<--- Received SIP request (312 bytes) from UDP:127.0.0.1:6101 --->
REGISTER sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:6101;branch=z9hG4bK-184368-1-0
From: <sip:1101@127.0.0.1:5060>;tag=184368SIPpTag001
To: <sip:1101@127.0.0.1:5060>
Contact: <sip:1101@127.0.0.1:6101;transport=UDP>
Expires: 10
Call-ID: 1-184368@127.0.0.1
CSeq: 1 REGISTER
Content-Length: 0


<--- Transmitting SIP response (458 bytes) to UDP:127.0.0.1:6101 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:6101;rport=6101;received=127.0.0.1;branch=z9hG4bK-184368-1-0
Call-ID: 1-184368@127.0.0.1
From: <sip:1101@127.0.0.1>;tag=184368SIPpTag001
To: <sip:1101@127.0.0.1>;tag=z9hG4bK-184368-1-0
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1654259670/a3abb5121c2092bba42b1a13dcef5f2b",opaque="27be19055e767599",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.12.1
Content-Length:  0


<--- Received SIP request (576 bytes) from UDP:127.0.0.1:6101 --->
REGISTER sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:6101;branch=z9hG4bK-184368-1-4
From: <sip:1101@127.0.0.1:5060>;tag=184368SIPpTag001
To: <sip:1101@127.0.0.1:5060>
Contact: <sip:1101@127.0.0.1:6101;transport=UDP>
Expires: 10
Authorization: Digest username="3605657CFB45",realm="asterisk",cnonce="6b8b4567",nc=00000001,qop=auth,uri="sip:127.0.0.1:5060",nonce="1654259670/a3abb5121c2092bba42b1a13dcef5f2b",response="f6edc6da92dcc5fe0cb381d6ce1f1ebd",algorithm=MD5,opaque="27be19055e767599"
Call-ID: 1-184368@127.0.0.1
CSeq: 2 REGISTER
Content-Length: 0


    -- Added contact 'sip:1101@127.0.0.1:6101;transport=UDP' to AOR '1101' with expiration of 60 seconds
<--- Transmitting SIP response (413 bytes) to UDP:127.0.0.1:6101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:6101;rport=6101;received=127.0.0.1;branch=z9hG4bK-184368-1-4
Call-ID: 1-184368@127.0.0.1
From: <sip:1101@127.0.0.1>;tag=184368SIPpTag001
To: <sip:1101@127.0.0.1>;tag=z9hG4bK-184368-1-4
CSeq: 2 REGISTER
Date: Fri, 03 Jun 2022 12:34:30 GMT
Contact: <sip:1101@127.0.0.1:6101;transport=UDP>;expires=59
Expires: 60
Server: Asterisk PBX 18.12.1
Content-Length:  0


  == Endpoint 1101 is now Reachable
<--- Received SIP request (484 bytes) from UDP:127.0.0.1:5062 --->
INVITE sip:1101@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK-184391-1-0
From: sip:1102@127.0.0.1:5062;tag=184391SIPpTag001
To: sip:1101@127.0.0.1:5060
Call-ID: 1-184391@127.0.0.1
CSeq: 1 INVITE
Contact: <sip:1102@127.0.0.1:5062;transport=UDP>
Content-Type: application/sdp
Content-Length:   153

v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 6000 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

<--- Transmitting SIP response (456 bytes) to UDP:127.0.0.1:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;received=127.0.0.1;branch=z9hG4bK-184391-1-0
Call-ID: 1-184391@127.0.0.1
From: <sip:1102@127.0.0.1>;tag=184391SIPpTag001
To: <sip:1101@127.0.0.1>;tag=z9hG4bK-184391-1-0
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1654259671/187d7f22fd7f6d69666273d85debed81",opaque="0c7133771a4fb3bf",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.12.1
Content-Length:  0


<--- Received SIP request (307 bytes) from UDP:127.0.0.1:5062 --->
ACK sip:1101@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK-184391-1-0
From: <sip:1102@127.0.0.1>;tag=184391SIPpTag001
To: <sip:1101@127.0.0.1>;tag=z9hG4bK-184391-1-0
Call-ID: 1-184391@127.0.0.1
CSeq: 1 ACK
Contact: <sip:1102@127.0.0.1:5062;transport=UDP>
Content-Length: 0


<--- Received SIP request (748 bytes) from UDP:127.0.0.1:5062 --->
INVITE sip:1101@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK-184391-1-5
From: sip:1102@127.0.0.1:5062;tag=184391SIPpTag001
To: sip:1101@127.0.0.1:5060
Authorization: Digest username="558EF2645DC7",realm="asterisk",cnonce="6b8b4567",nc=00000001,qop=auth,uri="sip:127.0.0.1:5060",nonce="1654259671/187d7f22fd7f6d69666273d85debed81",response="5c524b834853e4370a562f6e173ef35c",algorithm=MD5,opaque="0c7133771a4fb3bf"
Call-ID: 1-184391@127.0.0.1
CSeq: 2 INVITE
Contact: <sip:1102@127.0.0.1:5062;transport=UDP>
Content-Type: application/sdp
Content-Length:   153

v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 6000 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

<--- Transmitting SIP response (281 bytes) to UDP:127.0.0.1:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;received=127.0.0.1;branch=z9hG4bK-184391-1-5
Call-ID: 1-184391@127.0.0.1
From: <sip:1102@127.0.0.1>;tag=184391SIPpTag001
To: <sip:1101@127.0.0.1>
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0


    -- ExecutingNoOp("PJSIP/1102-00000000", "") in new stack
    -- ExecutingSet("PJSIP/1102-00000000", "CDR_PROP(disable)=1") in new stack
    -- ExecutingGoto("PJSIP/1102-00000000", "Internal-Main,1101,1") in new stack
    -- Goto (Internal-Main,1101,1)
    -- Executingain:1] Verbose("PJSIP/1102-00000000", "1, "User 1102 dialed 1101."") in new stack
  "User 1102 dialed 1101."
    -- Executingain:2] Set("PJSIP/1102-00000000", "SAC_DIALED_EXTEN=1101") in new stack
    -- Executingain:3] GotoIf("PJSIP/1102-00000000", "0?dialed-BUSY,1:") in new stack
    -- Executingain:4] Dial("PJSIP/1102-00000000", "PJSIP/1101,30") in new stack
    -- Called PJSIP/1101
<--- Transmitting SIP request (932 bytes) to UDP:127.0.0.1:6101 --->
INVITE sip:1101@127.0.0.1:6101;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj62dbb8fd-9fea-40ef-ad86-0f068fe57839
From: "Tommie Briar" <sip:1102@192.168.2.29>;tag=8814086a-6d27-4c24-a7be-cad6dc0b5b25
To: <sip:1101@127.0.0.1>
Contact: <sip:asterisk@127.0.0.1:5060>
Call-ID: 6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 5900 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.12.1
Content-Type: application/sdp
Content-Length:   255

v=0
o=- 1823610345 1823610345 IN IP4 127.0.0.1
s=Asterisk
c=IN IP4 127.0.0.1
t=0 0
m=audio 11716 RTP/AVP 0 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (375 bytes) from UDP:127.0.0.1:6101 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj62dbb8fd-9fea-40ef-ad86-0f068fe57839
From: "Tommie Briar" <sip:1102@192.168.2.29>;tag=8814086a-6d27-4c24-a7be-cad6dc0b5b25
To: <sip:1101@127.0.0.1>;tag=184389SIPpTag011
Call-ID: 6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 5900 INVITE
Contact: sip:service@127.0.0.1:6101
Content-Length:     0


    -- PJSIP/1101-00000001 is ringing
<--- Transmitting SIP response (465 bytes) to UDP:127.0.0.1:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;received=127.0.0.1;branch=z9hG4bK-184391-1-5
Call-ID: 1-184391@127.0.0.1
From: <sip:1102@127.0.0.1>;tag=184391SIPpTag001
To: <sip:1101@127.0.0.1>;tag=bbe6af1e-0f41-473a-b090-281f7d738bc5
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Contact: <sip:127.0.0.1:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP request (639 bytes) from UDP:127.0.0.1:5061 --->
INVITE sip:1101@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-184387-1-2
From: sip:1103@127.0.0.1:5061;tag=184387SIPpTag001
To: sip:1101@127.0.0.1:5060
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 1 INVITE
Contact: <sip:1103@127.0.0.1:5061;transport=UDP>
Replaces: 6e4bff9c-f36a-4ad4-b883-7d04c65da68e;to-tag=8814086a-6d27-4c24-a7be-cad6dc0b5b25;from-tag=184389SIPpTag011;early-only
Content-Type: application/sdp
Content-Length:   153

v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 6000 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

<--- Transmitting SIP response (482 bytes) to UDP:127.0.0.1:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5061;rport=5061;received=127.0.0.1;branch=z9hG4bK-184387-1-2
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
From: <sip:1103@127.0.0.1>;tag=184387SIPpTag001
To: <sip:1101@127.0.0.1>;tag=z9hG4bK-184387-1-2
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1654259672/a36b9b925c2766931173445ba6be05d3",opaque="6a5b696d2dd78d72",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.12.1
Content-Length:  0


<--- Received SIP request (317 bytes) from UDP:127.0.0.1:5061 --->
ACK sip:1101@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-184387-1-2
From: <sip:1103@127.0.0.1>;tag=184387SIPpTag001
To: <sip:1101@127.0.0.1>;tag=z9hG4bK-184387-1-2
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 1 ACK
Contact: sip:1103@127.0.0.1:5061
Content-Length: 0


<--- Received SIP request (903 bytes) from UDP:127.0.0.1:5061 --->
INVITE sip:1101@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-184387-1-7
From: sip:1103@127.0.0.1:5061;tag=184387SIPpTag001
To: sip:1101@127.0.0.1:5060
Authorization: Digest username="D5F646797302",realm="asterisk",cnonce="6b8b4567",nc=00000001,qop=auth,uri="sip:127.0.0.1:5060",nonce="1654259672/a36b9b925c2766931173445ba6be05d3",response="24c0687d7796263ac947a9e2b5192e10",algorithm=MD5,opaque="6a5b696d2dd78d72"
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 2 INVITE
Contact: <sip:1103@127.0.0.1:5061;transport=UDP>
Replaces: 6e4bff9c-f36a-4ad4-b883-7d04c65da68e;to-tag=8814086a-6d27-4c24-a7be-cad6dc0b5b25;from-tag=184389SIPpTag011;early-only
Content-Type: application/sdp
Content-Length:   153

v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 6000 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

<--- Transmitting SIP response (307 bytes) to UDP:127.0.0.1:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5061;rport=5061;received=127.0.0.1;branch=z9hG4bK-184387-1-7
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
From: <sip:1103@127.0.0.1>;tag=184387SIPpTag001
To: <sip:1101@127.0.0.1>
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0


       > 0x7f35e808bb90 -- Strict RTP learning after remote address set to: 127.0.0.1:6000
<--- Transmitting SIP response (740 bytes) to UDP:127.0.0.1:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5061;rport=5061;received=127.0.0.1;branch=z9hG4bK-184387-1-7
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
From: <sip:1103@127.0.0.1>;tag=184387SIPpTag001
To: <sip:1101@127.0.0.1>;tag=60a38516-b0c3-40f6-9564-6fe3d8d81c71
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Contact: <sip:127.0.0.1:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   172

v=0
o=- 53655765 2353687639 IN IP4 127.0.0.1
s=Asterisk
c=IN IP4 127.0.0.1
t=0 0
m=audio 9352 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (337 bytes) from UDP:127.0.0.1:5061 --->
ACK sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-184387-1-10
From: sip:1103@127.0.0.1:5061;tag=184387SIPpTag001
To: sip:1101@127.0.0.1:5060;tag=60a38516-b0c3-40f6-9564-6fe3d8d81c71
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 2 ACK
Contact: sip:1103@127.0.0.1:5061
Content-Length: 0


    -- PJSIP/1103-00000002 answered PJSIP/1102-00000000
<--- Transmitting SIP request (422 bytes) to UDP:127.0.0.1:6101 --->
CANCEL sip:1101@127.0.0.1:6101;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj62dbb8fd-9fea-40ef-ad86-0f068fe57839
From: "Tommie Briar" <sip:1102@192.168.2.29>;tag=8814086a-6d27-4c24-a7be-cad6dc0b5b25
To: <sip:1101@127.0.0.1>
Call-ID: 6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 5900 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: Asterisk PBX 18.12.1
Content-Length:  0


       > 0x7f35e801e230 -- Strict RTP learning after remote address set to: 127.0.0.1:6000
<--- Received SIP response (308 bytes) from UDP:127.0.0.1:6101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj62dbb8fd-9fea-40ef-ad86-0f068fe57839
From: "Tommie Briar" <sip:1102@192.168.2.29>;tag=8814086a-6d27-4c24-a7be-cad6dc0b5b25
To: <sip:1101@127.0.0.1>
Call-ID: 6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 5900 CANCEL
Content-Length: 0


<--- Transmitting SIP response (715 bytes) to UDP:127.0.0.1:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;received=127.0.0.1;branch=z9hG4bK-184391-1-5
Call-ID: 1-184391@127.0.0.1
From: <sip:1102@127.0.0.1>;tag=184391SIPpTag001
To: <sip:1101@127.0.0.1>;tag=bbe6af1e-0f41-473a-b090-281f7d738bc5
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:127.0.0.1:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   173

v=0
o=- 53655765 2353687639 IN IP4 127.0.0.1
s=Asterisk
c=IN IP4 127.0.0.1
t=0 0
m=audio 21754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv[1;30m
To: sip:1101@127.0.0.1:5060;tag=bbe6af1e-0f41-473a-b090-281f7d738bc5
Call-ID: 1-184391@127.0.0.1
CSeq: 2 ACK
Contact: <sip:1102@127.0.0.1:5062;transport=UDP>
Content-Length: 0


    -- Channel PJSIP/1103-00000002 joined 'simple_bridge' basic-bridge <d9a3b1da-a758-49f8-83e0-9b623267a8c5>
    -- Channel PJSIP/1102-00000000 joined 'simple_bridge' basic-bridge <d9a3b1da-a758-49f8-83e0-9b623267a8c5>
       > Bridge d9a3b1da-a758-49f8-83e0-9b623267a8c5: switching from simple_bridge technology to native_rtp
<--- Received SIP response (345 bytes) from UDP:127.0.0.1:6101 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj62dbb8fd-9fea-40ef-ad86-0f068fe57839
From: "Tommie Briar" <sip:1102@192.168.2.29>;tag=8814086a-6d27-4c24-a7be-cad6dc0b5b25
To: <sip:1101@127.0.0.1>;tag=184389SIPpTag011
Call-ID: 6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 5900 INVITE
Content-Length: 0


       > Locally RTP bridged 'PJSIP/1102-00000000' and 'PJSIP/1103-00000002' in stack
<--- Transmitting SIP request (414 bytes) to UDP:127.0.0.1:6101 --->
ACK sip:1101@127.0.0.1:6101;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj62dbb8fd-9fea-40ef-ad86-0f068fe57839
From: "Tommie Briar" <sip:1102@192.168.2.29>;tag=8814086a-6d27-4c24-a7be-cad6dc0b5b25
To: <sip:1101@127.0.0.1>;tag=184389SIPpTag011
Call-ID: 6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 5900 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.12.1
Content-Length:  0


       > 0x7f35e808bb90 -- Strict RTP switching to RTP target address 127.0.0.1:6000 as source
       > 0x7f35e801e230 -- Strict RTP qualifying stream type: audio
       > 0x7f35e801e230 -- Strict RTP switching source address to 127.0.0.1:6002
<--- Received SIP request (347 bytes) from UDP:127.0.0.1:5061 --->
BYE sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-184387-1-13
From: <sip:1103@127.0.0.1>;tag=184387SIPpTag001
To: <sip:1101@127.0.0.1>;tag=60a38516-b0c3-40f6-9564-6fe3d8d81c71
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
CSeq: 3 BYE
Contact: <sip:1103@127.0.0.1:5061;transport=UDP>
Content-Length: 0


<--- Transmitting SIP response (342 bytes) to UDP:127.0.0.1:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5061;rport=5061;received=127.0.0.1;branch=z9hG4bK-184387-1-13
Call-ID: dedup///6e4bff9c-f36a-4ad4-b883-7d04c65da68e
From: <sip:1103@127.0.0.1>;tag=184387SIPpTag001
To: <sip:1101@127.0.0.1>;tag=60a38516-b0c3-40f6-9564-6fe3d8d81c71
CSeq: 3 BYE
Server: Asterisk PBX 18.12.1
Content-Length:  0


    -- Channel PJSIP/1103-00000002 left 'native_rtp' basic-bridge <d9a3b1da-a758-49f8-83e0-9b623267a8c5>
    -- Channel PJSIP/1102-00000000 left 'native_rtp' basic-bridge <d9a3b1da-a758-49f8-83e0-9b623267a8c5>
  == Spawn extension (Internal-Main, 1101, 4) exited non-zero on 'PJSIP/1102-00000000'
    -- Executingain:1] Hangup("PJSIP/1102-00000000", "") in new stack
  == Spawn extension (Internal-Main, h, 1) exited non-zero on 'PJSIP/1102-00000000'
<--- Transmitting SIP request (403 bytes) to UDP:127.0.0.1:5062 --->
BYE sip:1102@127.0.0.1:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj80135594-99b2-4fc6-8d28-349926cc733c
From: <sip:1101@127.0.0.1>;tag=bbe6af1e-0f41-473a-b090-281f7d738bc5
To: <sip:1102@127.0.0.1>;tag=184391SIPpTag001
Call-ID: 1-184391@127.0.0.1
CSeq: 24683 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.12.1
Content-Length:  0


<--- Received SIP response (291 bytes) from UDP:127.0.0.1:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj80135594-99b2-4fc6-8d28-349926cc733c
From: <sip:1101@127.0.0.1>;tag=bbe6af1e-0f41-473a-b090-281f7d738bc5
To: <sip:1102@127.0.0.1>;tag=184391SIPpTag001
Call-ID: 1-184391@127.0.0.1
CSeq: 24683 BYE
Content-Length: 0