asterisk sip log

root@voip:~# asterisk -r
Asterisk 13.6.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.6.0 currently running on voip (pid = 32018)
voip*CLI> sip set debug on
SIP Debugging enabled
Retransmitting #6 (NAT) to 155.94.64.34:5089:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 155.94.64.34:5089;branch=z9hG4bK-0575c362977ea6340ca6fcbe6260b446;received=155.94.64.34;rport=5089
From: 101<sip:101@93.89.146.5>;tag=8a359551
To: 991130972597723173<sip:991130972597723173@93.89.146.5>;tag=as1e4415a6
Call-ID: 0575c362977ea6340ca6fcbe6260b446
CSeq: 1 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54467743"
Content-Length: 0


---

<--- SIP read from UDP:89.190.50.140:39059 --->
INVITE sip:101@eu.vancl.eu SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:39059;rport;branch=z9hG4bK1595020550
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>
Call-ID: 1937419984
CSeq: 20 INVITE
Contact: <sip:100@192.168.1.112:39059>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Max-Forwards: 70
User-Agent: antisip/5.1.0-549-g2028176-Oct-15-2015 amdroid/4.2.5 HM 1SW/4.4.4
Subject: Talk
Supported: 100rel, replaces
Content-Length: 744

v=0
o=amsip 1989349040 0 IN IP4 192.168.1.112
s=talk
c=IN IP4 192.168.1.112
t=0 0
m=audio 10000 RTP/AVP 98 105 100 97 99 103 0 8 101
a=rtpmap:98 SILK/16000
a=rtpmap:105 OPUS/48000/2
a=fmtp:105 useinbandfec=1
a=rtpmap:100 speex/16000
a=fmtp:100 mode="6,any"
a=rtpmap:97 SILK/8000
a=rtpmap:99 speex/8000
a=fmtp:99 mode="6,any"
a=rtpmap:103 iLBC/8000
a=fmtp:103 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp-mux
m=video 10020 RTP/AVP 118 117 116 115 34
b=AS:128
a=rtpmap:118 VP8/90000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42800c; packetization-mode=1
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000
a=rtpmap:34 H263/90000
a=rtcp-mux
<------------->
--- (14 headers 30 lines) ---
Sending to 89.190.50.140:39059 (NAT)
Sending to 89.190.50.140:39059 (NAT)
Using INVITE request as basis request - 1937419984
Found peer '100' for '100' from 89.190.50.140:39059

<--- Reliably Transmitting (NAT) to 89.190.50.140:39059 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK1595020550;received=89.190.50.140;rport=39059
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>;tag=as1d1f47bb
Call-ID: 1937419984
CSeq: 20 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="34225d72"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1937419984' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:89.190.50.140:39059 --->
ACK sip:101@eu.vancl.eu SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:39059;rport;branch=z9hG4bK1595020550
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>;tag=as1d1f47bb
Call-ID: 1937419984
CSeq: 20 ACK
Content-Length: 0

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

<--- SIP read from UDP:89.190.50.140:39059 --->
INVITE sip:101@eu.vancl.eu SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:39059;rport;branch=z9hG4bK938901029
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>
Call-ID: 1937419984
CSeq: 21 INVITE
Contact: <sip:100@192.168.1.112:39059>
Authorization: Digest username="100", realm="asterisk", nonce="34225d72", uri="sip:101@eu.vancl.eu", response="a36f5b293a0667c06d76504da7b5d857", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Max-Forwards: 70
User-Agent: antisip/5.1.0-549-g2028176-Oct-15-2015 amdroid/4.2.5 HM 1SW/4.4.4
Subject: Talk
Supported: 100rel, replaces
Content-Length: 744

v=0
o=amsip 1989349040 0 IN IP4 192.168.1.112
s=talk
c=IN IP4 192.168.1.112
t=0 0
m=audio 10000 RTP/AVP 98 105 100 97 99 103 0 8 101
a=rtpmap:98 SILK/16000
a=rtpmap:105 OPUS/48000/2
a=fmtp:105 useinbandfec=1
a=rtpmap:100 speex/16000
a=fmtp:100 mode="6,any"
a=rtpmap:97 SILK/8000
a=rtpmap:99 speex/8000
a=fmtp:99 mode="6,any"
a=rtpmap:103 iLBC/8000
a=fmtp:103 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp-mux
m=video 10020 RTP/AVP 118 117 116 115 34
b=AS:128
a=rtpmap:118 VP8/90000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42800c; packetization-mode=1
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000
a=rtpmap:34 H263/90000
a=rtcp-mux
<------------->
--- (15 headers 30 lines) ---
Sending to 89.190.50.140:39059 (NAT)
Using INVITE request as basis request - 1937419984
Found peer '100' for '100' from 89.190.50.140:39059
Found RTP audio format 98
Found RTP audio format 105
Found RTP audio format 100
Found RTP audio format 97
Found RTP audio format 99
Found RTP audio format 103
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format SILK for ID 98
Found audio description format OPUS for ID 105
Found audio description format speex for ID 100
Found unknown media description format SILK for ID 97
Found audio description format speex for ID 99
Found audio description format iLBC for ID 103
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
Found RTP video format 118
Found RTP video format 117
Found RTP video format 116
Found RTP video format 115
Found RTP video format 34
Found video description format VP8 for ID 118
Found video description format H264 for ID 117
Found video description format MP4V-ES for ID 116
Found video description format H263-1998 for ID 115
Found video description format H263 for ID 34
Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw|alaw|speex|speex16|ilbc|opus)/video=(h263|h263p|mpeg4|h264|vp8)/text=(nothing), combined - (ulaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.112:10000
Peer video RTP is at port 192.168.1.112:10020
Looking for 101 in pokus (domain eu.vancl.eu)
sip_route_dump: route/path hop: <sip:100@192.168.1.112:39059>

<--- Transmitting (NAT) to 89.190.50.140:39059 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>
Call-ID: 1937419984
CSeq: 21 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@192.168.1.7:5060>
Content-Length: 0


<------------>
Audio is at 10964
Video is at 192.168.1.7:10272
Adding codec ulaw to SDP
Adding video codec h263 to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 89.190.50.140:5060:
INVITE sip:101@89.190.50.140:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK3a997588;rport
Max-Forwards: 70
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
To: <sip:101@89.190.50.140:5060>
Contact: <sip:100@192.168.1.7:5060>
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.6.0
Date: Thu, 12 Nov 2015 16:11:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 2143136553 2143136553 IN IP4 192.168.1.7
s=Asterisk PBX 13.6.0
c=IN IP4 192.168.1.7
b=CT:384
t=0 0
m=audio 10964 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10272 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

---

<--- SIP read from UDP:89.190.50.140:5060 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK3a997588;rport=5060;received=93.89.146.5
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
To: <sip:101@89.190.50.140:5060>
Content-Length: 0

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

<--- SIP read from UDP:89.190.50.140:5060 --->
SIP/2.0 180 Ringing
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK3a997588;rport=5060;received=93.89.146.5
User-Agent: Ekiga/4.0.1
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
To: "101" <sip:101@89.190.50.140>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
Contact: "101" <sip:martin@89.190.50.140>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:martin@89.190.50.140>

<--- Transmitting (NAT) to 89.190.50.140:39059 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>;tag=as75b76746
Call-ID: 1937419984
CSeq: 21 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@192.168.1.7:5060>
Content-Length: 0


<------------>
Retransmitting #7 (NAT) to 155.94.64.34:5089:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 155.94.64.34:5089;branch=z9hG4bK-0575c362977ea6340ca6fcbe6260b446;received=155.94.64.34;rport=5089
From: 101<sip:101@93.89.146.5>;tag=8a359551
To: 991130972597723173<sip:991130972597723173@93.89.146.5>;tag=as1e4415a6
Call-ID: 0575c362977ea6340ca6fcbe6260b446
CSeq: 1 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54467743"
Content-Length: 0


---

<--- SIP read from UDP:89.190.50.140:5060 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK3a997588;rport=5060;received=93.89.146.5
User-Agent: Ekiga/4.0.1
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
To: "101" <sip:101@89.190.50.140>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
Contact: "101" <sip:martin@89.190.50.140>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Content-Length: 328
Content-Type: application/sdp

v=0
o=- 1447344693 1 IN IP4 89.190.50.140
s=Ekiga/4.0.1
c=IN IP4 89.190.50.140
t=0 0
m=audio 5066 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:240
m=video 5068 RTP/AVP 34
b=AS:4096
b=TIAS:4096000
a=recvonly
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=3
<------------->
--- (11 headers 17 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw)/video=(h263)/text=(nothing), combined - (ulaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 89.190.50.140:5066
Peer video RTP is at port 89.190.50.140:5068
sip_route_dump: route/path hop: <sip:martin@89.190.50.140>
Transmitting (NAT) to 89.190.50.140:5060:
ACK sip:martin@89.190.50.140 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK75e4afb8;rport
Max-Forwards: 70
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
To: <sip:101@89.190.50.140:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
Contact: <sip:100@192.168.1.7:5060>
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
Audio is at 10466
Video is at 192.168.1.7:10800
Adding codec ulaw to SDP
Adding video codec h263 to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 89.190.50.140:39059 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>;tag=as75b76746
Call-ID: 1937419984
CSeq: 21 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@192.168.1.7:5060>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 1498995089 1498995089 IN IP4 192.168.1.7
s=Asterisk PBX 13.6.0
c=IN IP4 192.168.1.7
b=CT:384
t=0 0
m=audio 10466 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10800 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

<------------>
Audio is at 10964
Video is at 192.168.1.112:10020
Adding codec ulaw to SDP
Adding video codec h263 to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 89.190.50.140:5060:
INVITE sip:martin@89.190.50.140 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK212a6db9;rport
Max-Forwards: 70
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
To: <sip:101@89.190.50.140:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
Contact: <sip:100@192.168.1.7:5060>
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 425

v=0
o=root 2143136553 2143136554 IN IP4 192.168.1.112
s=Asterisk PBX 13.6.0
c=IN IP4 192.168.1.112
b=CT:384
t=0 0
m=audio 10000 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10020 RTP/AVP 34
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=0;QCIF=3;CIF=0;CIF4=0;CIF16=0;VGA=0;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---

<--- SIP read from UDP:89.190.50.140:5060 --->
SIP/2.0 100 Trying
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK212a6db9;rport=5060;received=93.89.146.5
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
To: <sip:101@89.190.50.140:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
Content-Length: 0

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

<--- SIP read from UDP:89.190.50.140:5060 --->
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK212a6db9;rport=5060;received=93.89.146.5
User-Agent: Ekiga/4.0.1
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
To: "101" <sip:101@89.190.50.140:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
Contact: "101" <sip:martin@89.190.50.140>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Content-Length: 328
Content-Type: application/sdp

v=0
o=- 1447344693 2 IN IP4 89.190.50.140
s=Ekiga/4.0.1
c=IN IP4 89.190.50.140
t=0 0
m=audio 5066 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:240
m=video 5068 RTP/AVP 34
b=AS:4096
b=TIAS:4096000
a=recvonly
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=3
<------------->
--- (11 headers 17 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw)/video=(h263)/text=(nothing), combined - (ulaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 89.190.50.140:5066
Peer video RTP is at port 89.190.50.140:5068
Transmitting (NAT) to 89.190.50.140:5060:
ACK sip:martin@89.190.50.140 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK51ea1e97;rport
Max-Forwards: 70
From: "(Testovaci uzivatel" <sip:100@192.168.1.7>;tag=as7a28997b
To: <sip:101@89.190.50.140:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
Contact: <sip:100@192.168.1.7:5060>
Call-ID: 6963e10f581e1ff829408ab10ece2a4a@192.168.1.7:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
Retransmitting #1 (NAT) to 89.190.50.140:39059:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>;tag=as75b76746
Call-ID: 1937419984
CSeq: 21 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@192.168.1.7:5060>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 1498995089 1498995089 IN IP4 192.168.1.7
s=Asterisk PBX 13.6.0
c=IN IP4 192.168.1.7
b=CT:384
t=0 0
m=audio 10466 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10800 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

---
Retransmitting #2 (NAT) to 89.190.50.140:39059:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>;tag=as75b76746
Call-ID: 1937419984
CSeq: 21 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@192.168.1.7:5060>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 1498995089 1498995089 IN IP4 192.168.1.7
s=Asterisk PBX 13.6.0
c=IN IP4 192.168.1.7
b=CT:384
t=0 0
m=audio 10466 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10800 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

---

<--- SIP read from UDP:89.190.50.140:5060 --->
OPTIONS sip:102@eu.vancl.eu SIP/2.0
Route: <sip:93.89.146.5:5060;lr>
CSeq: 67 OPTIONS
Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK8ccb37b9-c587-e511-9b83-fcaa148f322d;rport
User-Agent: Ekiga/4.0.1
From: <sip:101@eu.vancl.eu>;tag=70c937b9-c587-e511-9b83-fcaa148f322d
Call-ID: 0cbf37b9-c587-e511-9b83-fcaa148f322d@meo2
To: <sip:102@eu.vancl.eu>
Accept: application/sdp, application/media_control+xml, application/dtmf, application/dtmf-relay
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Expires: 0
Content-Length: 0
Max-Forwards: 70

<------------->
--- (13 headers 0 lines) ---
Sending to 89.190.50.140:5060 (NAT)
Looking for 102 in public (domain eu.vancl.eu)

<--- Transmitting (NAT) to 89.190.50.140:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK8ccb37b9-c587-e511-9b83-fcaa148f322d;received=89.190.50.140;rport=5060
From: <sip:101@eu.vancl.eu>;tag=70c937b9-c587-e511-9b83-fcaa148f322d
To: <sip:102@eu.vancl.eu>;tag=as6661812b
Call-ID: 0cbf37b9-c587-e511-9b83-fcaa148f322d@meo2
CSeq: 67 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0cbf37b9-c587-e511-9b83-fcaa148f322d@meo2' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:89.190.50.140:5060 --->
OPTIONS sip:101@eu.vancl.eu SIP/2.0
Route: <sip:93.89.146.5:5060;lr>
CSeq: 68 OPTIONS
Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK38e837b9-c587-e511-9b83-fcaa148f322d;rport
User-Agent: Ekiga/4.0.1
From: <sip:101@eu.vancl.eu>;tag=76e637b9-c587-e511-9b83-fcaa148f322d
Call-ID: a6de37b9-c587-e511-9b83-fcaa148f322d@meo2
To: <sip:101@eu.vancl.eu>
Accept: application/sdp, application/media_control+xml, application/dtmf, application/dtmf-relay
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Expires: 0
Content-Length: 0
Max-Forwards: 70

<------------->
--- (13 headers 0 lines) ---
Sending to 89.190.50.140:5060 (NAT)
Looking for 101 in public (domain eu.vancl.eu)

<--- Transmitting (NAT) to 89.190.50.140:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK38e837b9-c587-e511-9b83-fcaa148f322d;received=89.190.50.140;rport=5060
From: <sip:101@eu.vancl.eu>;tag=76e637b9-c587-e511-9b83-fcaa148f322d
To: <sip:101@eu.vancl.eu>;tag=as46ba0e06
Call-ID: a6de37b9-c587-e511-9b83-fcaa148f322d@meo2
CSeq: 68 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a6de37b9-c587-e511-9b83-fcaa148f322d@meo2' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:89.190.50.140:5060 --->
PUBLISH sip:102@eu.vancl.eu SIP/2.0
CSeq: 69 PUBLISH
Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK2616e4b9-c587-e511-9b83-fcaa148f322d;rport
User-Agent: Ekiga/4.0.1
From: <sip:102@eu.vancl.eu>
Call-ID: 96496414-c587-e511-9b83-fcaa148f322d@meo2
To: <sip:102@eu.vancl.eu>
Expires: 300
Event: presence
Content-Length: 482
Content-Type: application/pidf+xml
Max-Forwards: 70

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:102@eu.vancl.eu"> <tuple id="TB628D577"> <status> <basic>open</basic> </status> <contact priority="1">sip:102@eu.vancl.eu</contact> <note>I&apos;m available using Ekiga</note> <timestamp>2015-11-12T17:07:21+01:00</timestamp> </tuple>
</presence>
<------------->
--- (12 headers 3 lines) ---
Retransmitting #3 (NAT) to 89.190.50.140:39059:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>;tag=as75b76746
Call-ID: 1937419984
CSeq: 21 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@192.168.1.7:5060>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 1498995089 1498995089 IN IP4 192.168.1.7
s=Asterisk PBX 13.6.0
c=IN IP4 192.168.1.7
b=CT:384
t=0 0
m=audio 10466 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10800 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

---

<--- SIP read from UDP:89.190.50.140:5060 --->
PUBLISH sip:102@eu.vancl.eu SIP/2.0
CSeq: 69 PUBLISH
Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK2616e4b9-c587-e511-9b83-fcaa148f322d;rport
User-Agent: Ekiga/4.0.1
From: <sip:102@eu.vancl.eu>
Call-ID: 96496414-c587-e511-9b83-fcaa148f322d@meo2
To: <sip:102@eu.vancl.eu>
Expires: 300
Event: presence
Content-Length: 482
Content-Type: application/pidf+xml
Max-Forwards: 70

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:102@eu.vancl.eu"> <tuple id="TB628D577"> <status> <basic>open</basic> </status> <contact priority="1">sip:102@eu.vancl.eu</contact> <note>I&apos;m available using Ekiga</note> <timestamp>2015-11-12T17:07:21+01:00</timestamp> </tuple>
</presence>
<------------->
--- (12 headers 3 lines) ---
Retransmitting #8 (NAT) to 155.94.64.34:5089:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 155.94.64.34:5089;branch=z9hG4bK-0575c362977ea6340ca6fcbe6260b446;received=155.94.64.34;rport=5089
From: 101<sip:101@93.89.146.5>;tag=8a359551
To: 991130972597723173<sip:991130972597723173@93.89.146.5>;tag=as1e4415a6
Call-ID: 0575c362977ea6340ca6fcbe6260b446
CSeq: 1 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54467743"
Content-Length: 0


---
Really destroying SIP dialog '562256a7-c587-e511-9b83-fcaa148f322d@meo2' Method: OPTIONS
Really destroying SIP dialog 'b44156a7-c587-e511-9b83-fcaa148f322d@meo2' Method: OPTIONS

<--- SIP read from UDP:89.190.50.140:5060 --->
PUBLISH sip:102@eu.vancl.eu SIP/2.0
CSeq: 69 PUBLISH
Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK2616e4b9-c587-e511-9b83-fcaa148f322d;rport
User-Agent: Ekiga/4.0.1
From: <sip:102@eu.vancl.eu>
Call-ID: 96496414-c587-e511-9b83-fcaa148f322d@meo2
To: <sip:102@eu.vancl.eu>
Expires: 300
Event: presence
Content-Length: 482
Content-Type: application/pidf+xml
Max-Forwards: 70

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:102@eu.vancl.eu"> <tuple id="TB628D577"> <status> <basic>open</basic> </status> <contact priority="1">sip:102@eu.vancl.eu</contact> <note>I&apos;m available using Ekiga</note> <timestamp>2015-11-12T17:07:21+01:00</timestamp> </tuple>
</presence>
<------------->
--- (12 headers 3 lines) ---

<--- SIP read from UDP:89.190.50.140:5060 --->
PUBLISH sip:102@eu.vancl.eu SIP/2.0
CSeq: 69 PUBLISH
Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK2616e4b9-c587-e511-9b83-fcaa148f322d;rport
User-Agent: Ekiga/4.0.1
From: <sip:102@eu.vancl.eu>
Call-ID: 96496414-c587-e511-9b83-fcaa148f322d@meo2
To: <sip:102@eu.vancl.eu>
Expires: 300
Event: presence
Content-Length: 482
Content-Type: application/pidf+xml
Max-Forwards: 70

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:102@eu.vancl.eu"> <tuple id="TB628D577"> <status> <basic>open</basic> </status> <contact priority="1">sip:102@eu.vancl.eu</contact> <note>I&apos;m available using Ekiga</note> <timestamp>2015-11-12T17:07:21+01:00</timestamp> </tuple>
</presence>
<------------->
--- (12 headers 3 lines) ---
Retransmitting #4 (NAT) to 89.190.50.140:39059:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
From: <sip:100@eu.vancl.eu>;tag=726823752
To: <sip:101@eu.vancl.eu>;tag=as75b76746
Call-ID: 1937419984
CSeq: 21 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@192.168.1.7:5060>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 1498995089 1498995089 IN IP4 192.168.1.7
s=Asterisk PBX 13.6.0
c=IN IP4 192.168.1.7
b=CT:384
t=0 0
m=audio 10466 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10800 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

---
voip*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
root@voip:~#