simple_bridge’ basic-bridge <307bb4b3-7a73-4199-bfc8-59b4f0164652>
– Channel Local/000752874-011917001626618@engine-outbound-1-0000006a;1 left ‘holding_bridge’ agent_hold-bridge <4caf1320-b78c-4e4e-a7c3-1f149fb4c2d6>
– Channel Local/000752874-011917001626618@engine-outbound-1-0000006a;1 joined ‘simple_bridge’ basic-bridge <307bb4b3-7a73-4199-bfc8-59b4f0164652>
– Local/752874@agent-request-0000006b;1 is ringing
– Local/752874@agent-request-0000006b;1 answered
– Executing [s@play-file:1] Wait(“Local/752874@agent-request-0000006b;1”, “.001”) in new stack
– Executing [s@play-file:2] Playback(“Local/752874@agent-request-0000006b;1”, “en/beep”) in new stack
– <Local/752874@agent-request-0000006b;1> Playing ‘en/beep.slin’ (language ‘en’)
– Executing [s@play-file:3] Hangup(“Local/752874@agent-request-0000006b;1”, “”) in new stack
== Spawn extension (play-file, s, 3) exited non-zero on ‘Local/752874@agent-request-0000006b;1’
– Channel Local/752874@agent-request-0000006b;2 left ‘simple_bridge’ basic-bridge <307bb4b3-7a73-4199-bfc8-59b4f0164652>
– Channel Local/000752874-011917001626618@engine-outbound-1-0000006a;1 left ‘simple_bridge’ basic-bridge <307bb4b3-7a73-4199-bfc8-59b4f0164652>
– Channel Local/000752874-011917001626618@engine-outbound-1-0000006a;1 joined ‘holding_bridge’ agent_hold-bridge <4caf1320-b78c-4e4e-a7c3-1f149fb4c2d6>
== Spawn extension (agent-request, 752874, 5) exited non-zero on ‘Local/752874@agent-request-0000006b;2’


Scheduling destruction of SIP dialog ‘6589d7104bf7c3d26f7900ae2bad582d@10.0.3.3:5060’ in 32000 ms (Method: INVITE)
Really destroying SIP dialog ‘6589d7104bf7c3d26f7900ae2bad582d@10.0.3.3:5060’ Method: INVITE
[2019-05-29 08:28:40.263] NOTICE[2422]: chan_sip.c:28805 check_rtp_timeout: Disconnecting call ‘SIP/Xo_main-0000004d’ for lack of RTP activity in 61 seconds
– Channel SIP/Xo_main-0000004d left ‘simple_bridge’ basic-bridge
Scheduling destruction of SIP dialog ‘7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:35.185.249.169:5060;r2=on;lr=on for address/port to send to
set_destination: set destination to 35.185.249.169:5060
Reliably Transmitting (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


-- Channel Local/000752874-011917001626618@engine-outbound-1-0000006a;2 left 'simple_bridge' basic-bridge <b2c56575-ee76-4189-a544-4536dea1de27>

== Spawn extension (engine-outbound-1, 000752874-011917001626618, 13) exited non-zero on ‘Local/000752874-011917001626618@engine-outbound-1-0000006a;2’
– Channel Local/000752874-011917001626618@engine-outbound-1-0000006a;1 left ‘holding_bridge’ agent_hold-bridge <4caf1320-b78c-4e4e-a7c3-1f149fb4c2d6>
== Agent ‘752874’ logged out. Logged in for 81 seconds.
== Spawn extension (login-agents, 752874, 6) exited non-zero on ‘Local/000752874-011917001626618@engine-outbound-1-0000006a;1’
– Executing [h@login-agents:1] GotoIf(“Local/000752874-011917001626618@engine-outbound-1-0000006a;1”, “0 & 1?casdevs”) in new stack
– Executing [h@login-agents:2] Goto(“Local/000752874-011917001626618@engine-outbound-1-0000006a;1”, “agent-logoff,752874,1”) in new stack
– Goto (agent-logoff,752874,1)
– Executing [752874@agent-logoff:1] Verbose(“Local/000752874-011917001626618@engine-outbound-1-0000006a;1”, “Logging off Agnet No. 752874”) in new stack
Logging off Agnet No. 752874
– Executing [752874@agent-logoff:2] System(“Local/000752874-011917001626618@engine-outbound-1-0000006a;1”, “/usr/sbin/asterisk -rx “ConfBridge kick 752874 all””) in new stack
– Remote UNIX connection
– Remote UNIX connection disconnected
– Executing [752874@agent-logoff:3] System(“Local/000752874-011917001626618@engine-outbound-1-0000006a;1”, "/usr/sbin/asterisk -rx “agent logoff 752874"”) in new stack
– Remote UNIX connection
– Remote UNIX connection disconnected
– Executing [752874@agent-logoff:4] Hangup(“Local/000752874-011917001626618@engine-outbound-1-0000006a;1”, “”) in new stack
== Spawn extension (agent-logoff, 752874, 4) exited non-zero on ‘Local/000752874-011917001626618@engine-outbound-1-0000006a;1’
Retransmitting #1 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #2 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #3 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #4 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #5 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #6 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #7 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #8 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #9 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Retransmitting #10 (no NAT) to 35.185.249.169:5060:
BYE sip:172.18.11.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.3:5060;branch=z9hG4bK793d0fac
Route: sip:35.185.249.169:5060;r2=on;lr=on,sip:10.0.1.5:5060;r2=on;lr=on,sip:54.172.60.3:5060;lr;ftag=as5f19a2de;twnat=sip:35.185.249.169:5060
Max-Forwards: 70
From: sip:3038727681@10.0.3.3;tag=as5f19a2de
To: sip:011917001626618@10.0.1.5;tag=56427181_6772d868_a41e4818-b384-4afa-b186-647d8263fbe8
Call-ID: 7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Really destroying SIP dialog ‘7a4438171362f8b517ccd3d238b4994f@10.0.3.3:5060’ Method: INVITE

What is the network topology and usage? Are you using a SIP proxy or device in beween? I ask because Asterisk is sending an ACK directly but it is not being received, so something may be misconfigured.

Hi,

We are using SIP proxy in between, and so, we have created a user of proxy on asterisk, and are directly sending requests to it, which is then being routed by it.

For ACK, when I am using nat setting on sip.conf, call is not getting disconnected, as ACK is properly received by proxy, but another problem arises as, when we disconnection the call received from our own device(phone), the call is not hanging up on asterisk, and channel is still there

Hi,

How to configure asterisk in a way, so that it sends request to private ip of proxy, instead of public ip?

This is the reason of ack being sent to public ip,a nd proxy not receiving it.

How have you configured chan_sip in the first place?