Untitled


```
<--- Received SIP response (492 bytes) from TLS:5.162.93.198:59422 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 5.37.215.237:5061;rport=5061;branch=z9hG4bKPj03719e65-e129-4dfc-9342-b88130248d1d;alias
From: <sip:141@192.168.1.17>;tag=17913ecc-3419-40b3-85dc-7811a706d050
To: <sip:141@5.162.93.198>;tag=799645839
Call-ID: 22ac6fa7-4fba-4daf-99ad-808b5d499427
CSeq: 64710 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (751 bytes) from UDP:192.168.1.161:5060 --->
INVITE sip:601@192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK354464728
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
To: <sip:601@192.168.1.17>
Call-ID: 631126076
CSeq: 20 INVITE
Contact: <sip:161@192.168.1.161:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length:   384

v=0
o=dnake 2100420779 2100420779 IN IP4 192.168.1.161
s=dnake
c=IN IP4 192.168.1.161
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42001F; packetization-mode=1
a=ex_fmtp:102 2CIF=1
a=sendrecv

<--- Transmitting SIP response (461 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK354464728
Call-ID: 631126076
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
To: <sip:601@192.168.1.17>;tag=z9hG4bK354464728
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1648107000/45e62a69faab6447076956fa02e3295d",opaque="1fa313eb06ed8b7f",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.5.1
Content-Length:  0


<--- Received SIP request (292 bytes) from UDP:192.168.1.161:5060 --->
ACK sip:601@192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK354464728
Route: <sip:192.168.1.17;lr>
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
To: <sip:601@192.168.1.17>;tag=z9hG4bK354464728
Call-ID: 631126076
CSeq: 20 ACK
Content-Length: 0


<--- Received SIP request (1017 bytes) from UDP:192.168.1.161:5060 --->
INVITE sip:601@192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK752274229
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
To: <sip:601@192.168.1.17>
Call-ID: 631126076
CSeq: 21 INVITE
Contact: <sip:161@192.168.1.161:5060>
Authorization: Digest username="161", realm="asterisk", nonce="1648107000/45e62a69faab6447076956fa02e3295d", uri="sip:601@192.168.1.17", response="038c7e84588e8505ddff8afafbafa6f0", algorithm=MD5, cnonce="0a4f113b", opaque="1fa313eb06ed8b7f", qop=auth, nc=00000001
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length:   384

v=0
o=dnake 2100420779 2100420779 IN IP4 192.168.1.161
s=dnake
c=IN IP4 192.168.1.161
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 102
<--- Transmitting SIP response (288 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK752274229
Call-ID: 631126076
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
To: <sip:601@192.168.1.17>
CSeq: 21 INVITE
Server: Asterisk PBX 18.5.1
Content-Length:  0


  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Video TOS bits 136
  == Using SIP RTP Video TOS bits 136 in TCLASS field.
    -- Executing [601@default:1] NoOp("PJSIP/161-00000018", "EXECUTING 601 call") in new stack
    -- Executing [601@default:2] Dial("PJSIP/161-00000018", "PJSIP/101&PJSIP/102&PJSIP/103&PJSIP/104&PJSIP/105&PJSIP/106&PJSIP/107&PJSIP/108&PJSIP/109&PJSIP/110&PJSIP/111&PJSIP/112&PJSIP/113&PJSIP/114&PJSIP/115&PJSIP/116&Local/mobilephones@default") in new stack
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '101': Could not create dialog to invalid URI '101'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '101'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '102': Could not create dialog to invalid URI '102'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '102'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '104': Could not create dialog to invalid URI '104'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '104'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '105': Could not create dialog to invalid URI '105'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '105'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '106': Could not create dialog to invalid URI '106'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '106'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '107': Could not create dialog to invalid URI '107'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '107'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '108': Could not create dialog to invalid URI '108'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '108'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '109': Could not create dialog to invalid URI '109'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '109'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '110': Could not create dialog to invalid URI '110'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '110'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '111': Could not create dialog to invalid URI '111'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '111'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '112': Could not create dialog to invalid URI '112'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '112'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '113': Could not create dialog to invalid URI '113'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '113'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '114': Could not create dialog to invalid URI '114'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '114'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '115': Could not create dialog to invalid URI '115'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '115'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Mar 24 07:30:00] ERROR[2721]: res_pjsip.c:4053 ast_sip_create_dialog_uac: Endpoint '116': Could not create dialog to invalid URI '116'.  Is endpoint registered and reachable?
[Mar 24 07:30:00] ERROR[2721]: chan_pjsip.c:2656 request: Failed to create outgoing session to endpoint '116'
[Mar 24 07:30:00] WARNING[7045][C-0000000a]: app_dial.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
    -- Called PJSIP/103
    -- Called Local/mobilephones@default
    -- Executing [mobilephones@default:1] Ringing("Local/mobilephones@default-00000006;2", "") in new stack
    -- Local/mobilephones@default-00000006;1 is ringing
<--- Transmitting SIP response (475 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK752274229
Call-ID: 631126076
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
To: <sip:601@192.168.1.17>;tag=7bcc20e0-e9d3-4561-8197-dd77a0c0ae01
CSeq: 21 INVITE
Server: Asterisk PBX 18.5.1
Contact: <sip:192.168.1.17:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


    -- Executing [mobilephones@default:2] System("Local/mobilephones@default-00000006;2", "/bin/sleep 6") in new stack
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Video TOS bits 136
  == Using SIP RTP Video TOS bits 136 in TCLASS field.
    -- PJSIP/103-00000019 connected line has changed. Saving it until answer for PJSIP/161-00000018
<--- Transmitting SIP request (1191 bytes) to UDP:192.168.1.101:5062 --->
INVITE sip:103@192.168.1.101:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj791cf278-e918-4b10-8126-f63fe21694b2
From: "161" <sip:Door_2@192.168.1.17>;tag=9ba4f9a0-eb51-43d9-8f46-1e4a9a35d9b3
To: <sip:103@192.168.1.101>
Contact: <sip:asterisk@192.168.1.17:5060>
Call-ID: 167b90fb-6724-451d-a6cc-7752845c30a3
CSeq: 10533 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
P-Asserted-Identity: "161" <sip:Door_2@192.168.1.17>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Type: application/sdp
Content-Length:   469

v=0
o=- 235227286 235227286 IN IP4 192.168.1.17
s=Asterisk
c=IN IP4 192.168.1.17
t=0 0
m=audio 32678 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtp<--- Received SIP response (489 bytes) from UDP:192.168.1.101:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj791cf278-e918-4b10-8126-f63fe21694b2
From: "161" <sip:Door_2@192.168.1.17>;tag=9ba4f9a0-eb51-43d9-8f46-1e4a9a35d9b3
To: <sip:103@192.168.1.101>
Call-ID: 167b90fb-6724-451d-a6cc-7752845c30a3
CSeq: 10533 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.227
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (576 bytes) from UDP:192.168.1.101:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj791cf278-e918-4b10-8126-f63fe21694b2
From: "161" <sip:Door_2@192.168.1.17>;tag=9ba4f9a0-eb51-43d9-8f46-1e4a9a35d9b3
To: <sip:103@192.168.1.101>;tag=234108663
Call-ID: 167b90fb-6724-451d-a6cc-7752845c30a3
CSeq: 10533 INVITE
Contact: <sip:103@192.168.1.101:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.227
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- PJSIP/103-00000019 is ringing
    -- Executing [mobilephones@default:3] Dial("Local/mobilephones@default-00000006;2", "PJSIP/140/sip:140@5.37.215.237:46854;transport=TLS;x-ast-orig-host=192.168.1.194:46854&PJSIP/141/sip:141@5.162.93.198:59422;transport=TLS;x-ast-orig-host=10.178.154.185:44184") in new stack
    -- Called PJSIP/140/sip:140@5.37.215.237:46854;transport=TLS;x-ast-orig-host=192.168.1.194:46854
    -- Called PJSIP/141/sip:141@5.162.93.198:59422;transport=TLS;x-ast-orig-host=10.178.154.185:44184
<--- Transmitting SIP request (721 bytes) to TLS:5.37.215.237:46854 --->
INVITE sip:140@5.37.215.237:46854;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 5.37.215.237:5061;rport;branch=z9hG4bKPj4c969059-03a7-4fed-b9a8-f576df4ed5ce;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=6182cad9-8baa-46d0-a7c0-a9f0343c4b1a
To: <sip:140@5.37.215.237>
Contact: <sip:asterisk@5.37.215.237:5061;transport=TLS>
Call-ID: a3b7d64f-98f2-4f93-9129-6d93319e90a4
CSeq: 11297 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
P-Asserted-Identity: "161" <sip:Door_2@192.168.1.17>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Length:  0


    -- PJSIP/141-0000001b connected line has changed. Saving it until answer for Local/mobilephones@default-00000006;2
    -- PJSIP/140-0000001a connected line has changed. Saving it until answer for Local/mobilephones@default-00000006;2
<--- Transmitting SIP request (721 bytes) to TLS:5.162.93.198:59422 --->
INVITE sip:141@5.162.93.198:59422;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 5.37.215.237:5061;rport;branch=z9hG4bKPj701f105c-390e-4e3e-9c66-cc5903848944;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
To: <sip:141@5.162.93.198>
Contact: <sip:asterisk@5.37.215.237:5061;transport=TLS>
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
CSeq: 19365 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
P-Asserted-Identity: "161" <sip:Door_2@192.168.1.17>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Length:  0


<--- Received SIP response (490 bytes) from TLS:5.37.215.237:46854 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 5.37.215.237:5061;rport=5061;branch=z9hG4bKPj4c969059-03a7-4fed-b9a8-f576df4ed5ce;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=6182cad9-8baa-46d0-a7c0-a9f0343c4b1a
To: <sip:140@5.37.215.237>
Call-ID: a3b7d64f-98f2-4f93-9129-6d93319e90a4
CSeq: 11297 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (592 bytes) from TLS:5.37.215.237:46854 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 5.37.215.237:5061;rport=5061;branch=z9hG4bKPj4c969059-03a7-4fed-b9a8-f576df4ed5ce;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=6182cad9-8baa-46d0-a7c0-a9f0343c4b1a
To: <sip:140@5.37.215.237>;tag=221396994
Call-ID: a3b7d64f-98f2-4f93-9129-6d93319e90a4
CSeq: 11297 INVITE
Contact: <sip:140@192.168.1.194:46854;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- PJSIP/140-0000001a is ringing
    -- Local/mobilephones@default-00000006;1 is ringing
<--- Received SIP response (490 bytes) from TLS:5.162.93.198:59422 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 5.37.215.237:5061;rport=5061;branch=z9hG4bKPj701f105c-390e-4e3e-9c66-cc5903848944;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
To: <sip:141@5.162.93.198>
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
CSeq: 19365 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (593 bytes) from TLS:5.162.93.198:59422 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 5.37.215.237:5061;rport=5061;branch=z9hG4bKPj701f105c-390e-4e3e-9c66-cc5903848944;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
To: <sip:141@5.162.93.198>;tag=184629041
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
CSeq: 19365 INVITE
Contact: <sip:141@10.178.154.185:44184;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- PJSIP/141-0000001b is ringing
<--- Received SIP response (2143 bytes) from TLS:5.162.93.198:59422 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 5.37.215.237:5061;rport=5061;branch=z9hG4bKPj701f105c-390e-4e3e-9c66-cc5903848944;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
To: <sip:141@5.162.93.198>;tag=184629041
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
CSeq: 19365 INVITE
Contact: <sip:141@10.178.154.185:44184;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:  1493

v=0
o=141 8000 8000 IN IP4 10.178.154.185
s=SIP Call
c=IN IP4 10.178.154.185
t=0 0
m=audio 38652 RTP/SAVP 0 8 9 123 2 97 3 18 101
a=sendrecv
a=rtcp:38653 IN IP4 10.178.154.185
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:       > 0x13a0680 -- Strict RTP learning after remote address set to: 10.178.154.185:38652
       > 0x13a4350 -- Strict RTP learning after remote address set to: 10.178.154.185:44930
<--- Transmitting SIP request (954 bytes) to TLS:5.162.93.198:59422 --->
ACK sip:141@5.162.93.198:59422;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 5.37.215.237:5061;rport;branch=z9hG4bKPjd418ca0c-fb0d-4b63-b5eb-bfc5d848cab4;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
To: <sip:141@5.162.93.198>;tag=184629041
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
CSeq: 19365 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Type: application/sdp
Content-Length:   506

v=0
o=- 8000 8002 IN IP4 5.37.215.237
s=Asterisk
c=IN IP4 5.37.215.237
t=0 0
m=audio 33170 RTP/SAVP 0 101
a=crypto:1 AES_CM_256_HMAC_SHA1_80 inline:DQPe3AugDY/XMtDqo0X1NFtUhUUhclCWdO/l00o+E7EeF+nnPuwk4aqqXKPdYg==
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
m=video 30736 RTP/SAVP 105
a=crypto:1 AES_CM_256_HMAC_SHA1_80 inline:fQUFCLn4wExXdsWXJu5xow6z9863kaONBVOkbmp786mh8xhv/l/Wv28/eTJl4g==
a=rtp    -- PJSIP/141-0000001b answered Local/mobilephones@default-00000006;2
    -- Local/mobilephones@default-00000006;1 connected line has changed. Saving it until answer for PJSIP/161-00000018
    -- Local/mobilephones@default-00000006;1 answered PJSIP/161-00000018
<--- Transmitting SIP request (485 bytes) to TLS:5.37.215.237:46854 --->
CANCEL sip:140@5.37.215.237:46854;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 5.37.215.237:5061;rport;branch=z9hG4bKPj4c969059-03a7-4fed-b9a8-f576df4ed5ce;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=6182cad9-8baa-46d0-a7c0-a9f0343c4b1a
To: <sip:140@5.37.215.237>
Call-ID: a3b7d64f-98f2-4f93-9129-6d93319e90a4
CSeq: 11297 CANCEL
Reason: SIP;cause=200;text="Call completed elsewhere"
Reason: Q.850;cause=26
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Length:  0


<--- Transmitting SIP request (466 bytes) to UDP:192.168.1.101:5062 --->
CANCEL sip:103@192.168.1.101:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj791cf278-e918-4b10-8126-f63fe21694b2
From: "161" <sip:Door_2@192.168.1.17>;tag=9ba4f9a0-eb51-43d9-8f46-1e4a9a35d9b3
To: <sip:103@192.168.1.101>
Call-ID: 167b90fb-6724-451d-a6cc-7752845c30a3
CSeq: 10533 CANCEL
Reason: SIP;cause=200;text="Call completed elsewhere"
Reason: Q.850;cause=26
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Length:  0


<--- Received SIP response (545 bytes) from UDP:192.168.1.101:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj791cf278-e918-4b10-8126-f63fe21694b2
From: "161" <sip:Door_2@192.168.1.17>;tag=9ba4f9a0-eb51-43d9-8f46-1e4a9a35d9b3
To: <sip:103@192.168.1.101>;tag=234108663
Call-ID: 167b90fb-6724-451d-a6cc-7752845c30a3
CSeq: 10533 CANCEL
Contact: <sip:103@192.168.1.101:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.227
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (522 bytes) from UDP:192.168.1.101:5062 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj791cf278-e918-4b10-8126-f63fe21694b2
From: "161" <sip:Door_2@192.168.1.17>;tag=9ba4f9a0-eb51-43d9-8f46-1e4a9a35d9b3
To: <sip:103@192.168.1.101>;tag=234108663
Call-ID: 167b90fb-6724-451d-a6cc-7752845c30a3
CSeq: 10533 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.227
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


       > 0x12c37c0 -- Strict RTP learning after remote address set to: 192.168.1.161:6000
<--- Transmitting SIP request (395 bytes) to UDP:192.168.1.101:5062 --->
ACK sip:103@192.168.1.101:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj791cf278-e918-4b10-8126-f63fe21694b2
From: "161" <sip:Door_2@192.168.1.17>;tag=9ba4f9a0-eb51-43d9-8f46-1e4a9a35d9b3
To: <sip:103@192.168.1.101>;tag=234108663
Call-ID: 167b90fb-6724-451d-a6cc-7752845c30a3
CSeq: 10533 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Length:  0


       > 0x14207a0 -- Strict RTP learning after remote address set to: 192.168.1.161:6200
    -- Channel PJSIP/141-0000001b joined 'simple_bridge' basic-bridge <42e1d601-c580-4034-9135-40d8de382c03>
<--- Transmitting SIP response (939 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK752274229
Call-ID: 631126076
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
To: <sip:601@192.168.1.17>;tag=7bcc20e0-e9d3-4561-8197-dd77a0c0ae01
CSeq: 21 INVITE
Server: Asterisk PBX 18.5.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.17:5060>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "140" <sip:MobileExten141@192.168.1.17>
Content-Type: application/sdp
Content-Length:   325

v=0
o=- 2100420779 2100420781 IN IP4 192.168.1.17
s=Asterisk
c=IN IP4 192.168.1.17
t=0 0
m=audio 37054 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 34452 RTP/AVP 102
a=rtpmap:102 H264    -- Channel Local/mobilephones@default-00000006;2 joined 'simple_bridge' basic-bridge <42e1d601-c580-4034-9135-40d8de382c03>
    -- Channel Local/mobilephones@default-00000006;1 joined 'simple_bridge' basic-bridge <14af89e8-ec68-42c6-abff-2c824e4a6342>
    -- Channel PJSIP/161-00000018 joined 'simple_bridge' basic-bridge <14af89e8-ec68-42c6-abff-2c824e4a6342>
<--- Received SIP response (561 bytes) from TLS:5.37.215.237:46854 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 5.37.215.237:5061;rport=5061;branch=z9hG4bKPj4c969059-03a7-4fed-b9a8-f576df4ed5ce;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=6182cad9-8baa-46d0-a7c0-a9f0343c4b1a
To: <sip:140@5.37.215.237>;tag=221396994
Call-ID: a3b7d64f-98f2-4f93-9129-6d93319e90a4
CSeq: 11297 CANCEL
Contact: <sip:140@192.168.1.194:46854;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (523 bytes) from TLS:5.37.215.237:46854 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/TLS 5.37.215.237:5061;rport=5061;branch=z9hG4bKPj4c969059-03a7-4fed-b9a8-f576df4ed5ce;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=6182cad9-8baa-46d0-a7c0-a9f0343c4b1a
To: <sip:140@5.37.215.237>;tag=221396994
Call-ID: a3b7d64f-98f2-4f93-9129-6d93319e90a4
CSeq: 11297 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (414 bytes) to TLS:5.37.215.237:46854 --->
ACK sip:140@5.37.215.237:46854;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 5.37.215.237:5061;rport;branch=z9hG4bKPj4c969059-03a7-4fed-b9a8-f576df4ed5ce;alias
From: "161" <sip:Door_2@192.168.1.17>;tag=6182cad9-8baa-46d0-a7c0-a9f0343c4b1a
To: <sip:140@5.37.215.237>;tag=221396994
Call-ID: a3b7d64f-98f2-4f93-9129-6d93319e90a4
CSeq: 11297 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Length:  0


<--- Received SIP request (792 bytes) from TLS:5.162.93.198:59422 --->
INFO sip:asterisk@5.37.215.237:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.178.154.185:44184;branch=z9hG4bK2103662661;rport
From: <sip:141@5.162.93.198>;tag=184629041
To: <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
CSeq: 19366 INFO
Contact: <sip:141@10.178.154.185:44184;transport=tls>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/media_control+xml
Content-Length:   164

<?xml version="1.0" encoding="utf-8" ?><media_control>  <vc_primitive>    <to_encoder>      <picture_fast_update/>    </to_encoder>  </vc_primitive></media_control>
<--- Transmitting SIP response (346 bytes) to TLS:5.162.93.198:59422 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.178.154.185:44184;rport=59422;received=5.162.93.198;branch=z9hG4bK2103662661
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
From: <sip:141@5.162.93.198>;tag=184629041
To: <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
CSeq: 19366 INFO
Server: Asterisk PBX 18.5.1
Content-Length:  0


<--- Transmitting SIP request (599 bytes) to UDP:192.168.1.161:5060 --->
INFO sip:161@192.168.1.161:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj1ffcd4ce-9c83-4977-8d25-ee1ec3e37811
From: <sip:601@192.168.1.17>;tag=7bcc20e0-e9d3-4561-8197-dd77a0c0ae01
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
Call-ID: 631126076
CSeq: 11727 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>

<--- Received SIP request (369 bytes) from UDP:192.168.1.161:5060 --->
ACK sip:192.168.1.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK1551013296
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
To: <sip:601@192.168.1.17>;tag=7bcc20e0-e9d3-4561-8197-dd77a0c0ae01
Call-ID: 631126076
CSeq: 21 ACK
Contact: <sip:161@192.168.1.161:5060>
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Received SIP response (368 bytes) from UDP:192.168.1.161:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj1ffcd4ce-9c83-4977-8d25-ee1ec3e37811
From: <sip:601@192.168.1.17>;tag=7bcc20e0-e9d3-4561-8197-dd77a0c0ae01
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
Call-ID: 631126076
CSeq: 11727 INFO
Contact: <sip:161@192.168.1.161:5060>
User-Agent: DnakeVoip v1.0
Content-Length: 0


       > 0x14207a0 -- Strict RTP switching to RTP target address 192.168.1.161:6200 as source
       > Move-swap optimizing Local/mobilephones@default-00000006;2 <-- PJSIP/161-00000018.
    -- Channel PJSIP/161-00000018 left 'simple_bridge' basic-bridge <14af89e8-ec68-42c6-abff-2c824e4a6342>
    -- Channel Local/mobilephones@default-00000006;2 left 'simple_bridge' basic-bridge <42e1d601-c580-4034-9135-40d8de382c03>
    -- Channel PJSIP/161-00000018 swapped with Local/mobilephones@default-00000006;2 into 'simple_bridge' basic-bridge <42e1d601-c580-4034-9135-40d8de382c03>
  == Spawn extension (default, mobilephones, 3) exited non-zero on 'Local/mobilephones@default-00000006;2'
    -- Channel Local/mobilephones@default-00000006;1 left 'simple_bridge' basic-bridge <14af89e8-ec68-42c6-abff-2c824e4a6342>
       > 0x12c37c0 -- Strict RTP switching to RTP target address 192.168.1.161:6000 as source
       > 0x13a0680 -- Strict RTP qualifying stream type: audio
       > 0x13a0680 -- Strict RTP switching source address to 5.162.93.198:59479
  == SRTP unprotect failed on SSRC 1977138230 because of authentication failure 10
       > 0x13a4350 -- Strict RTP qualifying stream type: video
  == SRTCP unprotect failed on SSRC 748447341 because of unsupported parameter
  == SRTP unprotect failed on SSRC 1977138230 because of authentication failure 160
       > 0x13a0680 -- Strict RTP learning complete - Locking on source address 5.162.93.198:59479
       > 0x12c37c0 -- Strict RTP learning complete - Locking on source address 192.168.1.161:6000
       > 0x14207a0 -- Strict RTP learning complete - Locking on source address 192.168.1.161:6200
  == SRTCP unprotect failed on SSRC 1977138230 because of authentication failure
  == SRTCP unprotect failed on SSRC 748447341 because of unable to perform desired validation
  == SRTP unprotect failed on SSRC 1977138230 because of authentication failure 160
<--- Received SIP request (301 bytes) from UDP:192.168.1.161:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK559604632
From: <sip:161@192.168.1.17>;tag=1848887314
To: <sip:192.168.1.17>
Call-ID: 2061239627
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (447 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK559604632
Call-ID: 2061239627
From: <sip:161@192.168.1.17>;tag=1848887314
To: <sip:192.168.1.17>;tag=z9hG4bK559604632
CSeq: 20 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1648107018/92d1c77fdc301db10c2c5774a362001f",opaque="64c7407d324e71f2",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.5.1
Content-Length:  0


<--- Received SIP request (563 bytes) from UDP:192.168.1.161:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK741077750
From: <sip:161@192.168.1.17>;tag=1848887314
To: <sip:192.168.1.17>
Call-ID: 2061239627
CSeq: 21 OPTIONS
Authorization: Digest username="161", realm="asterisk", nonce="1648107018/92d1c77fdc301db10c2c5774a362001f", uri="sip:192.168.1.17", response="879a58256b59d2ac5b17d2505c30f794", algorithm=MD5, cnonce="0a4f113b", opaque="64c7407d324e71f2", qop=auth, nc=00000001
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (776 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK741077750
Call-ID: 2061239627
From: <sip:161@192.168.1.17>;tag=1848887314
To: <sip:192.168.1.17>;tag=z9hG4bK741077750
CSeq: 21 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 18.5.1
Content-Length:  0


  == SRTP unprotect failed on SSRC 1977138230 because of authentication failure 160
  == SRTCP unprotect failed on SSRC 1977138230 because of authentication failure
  == SRTCP unprotect failed on SSRC 748447341 because of unable to perform desired validation
<--- Received SIP request (576 bytes) from TLS:5.162.93.198:59422 --->
BYE sip:asterisk@5.37.215.237:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.178.154.185:44184;branch=z9hG4bK311932311;rport
From: <sip:141@5.162.93.198>;tag=184629041
To: <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
CSeq: 19367 BYE
Contact: <sip:141@10.178.154.185:44184;transport=tls>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (344 bytes) to TLS:5.162.93.198:59422 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.178.154.185:44184;rport=59422;received=5.162.93.198;branch=z9hG4bK311932311
Call-ID: 55d11340-fa84-49b6-b222-6c0ad8ffe5ef
From: <sip:141@5.162.93.198>;tag=184629041
To: <sip:Door_2@192.168.1.17>;tag=616a1a08-c84e-4802-b4e9-e1f56339139c
CSeq: 19367 BYE
Server: Asterisk PBX 18.5.1
Content-Length:  0


    -- Channel PJSIP/141-0000001b left 'simple_bridge' basic-bridge <42e1d601-c580-4034-9135-40d8de382c03>
    -- Channel PJSIP/161-00000018 left 'simple_bridge' basic-bridge <42e1d601-c580-4034-9135-40d8de382c03>
  == Spawn extension (default, 601, 2) exited non-zero on 'PJSIP/161-00000018'
<--- Transmitting SIP request (395 bytes) to UDP:192.168.1.161:5060 --->
BYE sip:161@192.168.1.161:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj3106ad7e-8ad9-46d8-aa6e-c5f034fc1a0e
From: <sip:601@192.168.1.17>;tag=7bcc20e0-e9d3-4561-8197-dd77a0c0ae01
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
Call-ID: 631126076
CSeq: 11728 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.5.1
Content-Length:  0


<--- Received SIP response (328 bytes) from UDP:192.168.1.161:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj3106ad7e-8ad9-46d8-aa6e-c5f034fc1a0e
From: <sip:601@192.168.1.17>;tag=7bcc20e0-e9d3-4561-8197-dd77a0c0ae01
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=2107391701
Call-ID: 631126076
CSeq: 11728 BYE
User-Agent: DnakeVoip v1.0
Content-Length: 0
```