[asterisk-users] Fwd: early media (video)
Fronc Hias
fronc.hias at gmail.com
Thu May 8 06:19:38 CDT 2014
part #2
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<--- SIP read from UDP:10.10.1.144:5060 --->
INVITE sip:306 at 10.10.1.201 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;rport
From: <sip:301 at 10.10.1.201>;tag=XY9jSoWko
To: sip:306 at 10.10.1.201
CSeq: 20 INVITE
Call-ID: RYw7fDsLDF
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 562
Contact: <sip:301 at 10.10.1.144>;+sip.instance="<urn:uuid:c03a376f-325c-4764-a9c0-827fb2a23a79>"
User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
v=0
o=301 4060 2139 IN IP4 192.168.220.16
s=Talk
c=IN IP4 192.168.220.16
t=0 0
m=audio 7078 RTP/AVP 9 124 111 110 0 8 101
a=rtpmap:124 opus/48000
a=fmtp:124 useinbandfec=1; usedtx=1
a=rtpmap:111 speex/16000
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 9078 RTP/AVP 102 98 103 99
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
a=rtpmap:103 VP8/90000
a=rtpmap:99 MP4V-ES/90000
a=fmtp:99 profile-level-id=3
<------------->
--- (13 headers 22 lines) ---
Sending to 10.10.1.144:5060 (no NAT)
Sending to 10.10.1.144:5060 (no NAT)
Using INVITE request as basis request - RYw7fDsLDF
Found peer '301' for '301' from 10.10.1.144:5060
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 124
Found RTP audio format 111
Found RTP audio format 110
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format opus for ID 124
Found audio description format speex for ID 111
Found audio description format speex for ID 110
Found audio description format telephone-event for ID 101
Found RTP video format 102
Found RTP video format 98
Found RTP video format 103
Found RTP video format 99
Found video description format H264 for ID 102
Found video description format H263-1998 for ID 98
Found video description format VP8 for ID 103
Found video description format MP4V-ES for ID 99
Capabilities: us - (ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|speex|speex16|g722|opus)/video=(h263p|h264|mpeg4|vp8)/text=(nothing), combined - (ulaw|alaw|g722|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.220.16:7078
Peer video RTP is at port 192.168.220.16:9078
Peer doesn't provide T.140
Looking for 306 in LocalSets (domain 10.10.1.201)
list_route: route/path hop: <sip:301 at 10.10.1.144>
<--- Transmitting (NAT) to 10.10.1.144:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=XY9jSoWko
To: sip:306 at 10.10.1.201
Call-ID: RYw7fDsLDF
CSeq: 20 INVITE
Server: Asterisk PBX 12.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:306 at 10.10.1.201:5060>
Content-Length: 0
<------------>
-- Executing [306 at LocalSets:1] NoOp("SIP/301-0000000a", "dial 306") in new stack
-- Executing [306 at LocalSets:3] Dial("SIP/301-0000000a", "SIP/306") in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Audio is at 14884
Video is at 10.10.1.201:12672
Adding codec 100012 (g722) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.10.1.145:5062:
INVITE sip:306 at 10.10.1.145:5062 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ed3b996;rport
Max-Forwards: 70
From: <sip:301 at 10.10.1.201>;tag=as7052b518
To: <sip:306 at 10.10.1.145:5062>
Contact: <sip:301 at 10.10.1.201:5060>
Call-ID: 236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:02:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 433
v=0
o=root 286204086 286204086 IN IP4 10.10.1.201
s=Asterisk PBX 12.2.0-rc1
c=IN IP4 10.10.1.201
b=CT:384
t=0 0
m=audio 14884 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 12672 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F
a=sendrecv
---
-- Called SIP/306
<--- SIP read from UDP:10.10.1.145:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ed3b996;rport=5060
From: <sip:301 at 10.10.1.201>;tag=as7052b518
To: <sip:306 at 10.10.1.145:5062>
Call-ID: 236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3175v2 1.0.1.55
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.10.1.145:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ed3b996;rport=5060
From: <sip:301 at 10.10.1.201>;tag=as7052b518
To: <sip:306 at 10.10.1.145:5062>;tag=647474019
Call-ID: 236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060
CSeq: 102 INVITE
Contact: <sip:306 at 10.10.1.145:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3175v2 1.0.1.55
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
list_route: route/path hop: <sip:306 at 10.10.1.145:5062>
-- SIP/306-0000000b is ringing
<--- Transmitting (NAT) to 10.10.1.144:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=XY9jSoWko
To: sip:306 at 10.10.1.201;tag=as1221478d
Call-ID: RYw7fDsLDF
CSeq: 20 INVITE
Server: Asterisk PBX 12.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:306 at 10.10.1.201:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.10.1.144:5060 --->
<------------->
<--- SIP read from UDP:10.10.1.145:5062 --->
<------------->
Reliably Transmitting (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as28d2335b
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 4b67070a5acb26a820bb912928c276ed at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #1 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as28d2335b
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 4b67070a5acb26a820bb912928c276ed at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.10.1.145:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ed3b996;rport=5060
From: <sip:301 at 10.10.1.201>;tag=as7052b518
To: <sip:306 at 10.10.1.145:5062>;tag=647474019
Call-ID: 236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060
CSeq: 102 INVITE
Contact: <sip:306 at 10.10.1.145:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3175v2 1.0.1.55
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 447
v=0
o=306 8000 8000 IN IP4 10.10.1.145
s=SIP Call
c=IN IP4 10.10.1.145
t=0 0
m=audio 45382 RTP/AVP 9 0 8 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
m=video 55076 RTP/AVP 99
b=AS:320
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==; max-br=320
<------------->
--- (12 headers 19 lines) ---
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - (ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|g722)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g722|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.1.145:45382
Peer video RTP is at port 10.10.1.145:55076
Peer doesn't provide T.140
list_route: route/path hop: <sip:306 at 10.10.1.145:5062>
Transmitting (NAT) to 10.10.1.145:5062:
ACK sip:306 at 10.10.1.145:5062 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK4ff8e080;rport
Max-Forwards: 70
From: <sip:301 at 10.10.1.201>;tag=as7052b518
To: <sip:306 at 10.10.1.145:5062>;tag=647474019
Contact: <sip:301 at 10.10.1.201:5060>
Call-ID: 236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.2.0-rc1
Content-Length: 0
---
-- SIP/306-0000000b answered SIP/301-0000000a
Audio is at 10078
Video is at 10.10.1.201:16536
Adding codec 100012 (g722) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.10.1.144:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=XY9jSoWko
To: sip:306 at 10.10.1.201;tag=as1221478d
Call-ID: RYw7fDsLDF
CSeq: 20 INVITE
Server: Asterisk PBX 12.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:306 at 10.10.1.201:5060>
Content-Type: application/sdp
Content-Length: 438
v=0
o=root 1048282658 1048282658 IN IP4 10.10.1.201
s=Asterisk PBX 12.2.0-rc1
c=IN IP4 10.10.1.201
b=CT:384
t=0 0
m=audio 10078 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 16536 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=sendrecv
<------------>
-- Channel SIP/301-0000000a joined 'simple_bridge' basic-bridge <057aca53-4ef0-4568-b312-ff6e032da9d3>
<--- SIP read from UDP:10.10.1.144:5060 --->
ACK sip:306 at 10.10.1.201:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.144:5060;rport;branch=z9hG4bK.KNApcdgKs
From: <sip:301 at 10.10.1.201>;tag=XY9jSoWko
To: <sip:306 at 10.10.1.201>;tag=as1221478d
CSeq: 20 ACK
Call-ID: RYw7fDsLDF
Max-Forwards: 70
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:10.10.1.144:5060 --->
PUBLISH sip:301 at 10.10.1.201 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.AZJ6xMUEY;rport
From: <sip:301 at 10.10.1.201>;tag=TOdTkgRTc
To: sip:301 at 10.10.1.201
CSeq: 27 PUBLISH
Call-ID: LErqEXzaBt
Max-Forwards: 70
Supported: replaces, outbound
Event: presence
Expires: 3600
User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
Content-Type: application/pidf+xml
Content-Length: 514
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:301 at 10.10.1.201" xmlns="urn:ietf:params:xml:ns:pidf"><tuple id="yhk46v"><status><basic>open</basic></status><contact priority="0.8">sip:301 at 10.10.1.201</contact><timestamp>2014-05-07T12:03:06Z</timestamp></tuple><dm:person id="8rrehq"><rpid:activities><rpid:on-the-phone/></rpid:activities><timestamp>2014-05-07T12:03:06Z</timestamp></dm:person></presence>
<------------->
--- (13 headers 2 lines) ---
Sending to 10.10.1.144:5060 (no NAT)
<--- Transmitting (no NAT) to 10.10.1.144:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.AZJ6xMUEY;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=TOdTkgRTc
To: sip:301 at 10.10.1.201;tag=as54e8d3b1
Call-ID: LErqEXzaBt
CSeq: 27 PUBLISH
Server: Asterisk PBX 12.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog 'LErqEXzaBt' Method: PUBLISH
-- Channel SIP/306-0000000b joined 'simple_bridge' basic-bridge <057aca53-4ef0-4568-b312-ff6e032da9d3>
> Bridge 057aca53-4ef0-4568-b312-ff6e032da9d3: switching from simple_bridge technology to native_rtp
> 0x7f84dc00df30 -- Probation passed - setting RTP source address to 10.10.1.144:7078
> 0x7f84dc00df30 -- Probation passed - setting RTP source address to 10.10.1.144:7078
Retransmitting #2 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as28d2335b
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 4b67070a5acb26a820bb912928c276ed at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
> 0x7f84dc010ef0 -- Probation passed - setting RTP source address to 10.10.1.144:9078
> 0x7f84e8004660 -- Probation passed - setting RTP source address to 10.10.1.145:45382
> 0x7f84dc010ef0 -- Probation passed - setting RTP source address to 10.10.1.144:9078
> 0x7f84e800c180 -- Probation passed - setting RTP source address to 10.10.1.145:55076
Retransmitting #3 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as28d2335b
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 4b67070a5acb26a820bb912928c276ed at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #4 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK5ee0f102;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as28d2335b
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 4b67070a5acb26a820bb912928c276ed at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog '4b67070a5acb26a820bb912928c276ed at 10.10.1.201:5060' Method: OPTIONS
<--- SIP read from UDP:10.10.1.145:5062 --->
BYE sip:301 at 10.10.1.201:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.145:5062;branch=z9hG4bK651537380;rport
From: <sip:306 at 10.10.1.145:5062>;tag=647474019
To: <sip:301 at 10.10.1.201>;tag=as7052b518
Call-ID: 236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060
CSeq: 103 BYE
Contact: <sip:306 at 10.10.1.145:5062>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3175v2 1.0.1.55
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 10.10.1.145:5062 (NAT)
Scheduling destruction of SIP dialog '236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 10.10.1.145:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.145:5062;branch=z9hG4bK651537380;received=10.10.1.145;rport=5062
From: <sip:306 at 10.10.1.145:5062>;tag=647474019
To: <sip:301 at 10.10.1.201>;tag=as7052b518
Call-ID: 236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060
CSeq: 103 BYE
Server: Asterisk PBX 12.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
-- Channel SIP/306-0000000b left 'native_rtp' basic-bridge <057aca53-4ef0-4568-b312-ff6e032da9d3>
-- Channel SIP/301-0000000a left 'native_rtp' basic-bridge <057aca53-4ef0-4568-b312-ff6e032da9d3>
== Spawn extension (LocalSets, 306, 3) exited non-zero on 'SIP/301-0000000a'
Scheduling destruction of SIP dialog 'RYw7fDsLDF' in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 10.10.1.144:5060:
BYE sip:301 at 10.10.1.144 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK30d73897;rport
Max-Forwards: 70
From: sip:306 at 10.10.1.201;tag=as1221478d
To: <sip:301 at 10.10.1.201>;tag=XY9jSoWko
Call-ID: RYw7fDsLDF
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.2.0-rc1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:10.10.1.144:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK30d73897;rport
From: <sip:306 at 10.10.1.201>;tag=as1221478d
To: <sip:301 at 10.10.1.201>;tag=XY9jSoWko
Call-ID: RYw7fDsLDF
CSeq: 102 BYE
User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
Supported: replaces, outbound
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'RYw7fDsLDF' Method: ACK
<--- SIP read from UDP:10.10.1.144:5060 --->
PUBLISH sip:301 at 10.10.1.201 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.uy~NP6rxW;rport
From: <sip:301 at 10.10.1.201>;tag=TOdTkgRTc
To: sip:301 at 10.10.1.201
CSeq: 28 PUBLISH
Call-ID: LErqEXzaBt
Max-Forwards: 70
Supported: replaces, outbound
Event: presence
Expires: 3600
User-Agent: Linphone/3.7.0 (belle-sip/1.3.0)
Content-Type: application/pidf+xml
Content-Length: 381
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:301 at 10.10.1.201" xmlns="urn:ietf:params:xml:ns:pidf"><tuple id="nqcpwx"><status><basic>open</basic></status><contact priority="0.8">sip:301 at 10.10.1.201</contact><timestamp>2014-05-07T11:55:33Z</timestamp></tuple></presence>
<------------->
--- (13 headers 2 lines) ---
Sending to 10.10.1.144:5060 (no NAT)
<--- Transmitting (no NAT) to 10.10.1.144:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.uy~NP6rxW;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=TOdTkgRTc
To: sip:301 at 10.10.1.201;tag=as0a7851b1
Call-ID: LErqEXzaBt
CSeq: 28 PUBLISH
Server: Asterisk PBX 12.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog 'LErqEXzaBt' Method: PUBLISH
<--- SIP read from UDP:10.10.1.144:5060 --->
<------------->
Reliably Transmitting (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2e98e406
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 7a7bd73b4612f49f773990cf2b9a6a95 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #1 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2e98e406
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 7a7bd73b4612f49f773990cf2b9a6a95 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog '236cc12c00f94461054ac397244eda2a at 10.10.1.201:5060' Method: BYE
Retransmitting #2 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2e98e406
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 7a7bd73b4612f49f773990cf2b9a6a95 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #3 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2e98e406
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 7a7bd73b4612f49f773990cf2b9a6a95 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #4 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK2952aad8;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2e98e406
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 7a7bd73b4612f49f773990cf2b9a6a95 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog '7a7bd73b4612f49f773990cf2b9a6a95 at 10.10.1.201:5060' Method: OPTIONS
<--- SIP read from UDP:10.10.1.144:5060 --->
<------------->
<--- SIP read from UDP:10.10.1.145:5062 --->
<------------->
Reliably Transmitting (NAT) to 10.10.1.145:5062:
OPTIONS sip:306 at 10.10.1.145:5062 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK1bab16d3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as0733973c
To: <sip:306 at 10.10.1.145:5062>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 0dd5ca4d05b45112264ddfad376ae061 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.10.1.145:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK1bab16d3;rport=5060
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as0733973c
To: <sip:306 at 10.10.1.145:5062>;tag=1217888556
Call-ID: 0dd5ca4d05b45112264ddfad376ae061 at 10.10.1.201:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3175v2 1.0.1.55
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0dd5ca4d05b45112264ddfad376ae061 at 10.10.1.201:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2d557dca
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 01452f22339bb14b1c60e554620b0d59 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #1 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2d557dca
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 01452f22339bb14b1c60e554620b0d59 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Reliably Transmitting (NAT) to 10.10.1.144:5060:
OPTIONS sip:301 at 10.10.1.144 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK716d9273;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as7d31d5fe
To: <sip:301 at 10.10.1.144>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 1af7e5c73bb66bc80a29ec81089a3d77 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.10.1.144:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK716d9273;rport
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as7d31d5fe
To: <sip:301 at 10.10.1.144>;tag=slu9L
Call-ID: 1af7e5c73bb66bc80a29ec81089a3d77 at 10.10.1.201:5060
CSeq: 102 OPTIONS
<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '1af7e5c73bb66bc80a29ec81089a3d77 at 10.10.1.201:5060' Method: OPTIONS
<--- SIP read from UDP:10.10.1.144:5060 --->
<------------->
Retransmitting #2 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2d557dca
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 01452f22339bb14b1c60e554620b0d59 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #3 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2d557dca
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 01452f22339bb14b1c60e554620b0d59 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #4 (NAT) to 10.10.1.146:34602:
OPTIONS sip:307 at 10.10.1.146:34602 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK69e9074b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as2d557dca
To: <sip:307 at 10.10.1.146:34602>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 01452f22339bb14b1c60e554620b0d59 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:03:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog '01452f22339bb14b1c60e554620b0d59 at 10.10.1.201:5060' Method: OPTIONS
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