Call Hangup Issue

chandimadharmarathne · November 14, 2025, 4:43am

My dial plan has Hangup. but call actually not hangup. I provide the logs. How fix this issue?



<--- Transmitting SIP response (550 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bKmm2p6g7wvvg8v7mvx6pr6g0p7;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Record-Route: <sip:172.16.1.1:5060;lr;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=12297>
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>
CSeq: 1 INVITE
Server: Asterisk PBX 20.14.0
Content-Length:  0


    -- Executing [0112019515@testing:1] NoOp("PJSIP/gw-oneaccess-00000002", "Incoming call from 0761604153") in new stack
    -- Executing [0112019515@testing:1] NoOp("PJSIP/gw-oneaccess-00000002", "Incoming call from 0761604153") in new stack
    -- Executing [0112019515@testing:2] Dial("PJSIP/gw-oneaccess-00000002", "PJSIP/forward136,30,g") in new stack
    -- Executing [0112019515@testing:2] Dial("PJSIP/gw-oneaccess-00000002", "PJSIP/forward136,30,g") in new stack
    -- Called PJSIP/forward136
    -- Called PJSIP/forward136
<--- Transmitting SIP request (912 bytes) to UDP:172.20.10.136:5060 --->
INVITE sip:172.20.10.136:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.100:5060;rport;branch=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>
Contact: <sip:asterisk@192.168.50.100:5060>
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 32159 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Content-Type: application/sdp
Content-Length:   241

v=0
o=- 1720928544 1720928544 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 10492 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:140
a=sendrecv

<--- Transmitting SIP request (912 bytes) to UDP:172.20.10.136:5060 --->
INVITE sip:172.20.10.136:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.100:5060;rport;branch=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>
Contact: <sip:asterisk@192.168.50.100:5060>
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 32159 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Content-Type: application/sdp
Content-Length:   241

v=0
o=- 1720928544 1720928544 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 10492 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:140
a=sendrecv

<--- Received SIP response (572 bytes) from UDP:172.20.10.136:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.50.100:5060;rport=5060;received=192.168.50.100;branch=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>;tag=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
CSeq: 32159 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1763094860/98168534005b567fb04c97be4482cf05",opaque="720e1d3564aa9266",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.16.0
Content-Length:  0


<--- Received SIP response (572 bytes) from UDP:172.20.10.136:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.50.100:5060;rport=5060;received=192.168.50.100;branch=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>;tag=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
CSeq: 32159 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1763094860/98168534005b567fb04c97be4482cf05",opaque="720e1d3564aa9266",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.16.0
Content-Length:  0


<--- Transmitting SIP request (426 bytes) to UDP:172.20.10.136:5060 --->
ACK sip:172.20.10.136:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.100:5060;rport;branch=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>;tag=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 32159 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Content-Length:  0


<--- Transmitting SIP request (426 bytes) to UDP:172.20.10.136:5060 --->
ACK sip:172.20.10.136:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.100:5060;rport;branch=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>;tag=z9hG4bKPjc0aa7b23-03cf-4506-8467-5686bf192afa
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 32159 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Content-Length:  0


<--- Transmitting SIP request (1208 bytes) to UDP:172.20.10.136:5060 --->
INVITE sip:172.20.10.136:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.100:5060;rport;branch=z9hG4bKPj5fcb6cc3-762b-43c7-a016-0f4d1a1d5456
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>
Contact: <sip:asterisk@192.168.50.100:5060>
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 32160 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Authorization: Digest username="forward", realm="asterisk", nonce="1763094860/98168534005b567fb04c97be4482cf05", uri="sip:172.20.10.136:5060", response="a438827c892eea4fd06c527f4f29239c", algorithm=MD5, cnonce="79347bdaa81f41f693fb8d241ca6bf36", opaque="720e1d3564aa9266", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   241

v=0
o=- 1720928544 1720928544 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 10492 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:140
a=sendrecv

<--- Transmitting SIP request (1208 bytes) to UDP:172.20.10.136:5060 --->
INVITE sip:172.20.10.136:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.100:5060;rport;branch=z9hG4bKPj5fcb6cc3-762b-43c7-a016-0f4d1a1d5456
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>
Contact: <sip:asterisk@192.168.50.100:5060>
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 32160 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Authorization: Digest username="forward", realm="asterisk", nonce="1763094860/98168534005b567fb04c97be4482cf05", uri="sip:172.20.10.136:5060", response="a438827c892eea4fd06c527f4f29239c", algorithm=MD5, cnonce="79347bdaa81f41f693fb8d241ca6bf36", opaque="720e1d3564aa9266", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   241

v=0
o=- 1720928544 1720928544 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 10492 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:140
a=sendrecv

<--- Received SIP response (370 bytes) from UDP:172.20.10.136:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.50.100:5060;rport=5060;received=192.168.50.100;branch=z9hG4bKPj5fcb6cc3-762b-43c7-a016-0f4d1a1d5456
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>
CSeq: 32160 INVITE
Server: Asterisk PBX 20.16.0
Content-Length:  0


<--- Received SIP response (370 bytes) from UDP:172.20.10.136:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.50.100:5060;rport=5060;received=192.168.50.100;branch=z9hG4bKPj5fcb6cc3-762b-43c7-a016-0f4d1a1d5456
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>
CSeq: 32160 INVITE
Server: Asterisk PBX 20.16.0
Content-Length:  0


<--- Received SIP response (933 bytes) from UDP:172.20.10.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.100:5060;rport=5060;received=192.168.50.100;branch=z9hG4bKPj5fcb6cc3-762b-43c7-a016-0f4d1a1d5456
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>;tag=9292c238-1d80-4936-b2eb-7bb7dc03075a
CSeq: 32160 INVITE
Server: Asterisk PBX 20.16.0
Contact: <sip:172.20.10.136:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 1720928544 1720928546 IN IP4 172.20.10.136
s=Asterisk
c=IN IP4 172.20.10.136
t=0 0
m=audio 18208 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:140
a=sendrecv

<--- Received SIP response (933 bytes) from UDP:172.20.10.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.100:5060;rport=5060;received=192.168.50.100;branch=z9hG4bKPj5fcb6cc3-762b-43c7-a016-0f4d1a1d5456
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>;tag=9292c238-1d80-4936-b2eb-7bb7dc03075a
CSeq: 32160 INVITE
Server: Asterisk PBX 20.16.0
Contact: <sip:172.20.10.136:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 1720928544 1720928546 IN IP4 172.20.10.136
s=Asterisk
c=IN IP4 172.20.10.136
t=0 0
m=audio 18208 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:140
a=sendrecv

<--- Transmitting SIP request (417 bytes) to UDP:172.20.10.136:5060 --->
ACK sip:172.20.10.136:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.100:5060;rport;branch=z9hG4bKPj1ed1b033-bfdc-448c-9e5e-b8e05a862331
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>;tag=9292c238-1d80-4936-b2eb-7bb7dc03075a
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 32160 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Content-Length:  0


<--- Transmitting SIP request (417 bytes) to UDP:172.20.10.136:5060 --->
ACK sip:172.20.10.136:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.100:5060;rport;branch=z9hG4bKPj1ed1b033-bfdc-448c-9e5e-b8e05a862331
From: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
To: <sip:172.20.10.136>;tag=9292c238-1d80-4936-b2eb-7bb7dc03075a
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 32160 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Content-Length:  0


    -- PJSIP/forward136-00000003 answered PJSIP/gw-oneaccess-00000002
    -- PJSIP/forward136-00000003 answered PJSIP/gw-oneaccess-00000002
<--- Transmitting SIP response (1238 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bKmm2p6g7wvvg8v7mvx6pr6g0p7;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Record-Route: <sip:172.16.1.1:5060;lr;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=12297>
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 1 INVITE
Server: Asterisk PBX 20.14.0
Contact: <sip:192.168.50.100:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, replaces, norefersub
P-Asserted-Identity: <sip:0112019515@192.168.50.100;user=phone>
Remote-Party-ID: <sip:0112019515@192.168.50.100;user=phone>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   262

v=0
o=- 1076676320 1076676323 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 19272 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (1238 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bKmm2p6g7wvvg8v7mvx6pr6g0p7;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Record-Route: <sip:172.16.1.1:5060;lr;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=12297>
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 1 INVITE
Server: Asterisk PBX 20.14.0
Contact: <sip:192.168.50.100:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, replaces, norefersub
P-Asserted-Identity: <sip:0112019515@192.168.50.100;user=phone>
Remote-Party-ID: <sip:0112019515@192.168.50.100;user=phone>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   262

v=0
o=- 1076676320 1076676323 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 19272 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

    -- Channel PJSIP/forward136-00000003 joined 'simple_bridge' basic-bridge <518b178f-4bb8-4f1a-9f8f-746fba92cf85>
    -- Channel PJSIP/forward136-00000003 joined 'simple_bridge' basic-bridge <518b178f-4bb8-4f1a-9f8f-746fba92cf85>
    -- Channel PJSIP/gw-oneaccess-00000002 joined 'simple_bridge' basic-bridge <518b178f-4bb8-4f1a-9f8f-746fba92cf85>
    -- Channel PJSIP/gw-oneaccess-00000002 joined 'simple_bridge' basic-bridge <518b178f-4bb8-4f1a-9f8f-746fba92cf85>
       > Bridge 518b178f-4bb8-4f1a-9f8f-746fba92cf85: switching from simple_bridge technology to native_rtp
       > Bridge 518b178f-4bb8-4f1a-9f8f-746fba92cf85: switching from simple_bridge technology to native_rtp
       > Locally RTP bridged 'PJSIP/gw-oneaccess-00000002' and 'PJSIP/forward136-00000003' in stack
       > Locally RTP bridged 'PJSIP/gw-oneaccess-00000002' and 'PJSIP/forward136-00000003' in stack
<--- Transmitting SIP response (1238 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bKmm2p6g7wvvg8v7mvx6pr6g0p7;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Record-Route: <sip:172.16.1.1:5060;lr;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=12297>
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 1 INVITE
Server: Asterisk PBX 20.14.0
Contact: <sip:192.168.50.100:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, replaces, norefersub
P-Asserted-Identity: <sip:0112019515@192.168.50.100;user=phone>
Remote-Party-ID: <sip:0112019515@192.168.50.100;user=phone>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   262

v=0
o=- 1076676320 1076676323 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 19272 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (1238 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bKmm2p6g7wvvg8v7mvx6pr6g0p7;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Record-Route: <sip:172.16.1.1:5060;lr;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=12297>
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 1 INVITE
Server: Asterisk PBX 20.14.0
Contact: <sip:192.168.50.100:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, replaces, norefersub
P-Asserted-Identity: <sip:0112019515@192.168.50.100;user=phone>
Remote-Party-ID: <sip:0112019515@192.168.50.100;user=phone>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   262

v=0
o=- 1076676320 1076676323 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 19272 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP request (483 bytes) from UDP:172.16.1.1:5060 --->
ACK sip:192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK0r8vx0p8zq7gmqgg2yq8r7gsy;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153"<sip:0761604153@192.168.50.100;transport=udp;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515"<sip:0112019515@192.168.50.100;transport=udp;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 1 ACK
Max-Forwards: 23
Content-Length: 0


<--- Received SIP request (483 bytes) from UDP:172.16.1.1:5060 --->
ACK sip:192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK0r8vx0p8zq7gmqgg2yq8r7gsy;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153"<sip:0761604153@192.168.50.100;transport=udp;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515"<sip:0112019515@192.168.50.100;transport=udp;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 1 ACK
Max-Forwards: 23
Content-Length: 0


<--- Received SIP request (873 bytes) from UDP:172.16.1.1:5060 --->
INVITE sip:192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK2ssvxmvpy0g0msg0smrq6zq28;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153"<sip:0761604153@192.168.50.100;transport=udp;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515"<sip:0112019515@192.168.50.100;transport=udp;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 2 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE
Contact: <sip:172.16.1.1:5060;transport=udp;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff>
Max-Forwards: 23
Content-Length: 186
Content-Type: application/sdp

v=0
o=- 1076676320 1076676322 IN IP4 172.16.1.1
s=SBC call
c=IN IP4 172.16.1.1
t=0 0
m=audio 12008 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=ptime:20

<--- Received SIP request (873 bytes) from UDP:172.16.1.1:5060 --->
INVITE sip:192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK2ssvxmvpy0g0msg0smrq6zq28;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153"<sip:0761604153@192.168.50.100;transport=udp;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515"<sip:0112019515@192.168.50.100;transport=udp;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 2 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE
Contact: <sip:172.16.1.1:5060;transport=udp;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff>
Max-Forwards: 23
Content-Length: 186
Content-Type: application/sdp

v=0
o=- 1076676320 1076676322 IN IP4 172.16.1.1
s=SBC call
c=IN IP4 172.16.1.1
t=0 0
m=audio 12008 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=ptime:20

<--- Transmitting SIP response (1117 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bK2ssvxmvpy0g0msg0smrq6zq28;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 2 INVITE
Contact: <sip:192.168.50.100:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: <sip:0112019515@192.168.50.100;user=phone>
Remote-Party-ID: <sip:0112019515@192.168.50.100;user=phone>;party=called;privacy=off;screen=no
Server: Asterisk PBX 20.14.0
Content-Type: application/sdp
Content-Length:   238

v=0
o=- 1076676320 1076676324 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 19272 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (1117 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bK2ssvxmvpy0g0msg0smrq6zq28;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 2 INVITE
Contact: <sip:192.168.50.100:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: <sip:0112019515@192.168.50.100;user=phone>
Remote-Party-ID: <sip:0112019515@192.168.50.100;user=phone>;party=called;privacy=off;screen=no
Server: Asterisk PBX 20.14.0
Content-Type: application/sdp
Content-Length:   238

v=0
o=- 1076676320 1076676324 IN IP4 192.168.50.100
s=Asterisk
c=IN IP4 192.168.50.100
t=0 0
m=audio 19272 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

       > Locally RTP bridged 'PJSIP/gw-oneaccess-00000002' and 'PJSIP/forward136-00000003' in stack
       > Locally RTP bridged 'PJSIP/gw-oneaccess-00000002' and 'PJSIP/forward136-00000003' in stack
<--- Received SIP request (483 bytes) from UDP:172.16.1.1:5060 --->
ACK sip:192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK8gy0g2vysw2p28mgs8xg0s600;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153"<sip:0761604153@192.168.50.100;transport=udp;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515"<sip:0112019515@192.168.50.100;transport=udp;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 2 ACK
Max-Forwards: 23
Content-Length: 0


<--- Received SIP request (483 bytes) from UDP:172.16.1.1:5060 --->
ACK sip:192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK8gy0g2vysw2p28mgs8xg0s600;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153"<sip:0761604153@192.168.50.100;transport=udp;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515"<sip:0112019515@192.168.50.100;transport=udp;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 2 ACK
Max-Forwards: 23
Content-Length: 0


<--- Received SIP request (450 bytes) from UDP:172.20.10.136:5060 --->
BYE sip:asterisk@192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.136:5060;rport;branch=z9hG4bKPj6dc65907-e621-4ff8-b346-ae30f9da2939
From: <sip:172.20.10.136>;tag=9292c238-1d80-4936-b2eb-7bb7dc03075a
To: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 16447 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.16.0
Content-Length:  0


<--- Received SIP request (450 bytes) from UDP:172.20.10.136:5060 --->
BYE sip:asterisk@192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.136:5060;rport;branch=z9hG4bKPj6dc65907-e621-4ff8-b346-ae30f9da2939
From: <sip:172.20.10.136>;tag=9292c238-1d80-4936-b2eb-7bb7dc03075a
To: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
CSeq: 16447 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.16.0
Content-Length:  0


<--- Transmitting SIP response (402 bytes) to UDP:172.20.10.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.20.10.136:5060;rport=5060;received=172.20.10.136;branch=z9hG4bKPj6dc65907-e621-4ff8-b346-ae30f9da2939
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
From: <sip:172.20.10.136>;tag=9292c238-1d80-4936-b2eb-7bb7dc03075a
To: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
CSeq: 16447 BYE
Server: Asterisk PBX 20.14.0
Content-Length:  0


<--- Transmitting SIP response (402 bytes) to UDP:172.20.10.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.20.10.136:5060;rport=5060;received=172.20.10.136;branch=z9hG4bKPj6dc65907-e621-4ff8-b346-ae30f9da2939
Call-ID: 04e51b8e-9ada-4b80-add7-b65bd7bd6811
From: <sip:172.20.10.136>;tag=9292c238-1d80-4936-b2eb-7bb7dc03075a
To: <sip:0761604153@192.168.50.100>;tag=f6dc5a3d-1fa3-4d0d-a9b2-de986e86734b
CSeq: 16447 BYE
Server: Asterisk PBX 20.14.0
Content-Length:  0


    -- Channel PJSIP/forward136-00000003 left 'native_rtp' basic-bridge <518b178f-4bb8-4f1a-9f8f-746fba92cf85>
    -- Channel PJSIP/forward136-00000003 left 'native_rtp' basic-bridge <518b178f-4bb8-4f1a-9f8f-746fba92cf85>
    -- Channel PJSIP/gw-oneaccess-00000002 left 'native_rtp' basic-bridge <518b178f-4bb8-4f1a-9f8f-746fba92cf85>
    -- Channel PJSIP/gw-oneaccess-00000002 left 'native_rtp' basic-bridge <518b178f-4bb8-4f1a-9f8f-746fba92cf85>
    -- Executing [0112019515@testing:3] Hangup("PJSIP/gw-oneaccess-00000002", "") in new stack
    -- Executing [0112019515@testing:3] Hangup("PJSIP/gw-oneaccess-00000002", "") in new stack
  == Spawn extension (testing, 0112019515, 3) exited non-zero on 'PJSIP/gw-oneaccess-00000002'
  == Spawn extension (testing, 0112019515, 3) exited non-zero on 'PJSIP/gw-oneaccess-00000002'
<--- Transmitting SIP request (657 bytes) to TCP:172.16.1.1:5060 --->
BYE sip:172.16.1.1:5060;transport=udp;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff SIP/2.0
Via: SIP/2.0/TCP 192.168.50.100:5060;rport;branch=z9hG4bKPj4312ecb5-6227-42a1-b420-73b72b5b8fd5;alias
From: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
To: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
CSeq: 19609 BYE
Route: <sip:172.16.1.1:5060;lr;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=12297>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Content-Length:  0


<--- Transmitting SIP request (657 bytes) to TCP:172.16.1.1:5060 --->
BYE sip:172.16.1.1:5060;transport=udp;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff SIP/2.0
Via: SIP/2.0/TCP 192.168.50.100:5060;rport;branch=z9hG4bKPj4312ecb5-6227-42a1-b420-73b72b5b8fd5;alias
From: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
To: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
CSeq: 19609 BYE
Route: <sip:172.16.1.1:5060;lr;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=12297>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.14.0
Content-Length:  0


<--- Received SIP request (535 bytes) from UDP:172.16.1.1:5060 --->
BYE sip:192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK6msg87pxg8r22v028vmgs80pg;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153"<sip:0761604153@192.168.50.100;transport=udp;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515"<sip:0112019515@192.168.50.100;transport=udp;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 3 BYE
Max-Forwards: 23
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0


<--- Received SIP request (535 bytes) from UDP:172.16.1.1:5060 --->
BYE sip:192.168.50.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK6msg87pxg8r22v028vmgs80pg;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153"<sip:0761604153@192.168.50.100;transport=udp;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515"<sip:0112019515@192.168.50.100;transport=udp;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 3 BYE
Max-Forwards: 23
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0


<--- Transmitting SIP response (480 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bK6msg87pxg8r22v028vmgs80pg;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 3 BYE
Server: Asterisk PBX 20.14.0
Content-Length:  0


<--- Transmitting SIP response (480 bytes) to UDP:172.16.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;rport=5060;received=172.16.1.1;branch=z9hG4bK6msg87pxg8r22v028vmgs80pg;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff
Call-ID: isbc4uhxhc67ggkkclcu4l5vxhwyr5rchkv6@10.18.5.64
From: "0761604153" <sip:0761604153@192.168.50.100;user=phone>;tag=5g57kr6e-CC-1007-OFC-1391
To: "0112019515" <sip:0112019515@192.168.50.100;user=phone>;tag=2c6e02cd-35b9-4172-b502-6ef178407f82
CSeq: 3 BYE
Server: Asterisk PBX 20.14.0
Content-Length:  0

david551 · November 14, 2025, 8:19am

I don’t know how you captured the logs, as everything is doubled up, but there are no time stamps, which suggests you didn’t use the log file, and not dialplan trace, which means your verbose level wasn’t high enough.

Without the time stamps I can’t tell if there is even a problem,or the log is simply cut short, but the system producing the logs has definitely hung up, but is failing to get a confirmation of that from the other system., so you need the logs from that other system, which is also Asterisk.

You also have Record Route logged, which means you have a complex network, or a misconfiguration, but you haven’t describe that network, or provided the conifguration.

system · December 14, 2025, 8:19am

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