[May 10 00:30:50] Asterisk 13.8.2 built by abuild @ build74 on a x86_64 running Linux on 2016-05-04 08:31:01 UTC
[May 10 00:31:36] VERBOSE[32037] chan_sip.c: Reliably Transmitting (NAT) to 176.36.35.126:5060:
OPTIONS sip:302@176.36.35.126 SIP/2.0
Via: SIP/2.0/UDP 104.207.131.136:1991;branch=z9hG4bK78ee9bed;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@104.207.131.136:1991>;tag=as5393c0c9
To: <sip:302@176.36.35.126>
Contact: <sip:asterisk@104.207.131.136:1991>
Call-ID: 3f8f2b67135bb50c397e52a8331f0396@104.207.131.136:1991
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.8.2
Date: Mon, 09 May 2016 21:31:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[May 10 00:31:36] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 104.207.131.136:1991;branch=z9hG4bK78ee9bed;rport
From: "asterisk" <sip:asterisk@104.207.131.136:1991>;tag=as5393c0c9
To: <sip:302@176.36.35.126>;tag=qb1GI
Call-ID: 3f8f2b67135bb50c397e52a8331f0396@104.207.131.136:1991
CSeq: 102 OPTIONS

<------------->
[May 10 00:31:36] VERBOSE[32037] chan_sip.c: --- (6 headers 0 lines) ---
[May 10 00:31:36] VERBOSE[32037] chan_sip.c: Really destroying SIP dialog '3f8f2b67135bb50c397e52a8331f0396@104.207.131.136:1991' Method: OPTIONS
[May 10 00:31:36] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->
INVITE sip:301@natalenko.name SIP/2.0
Via: SIP/2.0/UDP 172.17.29.2:5060;branch=z9hG4bK.HKKby37eQ;rport
From: <sip:302@natalenko.name>;tag=PELXtwe~v
To: sip:301@natalenko.name
CSeq: 20 INVITE
Call-ID: CEISEtydo8
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 548
Contact: <sip:302@176.36.35.126>;+sip.instance="<urn:uuid:18bd9f8c-354e-43d8-b6d3-be7bdab65d34>"
User-Agent: Linphone/3.9.1 (belle-sip/1.4.2)

v=0
o=302 977 1473 IN IP4 172.17.29.2
s=Talk
c=IN IP4 172.17.29.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
m=video 9078 RTP/AVP 96 97
a=rtpmap:96 VP8/90000
a=rtpmap:97 MP4V-ES/90000
a=fmtp:97 profile-level-id=3
<------------->
[May 10 00:31:36] VERBOSE[32037] chan_sip.c: --- (13 headers 20 lines) ---
[May 10 00:31:36] VERBOSE[32037] chan_sip.c: Sending to 176.36.35.126:5060 (NAT)
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Sending to 176.36.35.126:5060 (NAT)
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Using INVITE request as basis request - CEISEtydo8
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found peer '302' for '302' from 176.36.35.126:5060
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c:
<--- Reliably Transmitting (NAT) to 176.36.35.126:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.17.29.2:5060;branch=z9hG4bK.HKKby37eQ;received=176.36.35.126;rport=5060
From: <sip:302@natalenko.name>;tag=PELXtwe~v
To: sip:301@natalenko.name;tag=as2474b0a5
Call-ID: CEISEtydo8
CSeq: 20 INVITE
Server: Asterisk PBX 13.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11ac851e"
Content-Length: 0


<------------>
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog 'CEISEtydo8' in 6400 ms (Method: INVITE)
[May 10 00:31:36] SECURITY[32100] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2016-05-10T00:31:36.831+0300",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:302@natalenko.name",SessionID="0x7f07e400a8e8",LocalAddress="IPV4/UDP/104.207.131.136/1991",RemoteAddress="IPV4/UDP/176.36.35.126/5060",Challenge="11ac851e"
[May 10 00:31:36] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->
ACK sip:301@natalenko.name SIP/2.0
Via: SIP/2.0/UDP 172.17.29.2:5060;branch=z9hG4bK.HKKby37eQ;rport
Call-ID: CEISEtydo8
From: <sip:302@natalenko.name>;tag=PELXtwe~v
To: <sip:301@natalenko.name>;tag=as2474b0a5
Contact: <sip:302@176.36.35.126>;+sip.instance="<urn:uuid:18bd9f8c-354e-43d8-b6d3-be7bdab65d34>"
Max-Forwards: 70
CSeq: 20 ACK

<------------->
[May 10 00:31:36] VERBOSE[32037] chan_sip.c: --- (8 headers 0 lines) ---
[May 10 00:31:36] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->
INVITE sip:301@natalenko.name SIP/2.0
Via: SIP/2.0/UDP 172.17.29.2:5060;branch=z9hG4bK.C8TEneddp;rport
From: <sip:302@natalenko.name>;tag=PELXtwe~v
To: sip:301@natalenko.name
CSeq: 21 INVITE
Call-ID: CEISEtydo8
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 548
Contact: <sip:302@176.36.35.126>;+sip.instance="<urn:uuid:18bd9f8c-354e-43d8-b6d3-be7bdab65d34>"
User-Agent: Linphone/3.9.1 (belle-sip/1.4.2)
Authorization: Digest realm="asterisk", nonce="11ac851e", algorithm=MD5, username="302", uri="sip:301@natalenko.name", response="9136179030a3aeb377e8cf1e11f1a2bf"

v=0
o=302 977 1473 IN IP4 172.17.29.2
s=Talk
c=IN IP4 172.17.29.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
m=video 9078 RTP/AVP 96 97
a=rtpmap:96 VP8/90000
a=rtpmap:97 MP4V-ES/90000
a=fmtp:97 profile-level-id=3
<------------->
[May 10 00:31:36] VERBOSE[32037] chan_sip.c: --- (14 headers 20 lines) ---
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Sending to 176.36.35.126:5060 (NAT)
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Using INVITE request as basis request - CEISEtydo8
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found peer '302' for '302' from 176.36.35.126:5060
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP audio format 96
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP audio format 97
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP audio format 98
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP audio format 0
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP audio format 8
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP audio format 101
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP audio format 99
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP audio format 100
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found audio description format opus for ID 96
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found audio description format speex for ID 97
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found audio description format speex for ID 98
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found unknown media description format telephone-event for ID 101
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found unknown media description format telephone-event for ID 99
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 100
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP video format 96
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found RTP video format 97
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found video description format VP8 for ID 96
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Found video description format MP4V-ES for ID 97
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Capabilities: us - (g729|alaw), peer - audio=(ulaw|alaw|opus|speex16|speex)/video=(vp8|mpeg4)/text=(nothing), combined - (alaw)
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Peer audio RTP is at port 172.17.29.2:7078
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c: Looking for 301 in phones (domain natalenko.name)
[May 10 00:31:36] VERBOSE[32037][C-00000001] sip/route.c: sip_route_dump: route/path hop: <sip:302@176.36.35.126>
[May 10 00:31:36] VERBOSE[32037][C-00000001] chan_sip.c:
<--- Transmitting (NAT) to 176.36.35.126:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.17.29.2:5060;branch=z9hG4bK.C8TEneddp;received=176.36.35.126;rport=5060
From: <sip:302@natalenko.name>;tag=PELXtwe~v
To: sip:301@natalenko.name
Call-ID: CEISEtydo8
CSeq: 21 INVITE
Server: Asterisk PBX 13.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:301@104.207.131.136:1991>
Content-Length: 0


<------------>
[May 10 00:31:36] SECURITY[32100] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2016-05-10T00:31:36.869+0300",Severity="Informational",Service="SIP",EventVersion="1",AccountID="301",SessionID="0x7f07e400a8e8",LocalAddress="IPV4/UDP/104.207.131.136/1991",RemoteAddress="IPV4/UDP/176.36.35.126/5060",UsingPassword="1"
[May 10 00:31:36] CC[32634][C-00000001] ccss.c: Agent policy for SIP/302-00000002 is 'never'. CC not possible
[May 10 00:31:36] VERBOSE[32634][C-00000001] chan_sip.c: Audio is at 63052
[May 10 00:31:36] VERBOSE[32634][C-00000001] chan_sip.c: Adding codec alaw to SDP
[May 10 00:31:36] VERBOSE[32634][C-00000001] chan_sip.c: Adding codec g729 to SDP
[May 10 00:31:36] VERBOSE[32634][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[May 10 00:31:36] VERBOSE[32634][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 176.36.35.126:55074:
INVITE sips:301@176.36.35.126:55074;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS 104.207.131.136:1992;branch=z9hG4bK0b82df89
Max-Forwards: 70
From: <sip:302@104.207.131.136:1992>;tag=as7310449b
To: <sips:301@176.36.35.126:55074;transport=TLS;ob>
Contact: <sip:302@104.207.131.136:1992;transport=TLS>
Call-ID: 1d687e980c4822f80907dc254eb3da44@104.207.131.136:1992
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.8.2
Date: Mon, 09 May 2016 21:31:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 389

v=0
o=root 2084712478 2084712478 IN IP4 104.207.131.136
s=Asterisk PBX 13.8.2
c=IN IP4 104.207.131.136
t=0 0
m=audio 63052 RTP/SAVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XZIiXcDOU39wzN8MhOYZYXlkk24rA1qWJyl0eIms

---
[May 10 00:31:37] VERBOSE[32344] chan_sip.c:
<--- SIP read from TLS:176.36.35.126:55074 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TLS 104.207.131.136:1992;received=104.207.131.136;branch=z9hG4bK0b82df89
Call-ID: 1d687e980c4822f80907dc254eb3da44@104.207.131.136:1992
From: <sip:302@104.207.131.136>;tag=as7310449b
To: <sips:301@176.36.35.126;ob>;tag=z9hG4bK0b82df89
CSeq: 102 INVITE
Warning: 381 localhost "SIPS Required"
Content-Length: 0

<------------->
[May 10 00:31:37] VERBOSE[32344] chan_sip.c: --- (8 headers 0 lines) ---
[May 10 00:31:37] VERBOSE[32344][C-00000001] chan_sip.c: Transmitting (no NAT) to 176.36.35.126:55074:
ACK sips:301@176.36.35.126:55074;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS 104.207.131.136:1992;branch=z9hG4bK0b82df89
Max-Forwards: 70
From: <sip:302@104.207.131.136:1992>;tag=as7310449b
To: <sips:301@176.36.35.126:55074;transport=TLS;ob>;tag=z9hG4bK0b82df89
Contact: <sip:302@104.207.131.136:1992;transport=TLS>
Call-ID: 1d687e980c4822f80907dc254eb3da44@104.207.131.136:1992
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.8.2
Content-Length: 0


---
[May 10 00:31:37] VERBOSE[32634][C-00000001] chan_sip.c:
<--- Transmitting (NAT) to 176.36.35.126:5060 --->
SIP/2.0 181 Call is being forwarded
Via: SIP/2.0/UDP 172.17.29.2:5060;branch=z9hG4bK.C8TEneddp;received=176.36.35.126;rport=5060
From: <sip:302@natalenko.name>;tag=PELXtwe~v
To: sip:301@natalenko.name;tag=as1c271a84
Call-ID: CEISEtydo8
CSeq: 21 INVITE
Server: Asterisk PBX 13.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:301@104.207.131.136:1991>
Content-Length: 0


<------------>
[May 10 00:31:37] VERBOSE[32634][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '1d687e980c4822f80907dc254eb3da44@104.207.131.136:1992' in 11584 ms (Method: INVITE)
[May 10 00:31:37] VERBOSE[32634][C-00000001] chan_sip.c:
<--- Reliably Transmitting (NAT) to 176.36.35.126:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 172.17.29.2:5060;branch=z9hG4bK.C8TEneddp;received=176.36.35.126;rport=5060
From: <sip:302@natalenko.name>;tag=PELXtwe~v
To: sip:301@natalenko.name;tag=as1c271a84
Call-ID: CEISEtydo8
CSeq: 21 INVITE
Server: Asterisk PBX 13.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


<------------>
[May 10 00:31:37] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->
ACK sip:301@natalenko.name SIP/2.0
Via: SIP/2.0/UDP 172.17.29.2:5060;branch=z9hG4bK.C8TEneddp;rport
Call-ID: CEISEtydo8
From: <sip:302@natalenko.name>;tag=PELXtwe~v
To: <sip:301@natalenko.name>;tag=as1c271a84
Contact: <sip:302@176.36.35.126>;+sip.instance="<urn:uuid:18bd9f8c-354e-43d8-b6d3-be7bdab65d34>"
Max-Forwards: 70
CSeq: 21 ACK

<------------->
[May 10 00:31:37] VERBOSE[32037] chan_sip.c: --- (8 headers 0 lines) ---
[May 10 00:31:37] VERBOSE[32037] chan_sip.c: Really destroying SIP dialog 'CEISEtydo8' Method: ACK
[May 10 00:31:42] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


<------------->
[May 10 00:31:45] VERBOSE[32037] chan_sip.c: Reliably Transmitting (no NAT) to 176.36.35.126:55074:
OPTIONS sips:301@176.36.35.126:55074;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS 104.207.131.136:1992;branch=z9hG4bK3a5fd236
Max-Forwards: 70
From: "asterisk" <sip:asterisk@104.207.131.136:1992>;tag=as65e4e795
To: <sips:301@176.36.35.126:55074;transport=TLS;ob>
Contact: <sip:asterisk@104.207.131.136:1992;transport=TLS>
Call-ID: 279565283f51180226335e017b4531bb@104.207.131.136:1992
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.8.2
Date: Mon, 09 May 2016 21:31:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[May 10 00:31:46] VERBOSE[32344] chan_sip.c:
<--- SIP read from TLS:176.36.35.126:55074 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 104.207.131.136:1992;received=104.207.131.136;branch=z9hG4bK3a5fd236
Call-ID: 279565283f51180226335e017b4531bb@104.207.131.136:1992
From: "asterisk" <sip:asterisk@104.207.131.136>;tag=as65e4e795
To: <sips:301@176.36.35.126;ob>;tag=z9hG4bK3a5fd236
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_P1ma40-22/r2459
Content-Type: application/sdp
Content-Length: 401

v=0
o=- 3671818309 3671818309 IN IP4 172.17.28.5
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 8 18 3 102 0 9 106 101
c=IN IP4 172.17.28.5
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 ILBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:106 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[May 10 00:31:46] VERBOSE[32344] chan_sip.c: --- (13 headers 18 lines) ---
[May 10 00:31:47] VERBOSE[32037] chan_sip.c: Really destroying SIP dialog '279565283f51180226335e017b4531bb@104.207.131.136:1992' Method: OPTIONS
[May 10 00:31:48] VERBOSE[32037] chan_sip.c: Really destroying SIP dialog '1d687e980c4822f80907dc254eb3da44@104.207.131.136:1992' Method: INVITE
[May 10 00:31:52] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


<------------->
[May 10 00:32:02] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


<------------->
[May 10 00:32:12] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


<------------->
[May 10 00:32:22] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


<------------->
[May 10 00:32:32] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


<------------->
[May 10 00:32:36] VERBOSE[32037] chan_sip.c: Reliably Transmitting (NAT) to 176.36.35.126:5060:
OPTIONS sip:302@176.36.35.126 SIP/2.0
Via: SIP/2.0/UDP 104.207.131.136:1991;branch=z9hG4bK370be4f4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@104.207.131.136:1991>;tag=as6a6cfe30
To: <sip:302@176.36.35.126>
Contact: <sip:asterisk@104.207.131.136:1991>
Call-ID: 02d06fef599d2bf57710eb6a6337e383@104.207.131.136:1991
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.8.2
Date: Mon, 09 May 2016 21:32:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[May 10 00:32:36] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 104.207.131.136:1991;branch=z9hG4bK370be4f4;rport
From: "asterisk" <sip:asterisk@104.207.131.136:1991>;tag=as6a6cfe30
To: <sip:302@176.36.35.126>;tag=P1nNN
Call-ID: 02d06fef599d2bf57710eb6a6337e383@104.207.131.136:1991
CSeq: 102 OPTIONS

<------------->
[May 10 00:32:36] VERBOSE[32037] chan_sip.c: --- (6 headers 0 lines) ---
[May 10 00:32:36] VERBOSE[32037] chan_sip.c: Really destroying SIP dialog '02d06fef599d2bf57710eb6a6337e383@104.207.131.136:1991' Method: OPTIONS
[May 10 00:32:42] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


<------------->
[May 10 00:32:46] VERBOSE[32037] chan_sip.c: Reliably Transmitting (no NAT) to 176.36.35.126:55074:
OPTIONS sips:301@176.36.35.126:55074;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS 104.207.131.136:1992;branch=z9hG4bK662259d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@104.207.131.136:1992>;tag=as1e689fdc
To: <sips:301@176.36.35.126:55074;transport=TLS;ob>
Contact: <sip:asterisk@104.207.131.136:1992;transport=TLS>
Call-ID: 050c9c755e26742b0b89f9742de96e19@104.207.131.136:1992
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.8.2
Date: Mon, 09 May 2016 21:32:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[May 10 00:32:50] VERBOSE[32344] chan_sip.c:
<--- SIP read from TLS:176.36.35.126:55074 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 104.207.131.136:1992;received=104.207.131.136;branch=z9hG4bK662259d3
Call-ID: 050c9c755e26742b0b89f9742de96e19@104.207.131.136:1992
From: "asterisk" <sip:asterisk@104.207.131.136>;tag=as1e689fdc
To: <sips:301@176.36.35.126;ob>;tag=z9hG4bK662259d3
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_P1ma40-22/r2459
Content-Type: application/sdp
Content-Length: 401

v=0
o=- 3671818373 3671818373 IN IP4 172.17.28.5
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 8 18 3 102 0 9 106 101
c=IN IP4 172.17.28.5
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 ILBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:106 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[May 10 00:32:50] VERBOSE[32344] chan_sip.c: --- (13 headers 18 lines) ---
[May 10 00:32:50] NOTICE[32344] chan_sip.c: Peer '301' is now Lagged. (3799ms / 2000ms)
[May 10 00:32:50] VERBOSE[32037] chan_sip.c: Really destroying SIP dialog '050c9c755e26742b0b89f9742de96e19@104.207.131.136:1992' Method: OPTIONS
[May 10 00:32:52] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


<------------->
[May 10 00:33:00] VERBOSE[32037] chan_sip.c: Reliably Transmitting (no NAT) to 176.36.35.126:55074:
OPTIONS sips:301@176.36.35.126:55074;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS 104.207.131.136:1992;branch=z9hG4bK5ee618f9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@104.207.131.136:1992>;tag=as1452a61b
To: <sips:301@176.36.35.126:55074;transport=TLS;ob>
Contact: <sip:asterisk@104.207.131.136:1992;transport=TLS>
Call-ID: 33f6bcd12f08282b5d1f01d43780e39e@104.207.131.136:1992
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.8.2
Date: Mon, 09 May 2016 21:33:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[May 10 00:33:00] VERBOSE[32344] chan_sip.c:
<--- SIP read from TLS:176.36.35.126:55074 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 104.207.131.136:1992;received=104.207.131.136;branch=z9hG4bK5ee618f9
Call-ID: 33f6bcd12f08282b5d1f01d43780e39e@104.207.131.136:1992
From: "asterisk" <sip:asterisk@104.207.131.136>;tag=as1452a61b
To: <sips:301@176.36.35.126;ob>;tag=z9hG4bK5ee618f9
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_P1ma40-22/r2459
Content-Type: application/sdp
Content-Length: 401

v=0
o=- 3671818383 3671818383 IN IP4 172.17.28.5
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 8 18 3 102 0 9 106 101
c=IN IP4 172.17.28.5
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 ILBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:106 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[May 10 00:33:00] VERBOSE[32344] chan_sip.c: --- (13 headers 18 lines) ---
[May 10 00:33:00] NOTICE[32344] chan_sip.c: Peer '301' is now Reachable. (218ms / 2000ms)
[May 10 00:33:01] VERBOSE[32037] chan_sip.c: Really destroying SIP dialog '33f6bcd12f08282b5d1f01d43780e39e@104.207.131.136:1992' Method: OPTIONS
[May 10 00:33:02] VERBOSE[32037] chan_sip.c:
<--- SIP read from UDP:176.36.35.126:5060 --->


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