<--- SIP read from UDP:192.168.58.212:5060 --->
INVITE sip:161@192.168.58.212 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.212:5060;rport;branch=z9hG4bK24009
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>
Call-ID: 11688
CSeq: 20 INVITE
Contact: <sip:191@192.168.58.212:5060>
Content-Type: application/sdp
Allow: REGISTER, INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, NOTIFY
Max-Forwards: 70
User-Agent: 2N Helios IP Verso 2.9.1.18.6
Content-Length:   579

v=0
o=- 1369652685 1080447644 IN IP4 192.168.58.212
s=HIP 2.9.1.18.6
t=0 0
m=audio 5000 RTP/AVP 9 0 8 101
c=IN IP4 192.168.58.212
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
m=video 5002 RTP/AVP 123 124 98 34
c=IN IP4 192.168.58.212
a=sendonly
a=rtpmap:123 H264/90000
a=fmtp:123 profile-level-id=42801E; packetization-mode=1
a=rtpmap:124 H264/90000
a=fmtp:124 profile-level-id=42801E; packetization-mode=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 QCIF=1; CIF=1
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1; CIF=1

<------------->
--- (12 headers 21 lines) ---
Sending to 192.168.58.212:5060 (no NAT)
Using INVITE request as basis request - 11688
Found peer '191' for '191' from 192.168.58.212:5060

<--- Reliably Transmitting (no NAT) to 192.168.58.212:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.58.212:5060;branch=z9hG4bK24009;received=192.168.58.212;rport=5060
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>;tag=as2bde6c06
Call-ID: 11688
CSeq: 20 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37846937"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '11688' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.58.212:5060 --->
ACK sip:161@192.168.58.212 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.212:5060;rport;branch=z9hG4bK24009
Route: <sip:192.168.58.2;lr>
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>;tag=as2bde6c06
Call-ID: 11688
CSeq: 20 ACK
Content-Length: 0


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

<--- SIP read from UDP:192.168.58.212:5060 --->
INVITE sip:161@192.168.58.212 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.212:5060;rport;branch=z9hG4bK21438
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>
Call-ID: 11688
CSeq: 21 INVITE
Contact: <sip:191@192.168.58.212:5060>
Authorization: Digest username="191", realm="asterisk", nonce="37846937", uri="sip:161@192.168.58.212", response="2ac3667d78246bd0433ed767c4cfb6e4", algorithm=MD5
Content-Type: application/sdp
Allow: REGISTER, INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, NOTIFY
Max-Forwards: 70
User-Agent: 2N Helios IP Verso 2.9.1.18.6
Content-Length:   579

v=0
o=- 1369652685 1080447644 IN IP4 192.168.58.212
s=HIP 2.9.1.18.6
t=0 0
m=audio 5000 RTP/AVP 9 0 8 101
c=IN IP4 192.168.58.212
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
m=video 5002 RTP/AVP 123 124 98 34
c=IN IP4 192.168.58.212
a=sendonly
a=rtpmap:123 H264/90000
a=fmtp:123 profile-level-id=42801E; packetization-mode=1
a=rtpmap:124 H264/90000
a=fmtp:124 profile-level-id=42801E; packetization-mode=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 QCIF=1; CIF=1
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1; CIF=1

<------------->
--- (13 headers 21 lines) ---
Sending to 192.168.58.212:5060 (no NAT)
Using INVITE request as basis request - 11688
Found peer '191' for '191' from 192.168.58.212:5060
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
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 audio description format telephone-event for ID 101
Found RTP video format 123
Found RTP video format 124
Found RTP video format 98
Found RTP video format 34
Found video description format H264 for ID 123
Found video description format H264 for ID 124
Found video description format H263-1998 for ID 98
Found video description format H263 for ID 34
Capabilities: us - 0x200006 (gsm|ulaw|h264), peer - audio=0x100c (ulaw|alaw|g722)/video=0x380000 (h263|h263p|h264)/text=0x0 (nothing), combined - 0x200004 (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.58.212:5000
Peer video RTP is at port 192.168.58.212:5002
Looking for 161 in DLPN_DialPlan1Local (domain 192.168.58.212)
list_route: hop: <sip:191@192.168.58.212:5060>

<--- Transmitting (no NAT) to 192.168.58.212:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.58.212:5060;branch=z9hG4bK21438;received=192.168.58.212;rport=5060
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>
Call-ID: 11688
CSeq: 21 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:161@192.168.58.2:5060>
Content-Length: 0


<------------>
Audio is at 17898
Video is at 192.168.58.2:18038
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.58.91:5060:
INVITE sip:161@192.168.58.91:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.2:5060;branch=z9hG4bK263cd635
Max-Forwards: 70
From: "Vratnik" <sip:191@192.168.58.2>;tag=as43f0de51
To: <sip:161@192.168.58.91:5060>
Contact: <sip:191@192.168.58.2:5060>
Call-ID: 1a3e9476172196981db1743b04644cfe@192.168.58.2:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 13 Jan 2015 09:45:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 331

v=0
o=root 753582894 753582894 IN IP4 192.168.58.2
s=Asterisk PBX 1.8.31.1
c=IN IP4 192.168.58.2
b=CT:2048
t=0 0
m=audio 17898 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=ptime:20
a=sendrecv
m=video 18038 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

---
[Jan 13 10:45:19] WARNING[5022]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Subscriber absent)

<--- SIP read from UDP:192.168.58.91:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.58.2:5060;branch=z9hG4bK263cd635
From: "Vratnik" <sip:191@192.168.58.2>;tag=as43f0de51
To: <sip:161@192.168.58.91:5060>
Call-ID: 1a3e9476172196981db1743b04644cfe@192.168.58.2:5060
CSeq: 102 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.6
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


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

<--- SIP read from UDP:192.168.58.91:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.58.2:5060;branch=z9hG4bK263cd635
From: "Vratnik" <sip:191@192.168.58.2>;tag=as43f0de51
To: <sip:161@192.168.58.91:5060>;tag=1966197021
Call-ID: 1a3e9476172196981db1743b04644cfe@192.168.58.2:5060
CSeq: 102 INVITE
Contact: <sip:161@192.168.58.91:5060>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.6
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
list_route: hop: <sip:161@192.168.58.91:5060>

<--- Transmitting (no NAT) to 192.168.58.212:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.58.212:5060;branch=z9hG4bK21438;received=192.168.58.212;rport=5060
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>;tag=as54ca8ac4
Call-ID: 11688
CSeq: 21 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:161@192.168.58.2:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.58.91:5060 --->

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

<--- SIP read from UDP:192.168.58.76:5060 --->

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

<--- SIP read from UDP:192.168.58.50:5060 --->

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

<--- SIP read from UDP:192.168.58.91:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.58.2:5060;branch=z9hG4bK263cd635
From: "Vratnik" <sip:191@192.168.58.2>;tag=as43f0de51
To: <sip:161@192.168.58.91:5060>;tag=1966197021
Call-ID: 1a3e9476172196981db1743b04644cfe@192.168.58.2:5060
CSeq: 102 INVITE
Contact: <sip:161@192.168.58.91:5060>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.6
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   308

v=0
o=161 8000 8000 IN IP4 192.168.58.91
s=SIP Call
c=IN IP4 192.168.58.91
t=0 0
m=audio 5004 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 5006 RTP/AVP 99
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42800B

<------------->
--- (12 headers 15 lines) ---
list_route: hop: <sip:161@192.168.58.91:5060>
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 99
Found video description format H264 for ID 99
Capabilities: us - 0x200006 (gsm|ulaw|h264), peer - audio=0x4 (ulaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x200004 (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.58.91:5004
Peer video RTP is at port 192.168.58.91:5006
Audio is at 15336
Video is at 192.168.58.2:11800
Adding codec 0x4 (ulaw) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.58.212:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.58.212:5060;branch=z9hG4bK21438;received=192.168.58.212;rport=5060
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>;tag=as54ca8ac4
Call-ID: 11688
CSeq: 21 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:161@192.168.58.2:5060>
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 353696357 353696357 IN IP4 192.168.58.2
s=Asterisk PBX 1.8.31.1
c=IN IP4 192.168.58.2
b=CT:2048
t=0 0
m=audio 15336 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 11800 RTP/AVP 123
a=rtpmap:123 H264/90000
a=sendrecv

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

<--- SIP read from UDP:192.168.58.175:5060 --->

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

<--- SIP read from UDP:192.168.58.159:5060 --->

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

<--- SIP read from UDP:192.168.58.77:5060 --->

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

<--- SIP read from UDP:192.168.58.91:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.58.2:5060;branch=z9hG4bK263cd635
From: "Vratnik" <sip:191@192.168.58.2>;tag=as43f0de51
To: <sip:161@192.168.58.91:5060>;tag=1966197021
Call-ID: 1a3e9476172196981db1743b04644cfe@192.168.58.2:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.6
Warning: 399 GS "The call is rejected"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:161@192.168.58.91:5060> for address/port to send to
set_destination: set destination to 192.168.58.91:5060
Transmitting (no NAT) to 192.168.58.91:5060:
ACK sip:161@192.168.58.91:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.2:5060;branch=z9hG4bK263cd635
Max-Forwards: 70
From: "Vratnik" <sip:191@192.168.58.2>;tag=as43f0de51
To: <sip:161@192.168.58.91:5060>;tag=1966197021
Contact: <sip:191@192.168.58.2:5060>
Call-ID: 1a3e9476172196981db1743b04644cfe@192.168.58.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- Reliably Transmitting (no NAT) to 192.168.58.212:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.58.212:5060;branch=z9hG4bK21438;received=192.168.58.212;rport=5060
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>;tag=as54ca8ac4
Call-ID: 11688
CSeq: 21 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0


<------------>
Really destroying SIP dialog '1a3e9476172196981db1743b04644cfe@192.168.58.2:5060' Method: INVITE

<--- SIP read from UDP:192.168.58.212:5060 --->
ACK sip:161@192.168.58.212 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.212:5060;rport;branch=z9hG4bK21438
Route: <sip:192.168.58.2;lr>
From: "Vratnik" <sip:191@192.168.58.212>;tag=11829
To: <sip:161@192.168.58.212>;tag=as54ca8ac4
Call-ID: 11688
CSeq: 21 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '11688' Method: ACK

<--- SIP read from UDP:192.168.58.75:11714 --->

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

<--- SIP read from UDP:192.168.58.16:5060 --->

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