SIP Debugging enabled
  == WebSocket connection from '192.168.0.105:58656' for protocol 'sip' accepted using version '13'

<--- SIP read from WS:192.168.0.105:58656 --->
REGISTER sip:192.168.0.106 SIP/2.0
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK3194103
Max-Forwards: 69
To: <sip:6001@192.168.0.106>
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=scovg0d3ri
Call-ID: 2budn7mpdc6vug2s3p0k52
CSeq: 1 REGISTER
Contact: <sip:6ombe7kc@md7d282amrbg.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:1f712b0d-01e3-4f0b-9651-64cf054ec8b3>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0

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

<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK3194103;received=192.168.0.105
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=scovg0d3ri
To: <sip:6001@192.168.0.106>;tag=as252e29bc
Call-ID: 2budn7mpdc6vug2s3p0k52
CSeq: 1 REGISTER
Server: Asterisk PBX 13.9.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="64120384"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2budn7mpdc6vug2s3p0k52' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:192.168.0.105:58656 --->
REGISTER sip:192.168.0.106 SIP/2.0
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK5714373
Max-Forwards: 69
To: <sip:6001@192.168.0.106>
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=scovg0d3ri
Call-ID: 2budn7mpdc6vug2s3p0k52
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="6001", realm="asterisk", nonce="64120384", uri="sip:192.168.0.106", response="c71552f115504226f6abb48e738c6edf"
Contact: <sip:6ombe7kc@md7d282amrbg.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:1f712b0d-01e3-4f0b-9651-64cf054ec8b3>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
    -- Registered SIP '6001' at 192.168.0.105:58656

<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK5714373;received=192.168.0.105
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=scovg0d3ri
To: <sip:6001@192.168.0.106>;tag=as252e29bc
Call-ID: 2budn7mpdc6vug2s3p0k52
CSeq: 2 REGISTER
Server: Asterisk PBX 13.9.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:6ombe7kc@md7d282amrbg.invalid;transport=ws>;expires=600
Date: Tue, 19 Jul 2016 10:59:47 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2budn7mpdc6vug2s3p0k52' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:192.168.0.105:58656 --->
INVITE sip:100@192.168.0.106 SIP/2.0
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK2245559
Max-Forwards: 69
To: <sip:100@192.168.0.106>
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=jjac6n4e7f
Call-ID: t4j8qsh032kml4frhk99
CSeq: 1824 INVITE
Contact: <sip:6ombe7kc@md7d282amrbg.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 2836

v=0
o=- 1107606867760237535 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS xLzvWZZD5EJhuzHSQgOObDaq80sFBDLsrlpa
m=audio 52819 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.56.1
a=rtcp:52823 IN IP4 192.168.56.1
a=candidate:2999745851 1 udp 2122260223 192.168.56.1 52819 typ host generation 0 network-id 4
a=candidate:2364966728 1 udp 2122194687 192.168.145.1 52820 typ host generation 0 network-id 3
a=candidate:931445828 1 udp 2122129151 192.168.152.1 52821 typ host generation 0 network-id 1
a=candidate:1221703924 1 udp 2122063615 192.168.0.105 52822 typ host generation 0 network-id 2
a=candidate:2999745851 2 udp 2122260222 192.168.56.1 52823 typ host generation 0 network-id 4
a=candidate:2364966728 2 udp 2122194686 192.168.145.1 52824 typ host generation 0 network-id 3
a=candidate:931445828 2 udp 2122129150 192.168.152.1 52825 typ host generation 0 network-id 1
a=candidate:1221703924 2 udp 2122063614 192.168.0.105 52826 typ host generation 0 network-id 2
a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 9 typ host tcptype active generation 0 network-id 4
a=candidate:3262479288 1 tcp 1518214911 192.168.145.1 9 typ host tcptype active generation 0 network-id 3
a=candidate:2030518452 1 tcp 1518149375 192.168.152.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:106054660 1 tcp 1518083839 192.168.0.105 9 typ host tcptype active generation 0 network-id 2
a=candidate:4233069003 2 tcp 1518280446 192.168.56.1 9 typ host tcptype active generation 0 network-id 4
a=candidate:3262479288 2 tcp 1518214910 192.168.145.1 9 typ host tcptype active generation 0 network-id 3
a=candidate:2030518452 2 tcp 1518149374 192.168.152.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:106054660 2 tcp 1518083838 192.168.0.105 9 typ host tcptype active generation 0 network-id 2
a=ice-ufrag:5qhvZRzwo/3PnbkN
a=ice-pwd:Ir7keeCyST562CrAuW5u1iF0
a=fingerprint:sha-256 AD:8B:1D:C2:1D:25:5A:42:E5:0F:94:99:AE:C4:26:F2:6A:68:1D:33:B1:67:F4:A6:AF:56:50:E0:2C:3C:26:89
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=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
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:959607047 cname:HJAv/8tOHPueGKe1
a=ssrc:959607047 msid:xLzvWZZD5EJhuzHSQgOObDaq80sFBDLsrlpa 862d0db4-25f2-4215-8346-41292752f8cb
a=ssrc:959607047 mslabel:xLzvWZZD5EJhuzHSQgOObDaq80sFBDLsrlpa
a=ssrc:959607047 label:862d0db4-25f2-4215-8346-41292752f8cb
<------------->
--- (14 headers 51 lines) ---
Using INVITE request as basis request - t4j8qsh032kml4frhk99
Found peer '6001' for '6001' from 192.168.0.105:58656
  == 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 - (gsm|ulaw|alaw), 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 192.168.56.1:52819
Looking for 100 in from-internal (domain 192.168.0.106)
sip_route_dump: route/path hop: <sip:6ombe7kc@md7d282amrbg.invalid;transport=ws;ob>

<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK2245559;received=192.168.0.105
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=jjac6n4e7f
To: <sip:100@192.168.0.106>
Call-ID: t4j8qsh032kml4frhk99
CSeq: 1824 INVITE
Server: Asterisk PBX 13.9.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:100@192.168.0.106:5060;transport=WS>
Content-Length: 0


<------------>
    -- Executing [100@from-internal:1] MeetMe("SIP/6001-00000001", "1234,M,0000") in new stack
Audio is at 15698
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK2245559;received=192.168.0.105
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=jjac6n4e7f
To: <sip:100@192.168.0.106>;tag=as3be602a5
Call-ID: t4j8qsh032kml4frhk99
CSeq: 1824 INVITE
Server: Asterisk PBX 13.9.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:100@192.168.0.106:5060;transport=WS>
Content-Type: application/sdp
Require: timer
Content-Length: 420

v=0
o=root 1094902875 1094902875 IN IP4 192.168.0.106
s=Asterisk PBX 13.9.1
c=IN IP4 192.168.0.106
t=0 0
m=audio 15698 RTP/SAVPF 0 8 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:150
a=connection:new
a=setup:active
a=fingerprint:SHA-256 AE:92:BF:E2:E6:A9:E5:A5:70:37:E6:59:50:99:10:AD:59:73:8D:57:33:CC:7B:BA:8D:FB:80:B8:C6:9A:36:EE
a=sendrecv

<------------>

<--- SIP read from WS:192.168.0.105:58656 --->
ACK sip:100@192.168.0.106:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK1626232
Max-Forwards: 69
To: <sip:100@192.168.0.106>;tag=as3be602a5
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=jjac6n4e7f
Call-ID: t4j8qsh032kml4frhk99
CSeq: 1824 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0

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

<--- SIP read from WS:192.168.0.105:58656 --->
BYE sip:100@192.168.0.106:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK2717876
Max-Forwards: 69
To: <sip:100@192.168.0.106>;tag=as3be602a5
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=jjac6n4e7f
Call-ID: t4j8qsh032kml4frhk99
CSeq: 1825 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Scheduling destruction of SIP dialog 't4j8qsh032kml4frhk99' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS md7d282amrbg.invalid;branch=z9hG4bK2717876;received=192.168.0.105
From: "User WebRTC 6001" <sip:6001@192.168.0.106>;tag=jjac6n4e7f
To: <sip:100@192.168.0.106>;tag=as3be602a5
Call-ID: t4j8qsh032kml4frhk99
CSeq: 1825 BYE
Server: Asterisk PBX 13.9.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '1234'
    -- <SIP/6001-00000001> Playing 'conf-onlyperson.gsm' (language 'en')
    -- Started music on hold, class 'default', on channel 'SIP/6001-00000001'
    -- Stopped music on hold on SIP/6001-00000001
    -- Hungup 'DAHDI/pseudo-802304624'
  == Spawn extension (from-internal, 100, 1) exited non-zero on 'SIP/6001-00000001'