Asterisk Logs Success 6-25-19

  == WebSocket connection from '172.17.0.3:60464' for protocol 'sip' accepted using version '13'

<--- SIP read from WS:172.17.0.3:60464 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS 192.0.2.132;branch=z9hG4bK7788412
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=lekdip34io
Call-ID: cv8srkuft2fkhffgcpmj
CSeq: 8542 INVITE
Contact: <sip:[email protected];transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Using INVITE request as basis request - cv8srkuft2fkhffgcpmj
No matching peer for '41185582633' from '172.17.0.3:60464'
  == Using SIP RTP CoS mark 5
Looking for 77819069506 in default (domain 192.168.201.196)
sip_route_dump: route/path hop: <sip:[email protected];transport=ws;ob>

<--- Transmitting (NAT) to 172.17.0.3:60464 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.0.2.132;branch=z9hG4bK7788412;received=172.17.0.3;rport=60464
From: <sip:[email protected]>;tag=lekdip34io
To: <sip:[email protected]>
Call-ID: cv8srkuft2fkhffgcpmj
CSeq: 8542 INVITE
Server: Asterisk PBX 15.5.0
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 [77819069506@default:1] Answer("SIP/192.168.201.196-000000b5", "") in new stack
Audio is at 39730
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP

<--- Reliably Transmitting (NAT) to 172.17.0.3:60464 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.132;branch=z9hG4bK7788412;received=172.17.0.3;rport=60464
From: <sip:[email protected]>;tag=lekdip34io
To: <sip:[email protected]>;tag=as6d1bd8ce
Call-ID: cv8srkuft2fkhffgcpmj
CSeq: 8542 INVITE
Server: Asterisk PBX 15.5.0
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: 742

v=0
o=root 1518115017 1518115017 IN IP4 127.0.0.1
s=Asterisk PBX 15.5.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 39730 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=maxptime:150
a=ice-ufrag:5a3b4251268d578642ba783f564616a3
a=ice-pwd:09d2548823d1ddce2ff486742e031813
a=candidate:Hc0a8c9c4 1 UDP 2130706431 192.168.201.196 39730 typ host
a=candidate:Hc0a8c9db 1 UDP 2130706431 192.168.201.219 39730 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 39730 typ host
a=candidate:Hc0a8c9c4 2 UDP 2130706430 192.168.201.196 39731 typ host
a=candidate:Hc0a8c9db 2 UDP 2130706430 192.168.201.219 39731 typ host
a=candidate:Hac110001 2 UDP 2130706430 172.17.0.1 39731 typ host
a=rtcp-mux
a=sendrecv

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

<--- SIP read from WS:172.17.0.3:60464 --->
ACK sip:[email protected]:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.0.2.132;branch=z9hG4bK3229607
Max-Forwards: 70
To: <sip:[email protected]>;tag=as6d1bd8ce
From: <sip:[email protected]>;tag=lekdip34io
Call-ID: cv8srkuft2fkhffgcpmj
CSeq: 8542 ACK
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Executing [77819069506@default:2] NoOp("SIP/192.168.201.196-000000b5", "77819069506") in new stack
    -- Executing [77819069506@default:3] Read("SIP/192.168.201.196-000000b5", "DT,,1,,,10000") in new stack
    -- Accepting a maximum of 1 digits.

<--- SIP read from WS:172.17.0.3:60464 --->
INFO sip:[email protected]:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.0.2.132;branch=z9hG4bK4641878
Max-Forwards: 70
To: <sip:[email protected]>;tag=as6d1bd8ce
From: <sip:[email protected]>;tag=lekdip34io
Call-ID: cv8srkuft2fkhffgcpmj
CSeq: 8543 INFO
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Type: application/dtmf-relay
Content-Length: 23

Signal= 1
Duration= 70
<------------->
--- (11 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 1

<--- Transmitting (NAT) to 172.17.0.3:60464 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.132;branch=z9hG4bK4641878;received=172.17.0.3;rport=60464
From: <sip:[email protected]>;tag=lekdip34io
To: <sip:[email protected]>;tag=as6d1bd8ce
Call-ID: cv8srkuft2fkhffgcpmj
CSeq: 8543 INFO
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- User entered '1'
    -- Executing [77819069506@default:4] Set("SIP/192.168.201.196-000000b5", "CHANNEL(language)=en") in new stack
    -- Executing [77819069506@default:5] ConfBridge("SIP/192.168.201.196-000000b5", "77819069506") in new stack
    -- Channel CBAnn/77819069506-00000040;2 joined 'softmix' base-bridge <41acad88-832f-42f5-b61d-0d96fe844db9>
    -- <SIP/192.168.201.196-000000b5> Playing 'conf-onlyperson.slin' (language 'en')
    -- Started music on hold, class 'default', on channel 'SIP/192.168.201.196-000000b5'
    -- Channel SIP/192.168.201.196-000000b5 joined 'softmix' base-bridge <41acad88-832f-42f5-b61d-0d96fe844db9>
    -- <CBAnn/77819069506-00000040;1> Playing 'confbridge-join.slin' (language 'en')