Working SIP

SIP Debugging enabled
Really destroying SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' Method: REGISTER

<--- SIP read from WS:192.168.88.174:49534 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOilEKV9f2C1Am9nd87SuyWsOZ3pBPUyf;rport
From: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
To: <sip:[email protected]>
Contact: "888"<sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 50788 INVITE
Content-Type: application/sdp
Content-Length: 1591
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 6673466038043864000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz
m=audio 62984 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.88.174
a=rtcp:62984 IN IP4 192.168.88.174
a=candidate:159100432 1 udp 2122194687 192.168.88.174 62984 typ host generation 0
a=candidate:159100432 2 udp 2122194687 192.168.88.174 62984 typ host generation 0
a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
a=ice-ufrag:508JZ2hjbUjZ3Hv3
a=ice-pwd:pX+OETY08O8y4tswCjrFROTO
a=ice-options:google-ice
a=fingerprint:sha-256 CE:1E:A7:8A:C9:F9:0D:CF:FB:54:C5:97:D0:9D:BE:F8:26:D8:DD:D8:F3:46:70:C1:B8:DB:DC:31:04:EA:A6:08
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:4287587852 cname:F+ii1f0GVr8omEdf
a=ssrc:4287587852 msid:1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz 10bec123-e65f-45ed-a025-120b4b7859d4
a=ssrc:4287587852 mslabel:1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz
a=ssrc:4287587852 label:10bec123-e65f-45ed-a025-120b4b7859d4
<------------->
--- (13 headers 39 lines) ---
Using INVITE request as basis request - 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
Found peer '888' for '888' from 192.168.88.174:49534

<--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOilEKV9f2C1Am9nd87SuyWsOZ3pBPUyf;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
To: <sip:[email protected]>;tag=as45cee7bb
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 50788 INVITE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="3797a877"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.88.174:49534 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOilEKV9f2C1Am9nd87SuyWsOZ3pBPUyf;rport
From: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
To: <sip:[email protected]>;tag=as45cee7bb
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 50788 ACK
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from WS:192.168.88.174:49534 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKThoFZfofque4cJIKAWzA5C8egG8cZUre;rport
From: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
To: <sip:[email protected]>
Contact: "888"<sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 50789 INVITE
Content-Type: application/sdp
Content-Length: 1591
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="888",realm="192.168.88.251",nonce="3797a877",uri="sip:[email protected]",response="a7c984770475702277553f1cf5ffcda2",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 6673466038043864000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz
m=audio 62984 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.88.174
a=rtcp:62984 IN IP4 192.168.88.174
a=candidate:159100432 1 udp 2122194687 192.168.88.174 62984 typ host generation 0
a=candidate:159100432 2 udp 2122194687 192.168.88.174 62984 typ host generation 0
a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
a=ice-ufrag:508JZ2hjbUjZ3Hv3
a=ice-pwd:pX+OETY08O8y4tswCjrFROTO
a=ice-options:google-ice
a=fingerprint:sha-256 CE:1E:A7:8A:C9:F9:0D:CF:FB:54:C5:97:D0:9D:BE:F8:26:D8:DD:D8:F3:46:70:C1:B8:DB:DC:31:04:EA:A6:08
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:4287587852 cname:F+ii1f0GVr8omEdf
a=ssrc:4287587852 msid:1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz 10bec123-e65f-45ed-a025-120b4b7859d4
a=ssrc:4287587852 mslabel:1vPRicpzXeePJRX5euyzv288i7F0HdMS5vbz
a=ssrc:4287587852 label:10bec123-e65f-45ed-a025-120b4b7859d4
<------------->
--- (14 headers 39 lines) ---
Using INVITE request as basis request - 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
Found peer '888' for '888' from 192.168.88.174:49534
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.88.174:62984
Looking for 889 in default (domain 192.168.88.251)
sip_route_dump: route/path hop: <sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>

<--- Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKThoFZfofque4cJIKAWzA5C8egG8cZUre;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
To: <sip:[email protected]>
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 50789 INVITE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060;transport=WS>
Content-Length: 0


<------------>
    -- Executing [889@default:1] Dial("SIP/888-00000057", "SIP/889") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 19510
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.88.187:49625:
INVITE sip:[email protected];rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK20e20b21
Max-Forwards: 70
From: "888" <sip:[email protected]>;tag=as31b619ed
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.2.0
Date: Fri, 13 Feb 2015 04:08:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 674

v=0
o=root 258976009 258976009 IN IP4 192.168.88.251
s=Asterisk PBX 13.2.0
c=IN IP4 192.168.88.251
t=0 0
m=audio 19510 RTP/SAVPF 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=ice-ufrag:55783ab604a96c221e1f9fca4b526dd4
a=ice-pwd:68e6006a5653d426317ee5fe2172d4c9
a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 19510 typ host
a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 19511 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
a=sendrecv

---
    -- Called SIP/889

<--- SIP read from WS:192.168.88.187:49625 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK20e20b21
From: "888"<sip:[email protected]>;tag=as31b619ed
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from WS:192.168.88.187:49625 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK20e20b21
From: "888"<sip:[email protected]>;tag=as31b619ed
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;tag=gwK0qRuZMIfigmgTWstQ
Contact: <sip:[email protected];transport=ws>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:[email protected];transport=ws>
    -- SIP/889-00000058 is ringing

<--- Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKThoFZfofque4cJIKAWzA5C8egG8cZUre;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
To: <sip:[email protected]>;tag=as0c09af77
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 50789 INVITE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060;transport=WS>
Content-Length: 0


<------------>
Really destroying SIP dialog '01e934ff-5052-fc68-6a3c-0bc36705ff46' Method: REGISTER

<--- SIP read from WS:192.168.88.187:49625 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK20e20b21
From: "888"<sip:[email protected]>;tag=as31b619ed
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;tag=gwK0qRuZMIfigmgTWstQ
Contact: <sip:[email protected];transport=ws>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 1167
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

v=0
o=- 5364208010358385000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=msid-semantic: WMS r74fGcJTtOg3IHU38SQeQ5cU3fKb75YAjI7G
m=audio 50364 UDP/TLS/RTP/SAVPF 0 8 101
c=IN IP4 192.168.88.187
a=rtcp:50365 IN IP4 192.168.88.187
a=candidate:2577307183 1 udp 2122194687 192.168.88.187 50364 typ host generation 0
a=candidate:2577307183 2 udp 2122194686 192.168.88.187 50365 typ host generation 0
a=candidate:3609029343 1 tcp 1518214911 192.168.88.187 0 typ host tcptype active generation 0
a=candidate:3609029343 2 tcp 1518214910 192.168.88.187 0 typ host tcptype active generation 0
a=ice-ufrag:GefBqC5O5qnV6nwt
a=ice-pwd:QzoXPid0PCMwYEtzJ2rZpbHJ
a=fingerprint:sha-256 D8:C4:BF:59:B9:A8:19:A0:4C:31:BA:92:F0:62:A0:3E:27:D4:90:9B:79:33:E3:B6:FC:E9:2A:EB:C3:D3:DF:E6
a=setup:active
a=mid:audio
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:99136968 cname:Cr5VtuG5xK5gmOI0
a=ssrc:99136968 msid:r74fGcJTtOg3IHU38SQeQ5cU3fKb75YAjI7G 0376de4e-6e18-4ca8-b7e7-f479bbd58814
a=ssrc:99136968 mslabel:r74fGcJTtOg3IHU38SQeQ5cU3fKb75YAjI7G
a=ssrc:99136968 label:0376de4e-6e18-4ca8-b7e7-f479bbd58814
<------------->
--- (10 headers 25 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.88.187:50364
sip_route_dump: route/path hop: <sip:[email protected];transport=ws>
[Feb 13 06:08:10] ERROR[1055][C-0000002d]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
[Feb 13 06:08:10] WARNING[1055][C-0000002d]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
set_destination: Parsing <sip:[email protected];transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (no NAT) to 192.168.88.187:49625:
ACK sip:[email protected];transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK00e39beb
Max-Forwards: 70
From: "888" <sip:[email protected]>;tag=as31b619ed
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;tag=gwK0qRuZMIfigmgTWstQ
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.2.0
Content-Length: 0


---
    -- SIP/889-00000058 answered SIP/888-00000057
Audio is at 18654
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKThoFZfofque4cJIKAWzA5C8egG8cZUre;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
To: <sip:[email protected]>;tag=as0c09af77
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 50789 INVITE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 673

v=0
o=root 645993891 645993891 IN IP4 192.168.88.251
s=Asterisk PBX 13.2.0
c=IN IP4 192.168.88.251
t=0 0
m=audio 18654 RTP/SAVPF 0 8 3 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:150
a=ice-ufrag:66a8fa20406da1b05cc9891248f19665
a=ice-pwd:4f5fa9126a825b6f38f1b4d515068a62
a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 18654 typ host
a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 18655 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
a=sendrecv

<------------>
    -- Channel SIP/888-00000057 joined 'simple_bridge' basic-bridge <09ed0589-0211-43ea-afbe-0580e91ecfc5>
    -- Channel SIP/889-00000058 joined 'simple_bridge' basic-bridge <09ed0589-0211-43ea-afbe-0580e91ecfc5>

<--- SIP read from WS:192.168.88.174:49534 --->
ACK sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKxlSp7r6AGZ9XNFnEio78;rport
From: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
To: <sip:[email protected]>;tag=as0c09af77
Contact: "888"<sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 50789 ACK
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="888",realm="192.168.88.251",nonce="3797a877",uri="sip:[email protected]:5060;transport=WS",response="9ffc84a96b9cdabc77990888b9fa8ab6",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
--- (13 headers 0 lines) ---
       > 0x7fd8389ed7e0 -- Probation passed - setting RTP source address to 192.168.88.187:50364
       > 0x7fd8388a9ba0 -- Probation passed - setting RTP source address to 192.168.88.174:62984

<--- SIP read from WS:192.168.88.187:49625 --->
BYE sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKNMvLQe0eTKLVlTNvykfChDzK6nPugf9n;rport
From: <sip:[email protected]>;tag=gwK0qRuZMIfigmgTWstQ
To: "888"<sip:[email protected]>;tag=as31b619ed
Call-ID: [email protected]:5060
CSeq: 58758 BYE
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
--- (15 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.88.187:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKNMvLQe0eTKLVlTNvykfChDzK6nPugf9n;rport;received=192.168.88.187
From: <sip:[email protected]>;tag=gwK0qRuZMIfigmgTWstQ
To: "888"<sip:[email protected]>;tag=as31b619ed
Call-ID: [email protected]:5060
CSeq: 58758 BYE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/889-00000058 left 'simple_bridge' basic-bridge <09ed0589-0211-43ea-afbe-0580e91ecfc5>
    -- Channel SIP/888-00000057 left 'simple_bridge' basic-bridge <09ed0589-0211-43ea-afbe-0580e91ecfc5>
  == Spawn extension (default, 889, 1) exited non-zero on 'SIP/888-00000057'
Scheduling destruction of SIP dialog '5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Reliably Transmitting (no NAT) to 192.168.88.174:5060:
BYE sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK5e8b871a
Max-Forwards: 70
From: <sip:[email protected]>;tag=as0c09af77
To: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.2.0
Proxy-Authorization: Digest username="888", realm="192.168.88.251", algorithm=MD5, uri="sip:192.168.88.251", nonce="3797a877", response="a04d3572e794d02d67f953d6bea7311d"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from WS:192.168.88.174:49534 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK5e8b871a
From: <sip:[email protected]>;tag=as0c09af77
To: "888"<sip:[email protected]>;tag=Jl1s49meNfTyNH6uJ7AI
Contact: <sip:[email protected];transport=ws>
Call-ID: 5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc
CSeq: 102 BYE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '5a0acc18-0dd2-b4c6-5e07-f38c12c6f5bc' Method: INVITE