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Повторный вызов не проходит...

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Повторный вызов не проходит...

Добавлено: 05 окт 2015, 21:40
ramen
Приветствую всех присутствующих! заметил очень странную вещь, если сделать звонок и поговорить, а далее в течении до пяти минут попробовать еще раз позвонить, то мы не слышим и не видим друг друга, такая ситуация возникает только с софтфоном Linphone, куда бы от него не звонили или на него... Со всеми остальными все ОК, куда можно копать в этом случае?
строки статуса:

Код: Выделить всё

 
 == Spawn extension (legenda, 8002, 1) exited non-zero on 'SIP/8003-00000000'
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Executing [8002@legenda:1] Dial("SIP/8005-00000002", "SIP/8002")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/8002
    -- SIP/8002-00000003 is ringing
    -- SIP/8002-00000003 answered SIP/8005-00000002
    -- Remotely bridging SIP/8005-00000002 and SIP/8002-00000003
       > 0x7f2704026570 -- Probation passed - setting RTP source address to 192.168.111.177:15000
       > 0x7f270400bc00 -- Probation passed - setting RTP source address to 192.168.111.177:15002
       > 0x7f26ec07d060 -- Probation passed - setting RTP source address to 5.200.43.195:7078
       > 0x7f26ec07d060 -- Probation passed - setting RTP source address to 5.200.43.195:7078
       > 0x7f26ec07d060 -- Probation passed - setting RTP source address to 5.200.43.195:7078
       > 0x7f26ec07d060 -- Probation passed - setting RTP source address to 5.200.43.195:7078
       > 0x7f26ec080900 -- Probation passed - setting RTP source address to 5.200.43.195:9070
       > 0x7f26ec080900 -- Probation passed - setting RTP source address to 5.200.43.195:9070
  == Spawn extension (legenda, 8002, 1) exited non-zero on 'SIP/8005-00000002'
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Executing [8002@legenda:1] Dial("SIP/8005-00000004", "SIP/8002")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/8002
    -- SIP/8002-00000005 is ringing
    -- SIP/8002-00000005 answered SIP/8005-00000004
    -- Remotely bridging SIP/8005-00000004 and SIP/8002-00000005
       > 0x7f27040030c0 -- Probation passed - setting RTP source address to 192.168.111.177:15000
       > 0x7f270400a240 -- Probation passed - setting RTP source address to 192.168.111.177:15002
  == Spawn extension (legenda, 8002, 1) exited non-zero on 'SIP/8005-00000004'
asterisk*CLI>

Re: Повторный вызов не проходит...

Добавлено: 05 окт 2015, 21:54
ramen
Удачный вызов

Код: Выделить всё




<--- SIP read from UDP:77.244.21.249:49428 --->


<------------->
Really destroying SIP dialog 'V0ZNvFHp-nn2Tt0aaXc7Ow..' Method: REGISTER

<--- SIP read from UDP:5.200.43.195:5066 --->
INVITE sip:8002@109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.uhoqUyOLE;rport
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: sip:8002@109.188.136.85
CSeq: 20 INVITE
Call-ID: JNI25YX8gR
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 308
Contact: <sip:8005@5.200.43.195:5066>;+sip.instance="<urn:uuid:bc2d01b7-91b4-4bd9-8108-ed9a7b46e94c>"
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)

v=0
o=8005 2290 2989 IN IP4 192.168.10.151
s=Talk
c=IN IP4 192.168.10.151
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
<------------->
--- (13 headers 11 lines) ---
Sending to 5.200.43.195:5066 (NAT)
Sending to 5.200.43.195:5066 (NAT)
Using INVITE request as basis request - JNI25YX8gR
Found peer '8005' for '8005' from 5.200.43.195:5066

<--- Reliably Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.uhoqUyOLE;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: sip:8002@109.188.136.85;tag=as0d8ffc9e
Call-ID: JNI25YX8gR
CSeq: 20 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a504327"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'JNI25YX8gR' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:217.66.159.103:8539 --->


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

<--- SIP read from UDP:5.200.43.195:5066 --->
ACK sip:8002@109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.uhoqUyOLE;rport
Call-ID: JNI25YX8gR
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as0d8ffc9e
Contact: <sip:8005@5.200.43.195:5066>;+sip.instance="<urn:uuid:bc2d01b7-91b4-4bd9-8108-ed9a7b46e94c>"
Max-Forwards: 70
CSeq: 20 ACK

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

<--- SIP read from UDP:5.200.43.195:5066 --->
INVITE sip:8002@109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.AnZoJnAwb;rport
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: sip:8002@109.188.136.85
CSeq: 21 INVITE
Call-ID: JNI25YX8gR
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 308
Contact: <sip:8005@5.200.43.195:5066>;+sip.instance="<urn:uuid:bc2d01b7-91b4-4bd9-8108-ed9a7b46e94c>"
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Authorization: Digest realm="asterisk", nonce="7a504327", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85", response="a1bd44a3c4e0e84cd480b4e4184f5fab"

v=0
o=8005 2290 2989 IN IP4 192.168.10.151
s=Talk
c=IN IP4 192.168.10.151
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
<------------->
--- (14 headers 11 lines) ---
Sending to 5.200.43.195:5066 (NAT)
Using INVITE request as basis request - JNI25YX8gR
Found peer '8005' for '8005' from 5.200.43.195:5066
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found video description format H264 for ID 96
Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer - audio=(ulaw|alaw)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.10.151:7078
Peer video RTP is at port 192.168.10.151:9070
Peer doesn't provide T.140
Looking for 8002 in legenda (domain 109.188.136.85)
list_route: hop: <sip:8005@5.200.43.195:5066>

<--- Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.AnZoJnAwb;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: sip:8002@109.188.136.85
Call-ID: JNI25YX8gR
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8002@109.188.136.85:5060>
Content-Length: 0


<------------>
    -- Executing [8002@legenda:1] Dial("SIP/8005-00000012", "SIP/8002")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 13156
Video is at 192.168.111.111:18530
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding video codec 200002 (h263) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3cae761d;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 05 Oct 2015 18:49:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 428

v=0
o=root 1375860194 1375860194 IN IP4 192.168.111.111
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.111
b=CT:2048
t=0 0
m=audio 13156 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 18530 RTP/AVP 96 34
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---
    -- Called SIP/8002

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3cae761d;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 102 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3cae761d;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 102 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
MaxRingingTime: 45
MaxConnectingTime: 120
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:8002@192.168.111.177:5060>
    -- SIP/8002-00000013 is ringing

<--- Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.AnZoJnAwb;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: sip:8002@109.188.136.85;tag=as67e5921a
Call-ID: JNI25YX8gR
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8002@109.188.136.85:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3cae761d;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 102 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
Found RTP video format 96
Found video description format H264 for ID 96
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.111.177:15000
Peer video RTP is at port 192.168.111.177:15002
Peer doesn't provide T.140
list_route: hop: <sip:8002@192.168.111.177:5060>
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK24e9f1fc;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
    -- SIP/8002-00000013 answered SIP/8005-00000012
Audio is at 15706
Video is at 109.188.136.85:15364
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100004 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.AnZoJnAwb;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: sip:8002@109.188.136.85;tag=as67e5921a
Call-ID: JNI25YX8gR
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8002@109.188.136.85:5060>
Content-Type: application/sdp
Content-Length: 329

v=0
o=root 1667269224 1667269224 IN IP4 109.188.136.85
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 109.188.136.85
b=CT:2048
t=0 0
m=audio 15706 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 15364 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

<------------>
    -- Remotely bridging SIP/8005-00000012 and SIP/8002-00000013
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 13156
Video is at 192.168.10.151:9070
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7bf28fa4;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 1375860194 1375860195 IN IP4 192.168.10.151
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.10.151
b=CT:2048
t=0 0
m=audio 7078 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7bf28fa4;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 103 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #1 (NAT) to 5.200.43.195:5066:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.AnZoJnAwb;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: sip:8002@109.188.136.85;tag=as67e5921a
Call-ID: JNI25YX8gR
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8002@109.188.136.85:5060>
Content-Type: application/sdp
Content-Length: 329

v=0
o=root 1667269224 1667269224 IN IP4 109.188.136.85
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 109.188.136.85
b=CT:2048
t=0 0
m=audio 15706 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 15364 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---
       > 0x7f2704025df0 -- Probation passed - setting RTP source address to 192.168.111.177:15000
       > 0x7f270400bc00 -- Probation passed - setting RTP source address to 192.168.111.177:15002

<--- SIP read from UDP:5.200.43.195:5066 --->
ACK sip:8002@109.188.136.85:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;rport;branch=z9hG4bK.nSABDKfKn
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as67e5921a
CSeq: 21 ACK
Call-ID: JNI25YX8gR
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="7a504327", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85", response="a1bd44a3c4e0e84cd480b4e4184f5fab"

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:8005@5.200.43.195:5066> for address/port to send to
set_destination: set destination to 5.200.43.195:5066
Audio is at 15706
Video is at 192.168.111.177:15002
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 5.200.43.195:5066:
INVITE sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK115f3017;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as67e5921a
To: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: JNI25YX8gR
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 1667269224 1667269225 IN IP4 192.168.111.177
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.177
b=CT:2048
t=0 0
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 15002 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---
Retransmitting #1 (NAT) to 5.200.43.195:5066:
INVITE sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK115f3017;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as67e5921a
To: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: JNI25YX8gR
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 1667269224 1667269225 IN IP4 192.168.111.177
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.177
b=CT:2048
t=0 0
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 15002 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---
       > 0x7f26ec004d50 -- Probation passed - setting RTP source address to 5.200.43.195:7078
       > 0x7f26ec004d50 -- Probation passed - setting RTP source address to 5.200.43.195:7078

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7bf28fa4;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 103 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK6bda1a99;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 13156
Video is at 192.168.10.151:9070
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK16ed3560;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 299

v=0
o=root 1375860194 1375860196 IN IP4 5.200.43.195
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 5.200.43.195
b=CT:2048
t=0 0
m=audio 7078 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK16ed3560;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 104 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #2 (NAT) to 5.200.43.195:5066:
INVITE sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK115f3017;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as67e5921a
To: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: JNI25YX8gR
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 1667269224 1667269225 IN IP4 192.168.111.177
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.177
b=CT:2048
t=0 0
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 15002 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK16ed3560;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 104 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK0c250aca;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Retransmitting #3 (NAT) to 5.200.43.195:5066:
INVITE sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK115f3017;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as67e5921a
To: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: JNI25YX8gR
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 1667269224 1667269225 IN IP4 192.168.111.177
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.177
b=CT:2048
t=0 0
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 15002 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:217.66.159.103:37230 --->


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

<--- SIP read from UDP:5.200.43.195:5066 --->
ACK sip:8002@109.188.136.85:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.nSABDKfKn;rport
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as67e5921a
CSeq: 21 ACK
Call-ID: JNI25YX8gR
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="7a504327", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85", response="a1bd44a3c4e0e84cd480b4e4184f5fab"

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

<--- SIP read from UDP:5.200.43.195:5066 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK115f3017;rport
From: <sip:8002@109.188.136.85>;tag=as67e5921a
To: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
Call-ID: JNI25YX8gR
CSeq: 102 INVITE

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

<--- SIP read from UDP:5.200.43.195:5066 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK115f3017;rport
From: <sip:8002@109.188.136.85>;tag=as67e5921a
To: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
Call-ID: JNI25YX8gR
CSeq: 102 INVITE
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: <sip:8005@5.200.43.195:5066>;+sip.instance="<urn:uuid:bc2d01b7-91b4-4bd9-8108-ed9a7b46e94c>"
Content-Type: application/sdp
Content-Length: 193

v=0
o=8005 2290 2991 IN IP4 192.168.10.151
s=Talk
c=IN IP4 192.168.10.151
t=0 0
m=audio 7078 RTP/AVP 0
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found RTP video format 96
Found video description format H264 for ID 96
Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.10.151:7078
Peer video RTP is at port 192.168.10.151:9070
Peer doesn't provide T.140
set_destination: Parsing <sip:8005@5.200.43.195:5066> for address/port to send to
set_destination: set destination to 5.200.43.195:5066
Transmitting (NAT) to 5.200.43.195:5066:
ACK sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK71908ff0;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as67e5921a
To: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: JNI25YX8gR
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 13156
Video is at 192.168.10.151:9070
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK6fe7a5f0;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 105 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 1375860194 1375860197 IN IP4 192.168.10.151
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.10.151
b=CT:2048
t=0 0
m=audio 7078 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK6fe7a5f0;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 105 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
       > 0x7f26ec004d50 -- Probation passed - setting RTP source address to 5.200.43.195:7078
       > 0x7f26ec004d50 -- Probation passed - setting RTP source address to 5.200.43.195:7078

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK6fe7a5f0;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 105 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK351eaa8e;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 105 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 13156
Video is at 192.168.10.151:9070
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK351b1dd8;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 106 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 299

v=0
o=root 1375860194 1375860198 IN IP4 5.200.43.195
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 5.200.43.195
b=CT:2048
t=0 0
m=audio 7078 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK351b1dd8;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 106 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK351b1dd8;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 106 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK432cf473;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 106 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
       > 0x7f26ec00bed0 -- Probation passed - setting RTP source address to 5.200.43.195:9070
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 13156
Video is at 5.200.43.195:9070
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK748cb090;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 107 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 299

v=0
o=root 1375860194 1375860199 IN IP4 5.200.43.195
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 5.200.43.195
b=CT:2048
t=0 0
m=audio 7078 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---
       > 0x7f26ec00bed0 -- Probation passed - setting RTP source address to 5.200.43.195:9070

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK748cb090;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 107 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK748cb090;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 107 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK5105cfed;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 107 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---

<--- SIP read from UDP:5.200.43.195:5066 --->
INFO sip:8002@109.188.136.85:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.Q273rInpU;rport
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as67e5921a
CSeq: 22 INFO
Call-ID: JNI25YX8gR
Max-Forwards: 70
Content-Type: application/media_control+xml
Content-Length: 185
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Authorization: Digest realm="asterisk", nonce="7a504327", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85:5060", response="20021b77886d81ab349b0805b4bd3361"

<?xml version="1.0" encoding="utf-8" ?><media_control> <vc_primitive> <to_encoder> <picture_fast_update></picture_fast_update> </to_encoder> </vc_primitive></media_control>
<------------->
--- (11 headers 1 lines) ---
Receiving INFO!

<--- Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.Q273rInpU;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as67e5921a
Call-ID: JNI25YX8gR
CSeq: 22 INFO
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INFO sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK275436a7;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 108 INFO
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK275436a7;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 108 INFO
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '126943OWQ0ODVmZGMwOTYyNGYwODQ0YzMxYWQ1MDRmZWE5ODI' Method: REGISTER

<--- SIP read from UDP:192.168.111.177:5060 --->
REGISTER sip:8002@192.168.111.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.177:5060;rport;branch=z9hG4bK2041236170
From: <sip:8002@192.168.111.111:5060>;tag=1326076078
To: <sip:8002@192.168.111.111:5060>
Call-ID: 504371200@192.168.111.177
CSeq: 1107 REGISTER
Contact: <sip:8002@192.168.111.177:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Expires: 3600
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.111.177:5060 (NAT)
Sending to 192.168.111.177:5060 (NAT)

<--- Transmitting (NAT) to 192.168.111.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.177:5060;branch=z9hG4bK2041236170;received=192.168.111.177;rport=5060
From: <sip:8002@192.168.111.111:5060>;tag=1326076078
To: <sip:8002@192.168.111.111:5060>;tag=as38948ef8
Call-ID: 504371200@192.168.111.177
CSeq: 1107 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2255220b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '504371200@192.168.111.177' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.111.177:5060 --->
REGISTER sip:8002@192.168.111.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.177:5060;rport;branch=z9hG4bK1762569806
From: <sip:8002@192.168.111.111:5060>;tag=1326076078
To: <sip:8002@192.168.111.111:5060>
Call-ID: 504371200@192.168.111.177
CSeq: 1108 REGISTER
Contact: <sip:8002@192.168.111.177:5060>
Authorization: Digest username="8002", realm="asterisk", nonce="2255220b", uri="sip:8002@192.168.111.111:5060", response="00dffd84bb71e773601e5fc4adb0845b", algorithm=MD5
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Expires: 3600
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.111.177:5060 (NAT)
Reliably Transmitting (NAT) to 192.168.111.177:5060:
OPTIONS sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK1c8ad8b4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as364622da
To: <sip:8002@192.168.111.177:5060>
Contact: <sip:asterisk@192.168.111.111:5060>
Call-ID: 4da2fef0338c7a844e421d933bf2649c@192.168.111.111:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 05 Oct 2015 18:49:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.177:5060;branch=z9hG4bK1762569806;received=192.168.111.177;rport=5060
From: <sip:8002@192.168.111.111:5060>;tag=1326076078
To: <sip:8002@192.168.111.111:5060>;tag=as38948ef8
Call-ID: 504371200@192.168.111.177
CSeq: 1108 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:8002@192.168.111.177:5060>;expires=3600
Date: Mon, 05 Oct 2015 18:49:29 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '504371200@192.168.111.177' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK1c8ad8b4;rport=5060
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as364622da
To: <sip:8002@192.168.111.177:5060>;tag=1895330675
Call-ID: 4da2fef0338c7a844e421d933bf2649c@192.168.111.111:5060
CSeq: 102 OPTIONS
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4da2fef0338c7a844e421d933bf2649c@192.168.111.111:5060' Method: OPTIONS

<--- SIP read from UDP:5.200.43.195:5066 --->


<------------->
Really destroying SIP dialog '1975867064@192.168.111.34' Method: REGISTER

<--- SIP read from UDP:5.200.43.195:5066 --->
INFO sip:8002@109.188.136.85:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.AUtl4zwmO;rport
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as67e5921a
CSeq: 23 INFO
Call-ID: JNI25YX8gR
Max-Forwards: 70
Content-Type: application/media_control+xml
Content-Length: 185
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Authorization: Digest realm="asterisk", nonce="7a504327", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85:5060", response="20021b77886d81ab349b0805b4bd3361"

<?xml version="1.0" encoding="utf-8" ?><media_control> <vc_primitive> <to_encoder> <picture_fast_update></picture_fast_update> </to_encoder> </vc_primitive></media_control>
<------------->
--- (11 headers 1 lines) ---
Receiving INFO!

<--- Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.AUtl4zwmO;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as67e5921a
Call-ID: JNI25YX8gR
CSeq: 23 INFO
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INFO sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7d7df9cd;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 109 INFO
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7d7df9cd;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 109 INFO
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

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

<--- SIP read from UDP:5.200.43.195:5066 --->
BYE sip:8002@109.188.136.85:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.gShdnr8JH;rport
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as67e5921a
CSeq: 24 BYE
Call-ID: JNI25YX8gR
Max-Forwards: 70
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Authorization: Digest realm="asterisk", nonce="7a504327", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85:5060", response="b26ca6f234d24d31576bc5810f9ee0dd"

<------------->
--- (9 headers 0 lines) ---
Sending to 5.200.43.195:5066 (NAT)
Scheduling destruction of SIP dialog 'JNI25YX8gR' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.gShdnr8JH;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=ZtBzlpHaZ
To: <sip:8002@109.188.136.85>;tag=as67e5921a
Call-ID: JNI25YX8gR
CSeq: 24 BYE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 13156
Video is at 192.168.111.111:18530
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3a0c57db;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 110 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 366

v=0
o=root 1375860194 1375860200 IN IP4 192.168.111.111
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.111
b=CT:2048
t=0 0
m=audio 13156 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 18530 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3a0c57db;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 110 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060' in 6400 ms (Method: INVITE)
  == Spawn extension (legenda, 8002, 1) exited non-zero on 'SIP/8005-00000012'

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3a0c57db;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 110 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK0eedb2f1;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 110 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Reliably Transmitting (NAT) to 192.168.111.177:5060:
BYE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK4dd22a08;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 111 BYE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog '1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK4dd22a08;rport=5060
From: <sip:8005@192.168.111.111>;tag=as1c90380d
To: <sip:8002@192.168.111.177:5060>;tag=1175231195
Call-ID: 1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060
CSeq: 111 BYE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1f2c64893488d0b40c72561a1eb62a3a@192.168.111.111:5060' Method: INVITE

<--- SIP read from UDP:217.66.159.103:8539 --->


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

<--- SIP read from UDP:217.66.159.103:37230 --->


<------------->
asterisk*CLI>

Re: Повторный вызов не проходит...

Добавлено: 05 окт 2015, 21:54
ramen
далее не удачный

Код: Выделить всё

<------------>
Scheduling destruction of SIP dialog '1975867064@192.168.111.34' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.111.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK6d53d9bb;rport=5060
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as34570daa
To: <sip:8004@192.168.111.34:5060>;tag=610532607
Call-ID: 57c6c24c3dbba401135a35a56157fbc7@192.168.111.111:5060
CSeq: 102 OPTIONS
User-Agent: Dahua UAC/3.0 VTH1510 V1.100.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '57c6c24c3dbba401135a35a56157fbc7@192.168.111.111:5060' Method: OPTIONS

<--- SIP read from UDP:217.66.159.103:8539 --->


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

<--- SIP read from UDP:5.200.43.195:5066 --->


<------------->
Really destroying SIP dialog '504371200@192.168.111.177' Method: REGISTER

<--- SIP read from UDP:217.66.159.103:37230 --->


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

<--- SIP read from UDP:217.66.159.103:8539 --->


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

<--- SIP read from UDP:5.200.43.195:5066 --->


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

<--- SIP read from UDP:217.66.159.103:37230 --->
REGISTER sip:109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 10.153.220.38:5060;branch=z9hG4bK.4cIjk9zt0;rport
From: <sip:80030@109.188.136.85>;tag=VAKzRJcMf
To: sip:80030@109.188.136.85
CSeq: 104 REGISTER
Call-ID: aThNjQS3Lz
Max-Forwards: 70
Supported: outbound
Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml
Contact: <sip:80030@217.66.159.103:37230>;+sip.instance="<urn:uuid:1360ee8d-c008-473a-95d5-f0f2bae3e92f>"
Expires: 2000
User-Agent: linphone/3.8.5-395-ge7dd35e (belle-sip/1.4.1)

<------------->
--- (12 headers 0 lines) ---
Sending to 217.66.159.103:37230 (NAT)
Sending to 217.66.159.103:37230 (NAT)

<--- Transmitting (NAT) to 217.66.159.103:37230 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.153.220.38:5060;branch=z9hG4bK.4cIjk9zt0;received=217.66.159.103;rport=37230
From: <sip:80030@109.188.136.85>;tag=VAKzRJcMf
To: sip:80030@109.188.136.85;tag=as229b0dfc
Call-ID: aThNjQS3Lz
CSeq: 104 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ffa5b3d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'aThNjQS3Lz' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:77.244.21.249:49428 --->


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

<--- SIP read from UDP:217.66.159.103:37230 --->
REGISTER sip:109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 10.153.220.38:5060;branch=z9hG4bK.4cIjk9zt0;rport
From: <sip:80030@109.188.136.85>;tag=VAKzRJcMf
To: sip:80030@109.188.136.85
CSeq: 104 REGISTER
Call-ID: aThNjQS3Lz
Max-Forwards: 70
Supported: outbound
Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml
Contact: <sip:80030@217.66.159.103:37230>;+sip.instance="<urn:uuid:1360ee8d-c008-473a-95d5-f0f2bae3e92f>"
Expires: 2000
User-Agent: linphone/3.8.5-395-ge7dd35e (belle-sip/1.4.1)

<------------->
--- (12 headers 0 lines) ---
Sending to 217.66.159.103:37230 (NAT)

<--- Transmitting (NAT) to 217.66.159.103:37230 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.153.220.38:5060;branch=z9hG4bK.4cIjk9zt0;received=217.66.159.103;rport=37230
From: <sip:80030@109.188.136.85>;tag=VAKzRJcMf
To: sip:80030@109.188.136.85;tag=as229b0dfc
Call-ID: aThNjQS3Lz
CSeq: 104 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ffa5b3d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'aThNjQS3Lz' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:217.66.159.103:37230 --->
REGISTER sip:109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 10.153.220.38:5060;branch=z9hG4bK.~ftYHbwzX;rport
From: <sip:80030@109.188.136.85>;tag=VAKzRJcMf
To: sip:80030@109.188.136.85
CSeq: 105 REGISTER
Call-ID: aThNjQS3Lz
Max-Forwards: 70
Supported: outbound
Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml
Contact: <sip:80030@217.66.159.103:37230>;+sip.instance="<urn:uuid:1360ee8d-c008-473a-95d5-f0f2bae3e92f>"
Expires: 2000
User-Agent: linphone/3.8.5-395-ge7dd35e (belle-sip/1.4.1)
Authorization: Digest realm="asterisk", nonce="6ffa5b3d", algorithm=MD5, username="80030", uri="sip:109.188.136.85", response="ca815baa1e827a97d976342ec0c43368"

<------------->
--- (13 headers 0 lines) ---
Sending to 217.66.159.103:37230 (NAT)

<--- Transmitting (NAT) to 217.66.159.103:37230 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.153.220.38:5060;branch=z9hG4bK.~ftYHbwzX;received=217.66.159.103;rport=37230
From: <sip:80030@109.188.136.85>;tag=VAKzRJcMf
To: sip:80030@109.188.136.85;tag=as229b0dfc
Call-ID: aThNjQS3Lz
CSeq: 105 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct  5 21:50:17] NOTICE[6342]: chan_sip.c:28003 handle_request_register: Registration from 'sip:80030@109.188.136.85' failed for '217.66.159.103:37230' - Wrong password
Scheduling destruction of SIP dialog 'aThNjQS3Lz' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:5.200.43.195:5066 --->
INVITE sip:8002@109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.UikGlUirJ;rport
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: sip:8002@109.188.136.85
CSeq: 20 INVITE
Call-ID: NGAJnsn50x
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 308
Contact: <sip:8005@5.200.43.195:5066>;+sip.instance="<urn:uuid:bc2d01b7-91b4-4bd9-8108-ed9a7b46e94c>"
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)

v=0
o=8005 2909 3959 IN IP4 192.168.10.151
s=Talk
c=IN IP4 192.168.10.151
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
<------------->
--- (13 headers 11 lines) ---
Sending to 5.200.43.195:5066 (NAT)
Sending to 5.200.43.195:5066 (NAT)
Using INVITE request as basis request - NGAJnsn50x
Found peer '8005' for '8005' from 5.200.43.195:5066

<--- Reliably Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.UikGlUirJ;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: sip:8002@109.188.136.85;tag=as57d9b065
Call-ID: NGAJnsn50x
CSeq: 20 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79293034"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NGAJnsn50x' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:5.200.43.195:5066 --->
ACK sip:8002@109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.UikGlUirJ;rport
Call-ID: NGAJnsn50x
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: <sip:8002@109.188.136.85>;tag=as57d9b065
Contact: <sip:8005@5.200.43.195:5066>;+sip.instance="<urn:uuid:bc2d01b7-91b4-4bd9-8108-ed9a7b46e94c>"
Max-Forwards: 70
CSeq: 20 ACK

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

<--- SIP read from UDP:5.200.43.195:5066 --->
INVITE sip:8002@109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.YORLwMupU;rport
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: sip:8002@109.188.136.85
CSeq: 21 INVITE
Call-ID: NGAJnsn50x
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 308
Contact: <sip:8005@5.200.43.195:5066>;+sip.instance="<urn:uuid:bc2d01b7-91b4-4bd9-8108-ed9a7b46e94c>"
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Authorization: Digest realm="asterisk", nonce="79293034", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85", response="e4d8ca2ab68e5452d32920eb21920888"

v=0
o=8005 2909 3959 IN IP4 192.168.10.151
s=Talk
c=IN IP4 192.168.10.151
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
<------------->
--- (14 headers 11 lines) ---
Sending to 5.200.43.195:5066 (NAT)
Using INVITE request as basis request - NGAJnsn50x
Found peer '8005' for '8005' from 5.200.43.195:5066
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found video description format H264 for ID 96
Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer - audio=(ulaw|alaw)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.10.151:7078
Peer video RTP is at port 192.168.10.151:9070
Peer doesn't provide T.140
Looking for 8002 in legenda (domain 109.188.136.85)
list_route: hop: <sip:8005@5.200.43.195:5066>

<--- Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.YORLwMupU;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: sip:8002@109.188.136.85
Call-ID: NGAJnsn50x
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8002@109.188.136.85:5060>
Content-Length: 0


<------------>
    -- Executing [8002@legenda:1] Dial("SIP/8005-00000014", "SIP/8002")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 14314
Video is at 192.168.111.111:19168
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding video codec 200002 (h263) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7d80648b;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 05 Oct 2015 18:50:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 428

v=0
o=root 1186334941 1186334941 IN IP4 192.168.111.111
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.111
b=CT:2048
t=0 0
m=audio 14314 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 19168 RTP/AVP 96 34
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---
    -- Called SIP/8002

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7d80648b;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 102 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7d80648b;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 102 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
MaxRingingTime: 45
MaxConnectingTime: 120
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:8002@192.168.111.177:5060>
    -- SIP/8002-00000015 is ringing

<--- Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.YORLwMupU;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: sip:8002@109.188.136.85;tag=as42109e6e
Call-ID: NGAJnsn50x
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8002@109.188.136.85:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK7d80648b;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 102 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
Found RTP video format 96
Found video description format H264 for ID 96
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.111.177:15000
Peer video RTP is at port 192.168.111.177:15002
Peer doesn't provide T.140
list_route: hop: <sip:8002@192.168.111.177:5060>
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK2565bc11;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
    -- SIP/8002-00000015 answered SIP/8005-00000014
Audio is at 11218
Video is at 109.188.136.85:18604
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100004 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.YORLwMupU;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: sip:8002@109.188.136.85;tag=as42109e6e
Call-ID: NGAJnsn50x
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8002@109.188.136.85:5060>
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 481211304 481211304 IN IP4 109.188.136.85
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 109.188.136.85
b=CT:2048
t=0 0
m=audio 11218 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 18604 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

<------------>
    -- Remotely bridging SIP/8005-00000014 and SIP/8002-00000015
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 14314
Video is at 192.168.10.151:9070
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK0fb97a8b;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 1186334941 1186334942 IN IP4 192.168.10.151
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.10.151
b=CT:2048
t=0 0
m=audio 7078 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK0fb97a8b;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 103 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #1 (NAT) to 5.200.43.195:5066:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.YORLwMupU;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: sip:8002@109.188.136.85;tag=as42109e6e
Call-ID: NGAJnsn50x
CSeq: 21 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8002@109.188.136.85:5060>
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 481211304 481211304 IN IP4 109.188.136.85
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 109.188.136.85
b=CT:2048
t=0 0
m=audio 11218 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 18604 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:5.200.43.195:5066 --->
ACK sip:8002@109.188.136.85:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;rport;branch=z9hG4bK.CjyJBioP8
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: <sip:8002@109.188.136.85>;tag=as42109e6e
CSeq: 21 ACK
Call-ID: NGAJnsn50x
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="79293034", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85", response="e4d8ca2ab68e5452d32920eb21920888"

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:8005@5.200.43.195:5066> for address/port to send to
set_destination: set destination to 5.200.43.195:5066
Audio is at 11218
Video is at 192.168.111.177:15002
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 5.200.43.195:5066:
INVITE sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK558d4d2e;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as42109e6e
To: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: NGAJnsn50x
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 481211304 481211305 IN IP4 192.168.111.177
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.177
b=CT:2048
t=0 0
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 15002 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---
       > 0x7f27040227f0 -- Probation passed - setting RTP source address to 192.168.111.177:15000

<--- SIP read from UDP:217.66.159.103:8539 --->
REGISTER sip:109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 217.66.159.103:8539;branch=z9hG4bK-524287-1---8fca0648f47e357d
Max-Forwards: 70
Contact: <sip:8003@217.66.159.103:8539;rinstance=36c7a51106e71528>
To: "8003"<sip:8003@109.188.136.85>
From: "8003"<sip:8003@109.188.136.85>;tag=0a34a45e
Call-ID: 126943OWQ0ODVmZGMwOTYyNGYwODQ0YzMxYWQ1MDRmZWE5ODI
CSeq: 49 REGISTER
Expires: 90
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, UPDATE, PRACK, OPTIONS, SUBSCRIBE, MESSAGE
User-Agent: Bria Android release 3.4.3 stamp 81731
Authorization: Digest username="8003",realm="asterisk",nonce="44cf0dcc",uri="sip:109.188.136.85",response="4a19be68fbe2d959733f83186cc96b56",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 217.66.159.103:8539 (NAT)
Sending to 217.66.159.103:8539 (NAT)

<--- Transmitting (NAT) to 217.66.159.103:8539 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.66.159.103:8539;branch=z9hG4bK-524287-1---8fca0648f47e357d;received=217.66.159.103;rport=8539
From: "8003"<sip:8003@109.188.136.85>;tag=0a34a45e
To: "8003"<sip:8003@109.188.136.85>;tag=as717fdd0b
Call-ID: 126943OWQ0ODVmZGMwOTYyNGYwODQ0YzMxYWQ1MDRmZWE5ODI
CSeq: 49 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11fc4630"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '126943OWQ0ODVmZGMwOTYyNGYwODQ0YzMxYWQ1MDRmZWE5ODI' in 32000 ms (Method: REGISTER)
       > 0x7f27040030c0 -- Probation passed - setting RTP source address to 192.168.111.177:15002
Retransmitting #1 (NAT) to 5.200.43.195:5066:
INVITE sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK558d4d2e;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as42109e6e
To: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: NGAJnsn50x
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 481211304 481211305 IN IP4 192.168.111.177
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.177
b=CT:2048
t=0 0
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 15002 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:217.66.159.103:8539 --->
REGISTER sip:109.188.136.85 SIP/2.0
Via: SIP/2.0/UDP 217.66.159.103:8539;branch=z9hG4bK-524287-1---be6253771904cf70
Max-Forwards: 70
Contact: <sip:8003@217.66.159.103:8539;rinstance=36c7a51106e71528>
To: "8003"<sip:8003@109.188.136.85>
From: "8003"<sip:8003@109.188.136.85>;tag=0a34a45e
Call-ID: 126943OWQ0ODVmZGMwOTYyNGYwODQ0YzMxYWQ1MDRmZWE5ODI
CSeq: 50 REGISTER
Expires: 90
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, UPDATE, PRACK, OPTIONS, SUBSCRIBE, MESSAGE
User-Agent: Bria Android release 3.4.3 stamp 81731
Authorization: Digest username="8003",realm="asterisk",nonce="11fc4630",uri="sip:109.188.136.85",response="c1952643cac0bbd117ccad80cbcf2424",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 217.66.159.103:8539 (NAT)
Reliably Transmitting (NAT) to 217.66.159.103:8539:
OPTIONS sip:8003@217.66.159.103:8539;rinstance=36c7a51106e71528 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK08763ded;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@109.188.136.85>;tag=as63f4696a
To: <sip:8003@217.66.159.103:8539;rinstance=36c7a51106e71528>
Contact: <sip:asterisk@109.188.136.85:5060>
Call-ID: 69368e40566fc4ed5cadc36a08720073@109.188.136.85:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 05 Oct 2015 18:50:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 217.66.159.103:8539 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.66.159.103:8539;branch=z9hG4bK-524287-1---be6253771904cf70;received=217.66.159.103;rport=8539
From: "8003"<sip:8003@109.188.136.85>;tag=0a34a45e
To: "8003"<sip:8003@109.188.136.85>;tag=as717fdd0b
Call-ID: 126943OWQ0ODVmZGMwOTYyNGYwODQ0YzMxYWQ1MDRmZWE5ODI
CSeq: 50 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 90
Contact: <sip:8003@217.66.159.103:8539;rinstance=36c7a51106e71528>;expires=90
Date: Mon, 05 Oct 2015 18:50:17 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '126943OWQ0ODVmZGMwOTYyNGYwODQ0YzMxYWQ1MDRmZWE5ODI' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK0fb97a8b;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 103 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK6b62834a;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Retransmitting #2 (NAT) to 5.200.43.195:5066:
INVITE sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK558d4d2e;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as42109e6e
To: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: NGAJnsn50x
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 481211304 481211305 IN IP4 192.168.111.177
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.177
b=CT:2048
t=0 0
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 15002 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:217.66.159.103:8539 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK08763ded;rport=5060
Contact: <sip:10.153.220.38:37213>
To: <sip:8003@217.66.159.103:8539;rinstance=36c7a51106e71528>;tag=fc8b5d2e
From: "asterisk" <sip:asterisk@109.188.136.85>;tag=as63f4696a
Call-ID: 69368e40566fc4ed5cadc36a08720073@109.188.136.85:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, UPDATE, PRACK, OPTIONS, SUBSCRIBE, MESSAGE
User-Agent: Bria Android release 3.4.3 stamp 81731
Allow-Events: talk, hold
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '69368e40566fc4ed5cadc36a08720073@109.188.136.85:5060' Method: OPTIONS
Retransmitting #3 (NAT) to 5.200.43.195:5066:
INVITE sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK558d4d2e;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as42109e6e
To: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: NGAJnsn50x
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 481211304 481211305 IN IP4 192.168.111.177
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.177
b=CT:2048
t=0 0
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 15002 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---
Really destroying SIP dialog 'V0ZNvFHp-nn2Tt0aaXc7Ow..' Method: REGISTER

<--- SIP read from UDP:5.200.43.195:5066 --->
ACK sip:8002@109.188.136.85:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.CjyJBioP8;rport
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: <sip:8002@109.188.136.85>;tag=as42109e6e
CSeq: 21 ACK
Call-ID: NGAJnsn50x
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="79293034", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85", response="e4d8ca2ab68e5452d32920eb21920888"

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

<--- SIP read from UDP:5.200.43.195:5066 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK558d4d2e;rport
From: <sip:8002@109.188.136.85>;tag=as42109e6e
To: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
Call-ID: NGAJnsn50x
CSeq: 102 INVITE

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

<--- SIP read from UDP:5.200.43.195:5066 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK558d4d2e;rport
From: <sip:8002@109.188.136.85>;tag=as42109e6e
To: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
Call-ID: NGAJnsn50x
CSeq: 102 INVITE
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: <sip:8005@5.200.43.195:5066>;+sip.instance="<urn:uuid:bc2d01b7-91b4-4bd9-8108-ed9a7b46e94c>"
Content-Type: application/sdp
Content-Length: 193

v=0
o=8005 2909 3961 IN IP4 192.168.10.151
s=Talk
c=IN IP4 192.168.10.151
t=0 0
m=audio 7078 RTP/AVP 0
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found RTP video format 96
Found video description format H264 for ID 96
Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.10.151:7078
Peer video RTP is at port 192.168.10.151:9070
Peer doesn't provide T.140
set_destination: Parsing <sip:8005@5.200.43.195:5066> for address/port to send to
set_destination: set destination to 5.200.43.195:5066
Transmitting (NAT) to 5.200.43.195:5066:
ACK sip:8005@5.200.43.195:5066 SIP/2.0
Via: SIP/2.0/UDP 109.188.136.85:5060;branch=z9hG4bK7eea0024;rport
Max-Forwards: 70
From: sip:8002@109.188.136.85;tag=as42109e6e
To: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
Contact: <sip:8002@109.188.136.85:5060>
Call-ID: NGAJnsn50x
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 14314
Video is at 192.168.10.151:9070
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK08c1afca;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 1186334941 1186334943 IN IP4 192.168.10.151
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.10.151
b=CT:2048
t=0 0
m=audio 7078 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 9070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK08c1afca;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 104 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK08c1afca;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 104 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK486f1c19;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---

<--- SIP read from UDP:217.66.159.103:37230 --->


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

<--- SIP read from UDP:5.200.43.195:5066 --->


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

<--- SIP read from UDP:5.200.43.195:5066 --->
BYE sip:8002@109.188.136.85:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.ApYZVb5xc;rport
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: <sip:8002@109.188.136.85>;tag=as42109e6e
CSeq: 22 BYE
Call-ID: NGAJnsn50x
Max-Forwards: 70
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Authorization: Digest realm="asterisk", nonce="79293034", algorithm=MD5, username="8005", uri="sip:8002@109.188.136.85:5060", response="975578f9d5ec218cf5809338d031db6a"

<------------->
--- (9 headers 0 lines) ---
Sending to 5.200.43.195:5066 (NAT)
Scheduling destruction of SIP dialog 'NGAJnsn50x' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 5.200.43.195:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.151:5066;branch=z9hG4bK.ApYZVb5xc;received=5.200.43.195;rport=5066
From: <sip:8005@109.188.136.85>;tag=~yHyQRuWS
To: <sip:8002@109.188.136.85>;tag=as42109e6e
Call-ID: NGAJnsn50x
CSeq: 22 BYE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Audio is at 14314
Video is at 192.168.111.111:19168
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Reliably Transmitting (NAT) to 192.168.111.177:5060:
INVITE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK5f95d322;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 105 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 366

v=0
o=root 1186334941 1186334944 IN IP4 192.168.111.111
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.111
b=CT:2048
t=0 0
m=audio 14314 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
m=video 19168 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=sendrecv

---

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK5f95d322;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 105 INVITE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

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

<--- SIP read from UDP:217.66.159.103:8539 --->


<------------->
Scheduling destruction of SIP dialog '646645c42c7be10209137f683f168165@192.168.111.111:5060' in 6400 ms (Method: INVITE)
  == Spawn extension (legenda, 8002, 1) exited non-zero on 'SIP/8005-00000014'

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK5f95d322;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 105 INVITE
Contact: <sip:8002@192.168.111.177:5060>
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Type: application/sdp
Content-Length: 210

v=0
o=0 0 0 IN IP4 192.168.111.177
s=Dahua VT 1.5
c=IN IP4 192.168.111.177
t=0 0
m=video 15002 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 15000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Transmitting (NAT) to 192.168.111.177:5060:
ACK sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK55e0b47d;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Contact: <sip:8005@192.168.111.111:5060>
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 105 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
set_destination: Parsing <sip:8002@192.168.111.177:5060> for address/port to send to
set_destination: set destination to 192.168.111.177:5060
Reliably Transmitting (NAT) to 192.168.111.177:5060:
BYE sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3591b91b;rport
Max-Forwards: 70
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 106 BYE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog '646645c42c7be10209137f683f168165@192.168.111.111:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK3591b91b;rport=5060
From: <sip:8005@192.168.111.111>;tag=as582d7d63
To: <sip:8002@192.168.111.177:5060>;tag=52182251
Call-ID: 646645c42c7be10209137f683f168165@192.168.111.111:5060
CSeq: 106 BYE
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '646645c42c7be10209137f683f168165@192.168.111.111:5060' Method: INVITE

<--- SIP read from UDP:192.168.111.177:5060 --->
REGISTER sip:8002@192.168.111.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.177:5060;rport;branch=z9hG4bK136969976
From: <sip:8002@192.168.111.111:5060>;tag=1326076078
To: <sip:8002@192.168.111.111:5060>
Call-ID: 504371200@192.168.111.177
CSeq: 1109 REGISTER
Contact: <sip:8002@192.168.111.177:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Expires: 3600
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.111.177:5060 (NAT)
Sending to 192.168.111.177:5060 (NAT)

<--- Transmitting (NAT) to 192.168.111.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.177:5060;branch=z9hG4bK136969976;received=192.168.111.177;rport=5060
From: <sip:8002@192.168.111.111:5060>;tag=1326076078
To: <sip:8002@192.168.111.111:5060>;tag=as4e73ddae
Call-ID: 504371200@192.168.111.177
CSeq: 1109 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3576e8b5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '504371200@192.168.111.177' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.111.177:5060 --->
REGISTER sip:8002@192.168.111.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.177:5060;rport;branch=z9hG4bK142591642
From: <sip:8002@192.168.111.111:5060>;tag=1326076078
To: <sip:8002@192.168.111.111:5060>
Call-ID: 504371200@192.168.111.177
CSeq: 1110 REGISTER
Contact: <sip:8002@192.168.111.177:5060>
Authorization: Digest username="8002", realm="asterisk", nonce="3576e8b5", uri="sip:8002@192.168.111.111:5060", response="b53c6814233237d331d70e3eb481235e", algorithm=MD5
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Expires: 3600
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.111.177:5060 (NAT)
Reliably Transmitting (NAT) to 192.168.111.177:5060:
OPTIONS sip:8002@192.168.111.177:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK194bbe1f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as76a23497
To: <sip:8002@192.168.111.177:5060>
Contact: <sip:asterisk@192.168.111.111:5060>
Call-ID: 79cb074e43b66db8649a2ce631a4bd32@192.168.111.111:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 05 Oct 2015 18:50:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.177:5060;branch=z9hG4bK142591642;received=192.168.111.177;rport=5060
From: <sip:8002@192.168.111.111:5060>;tag=1326076078
To: <sip:8002@192.168.111.111:5060>;tag=as4e73ddae
Call-ID: 504371200@192.168.111.177
CSeq: 1110 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:8002@192.168.111.177:5060>;expires=3600
Date: Mon, 05 Oct 2015 18:50:29 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '504371200@192.168.111.177' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.111.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK194bbe1f;rport=5060
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as76a23497
To: <sip:8002@192.168.111.177:5060>;tag=482722766
Call-ID: 79cb074e43b66db8649a2ce631a4bd32@192.168.111.111:5060
CSeq: 102 OPTIONS
User-Agent: Dahua UAC/3.0 VTO1210C-X V1.0.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '79cb074e43b66db8649a2ce631a4bd32@192.168.111.111:5060' Method: OPTIONS
asterisk*CLI>

Re: Повторный вызов не проходит...

Добавлено: 06 окт 2015, 01:06
Zavr2008
странно скачущие настроки NAT - в астере четко externip и localnet прописан? у роутика злобный чёрный кащей абама то биш SIP ALG - отключен?

Re: Повторный вызов не проходит...

Добавлено: 06 окт 2015, 09:23
ded
Это - понятно, Linphone

Код: Выделить всё

User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
А это - кто?

Код: Выделить всё

User-Agent: Dahua UAC/3.0 VTH1510 V1.100.0.0
Не Dahua ли у вас всяких видеопиров? Каждый со своими причудами?
При удачном звонке все склеивается по h264, а при неудачном ясно видно, что пытается ответить по h263
Adding video codec 200002 (h263) to SDP
и это не получится. Видеокодеки не транскодятся, в отличии от аудио, то есть на аппаратах и в настройках пиров надо оставить только h264

Re: Повторный вызов не проходит...

Добавлено: 06 окт 2015, 19:10
ramen
Zavr2008 писал(а):странно скачущие настроки NAT - в астере четко externip и localnet прописан? у роутика злобный чёрный кащей абама то биш SIP ALG - отключен?
Да прописано все жестко, SIP ALG на Кинетике с первой версией прошивки и вовсе отсутствует, роутер конечно попахивает... вечно по модему виснет (раз в два дня, но для 4G свистков - это норма)
Это - понятно, Linphone
К сожалению не использовать его не можем, в чем прикол на другом сервере норм, с зеркально настроенными конфигами, не понят. только вот откуда такое поведение, не помогает рестарт астера, перерегистрация обоих устройств....

а при неудачном ясно видно, что пытается ответить по h263 Adding video codec 200002 (h263) to SDP
Как раз таки Adding video codec 200004 (h264) to SDP в первом и втором случае, складывается ulaw|h264
Проблема скорее в NAT, но что может быть не, так. если остальные софтфоны, даже безсервисные отрабатывают как положено...

Re: Повторный вызов не проходит...

Добавлено: 06 окт 2015, 21:11
ded
Так и не ответили про Dahua....
Так вижу,что предлагается не просто h264, а с опциями profile-level-id=42801F и a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
которые ваш Dahua ниhua непонимэ.

Код: Выделить всё

v=0
o=root 1186334941 1186334941 IN IP4 192.168.111.111
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.111.111
b=CT:2048
t=0 0
m=audio 14314 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 19168 RTP/AVP 96 34
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
Проверить свою версию про NAT можете через rtp set debug on при удачном и неудачном звонке.

Re: Повторный вызов не проходит...

Добавлено: 08 окт 2015, 17:12
ramen
Проблема решилась, таилась в роутере Zyxel, решение: в помойку эту дичь и установки Асуса))
Часовой пояс: UTC+03:00
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