[asterisk-users] Fwd: early media (video)
Fronc Hias
fronc.hias at gmail.com
Thu May 8 06:18:16 CDT 2014
ok then... sorry it took so long, but I already posted it yesterday but it
was rejected by the list's moderator as it exceeded 40k in size :(
so i'll resend it now: part #1 is this, part #2 will only contain the
attachment with working "direct"-Video (not early media) for comparison...
---
1) i tried with asterisk 11.6... no change. i.e. same behaviour as with
asterisk 12.2 --> no preview video
(i verified with wireshark that - as with asterisk 12.2 - the h264 data is
sent from the caller to asterisk, but isn't forwarded to the callee... only
the audio (g722) is passed on)
2) setup:
the caller is 301 (linphone on windows) calling 306 (Grandstream 3175v2
with preview enabled)
2.1) sip.conf:
[general]
context=unauthenticated
allowguest=yes
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
callcounter=yes
textsupport=yes
accept_outofcall_messages=yes
outofcall_message_context=messages-sip
auth_message_requests=no
insecure=invite
directmedia=no
prematuremedia=yes
progressinband=never
[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw
allow=alaw
videosupport=yes
allow=h264
# define some devices
[301](office-phone)
secret=pwd301
[306](office-phone)
secret=pwd306
2.2) extensions.conf
...
[LocalSets]
...
exten => 306,hint,SIP/306
exten => 306,1,NoOp(dial 306)
same => n,Dial(SIP/306)
...
3) debug-logs:
I used "core set debug 9" and "core set verbose 9" and "sip set debug on"
i attached two files:
3.1) "asterisk-debug-earlymedia.txt" in which I dial 306 from 301 and after
some (5?) seconds i press "preview" on the grandstream. after that, the
grandstream only allows me to choose beween "Accept Audio" and "Reject" -
no "Accept Video" as I would expect... another 5 seconds later i hang up.
3.2) "asterisk-debug-videodirect.txt" here I push the "Accept Video" button
instead of the "preview" button; here the video works!
thanks a lot for your support!
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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.~izIMUo0u;rport
From: <sip:301 at 10.10.1.201>;tag=WSFA2GLK9
To: sip:306 at 10.10.1.201
CSeq: 20 INVITE
Call-ID: 64TsxOvORf
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 4066 3784 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 - 64TsxOvORf
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.~izIMUo0u;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=WSFA2GLK9
To: sip:306 at 10.10.1.201
Call-ID: 64TsxOvORf
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/301-00000008", "dial 306") in new stack
-- Executing [306 at LocalSets:3] Dial("SIP/301-00000008", "SIP/306") in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Audio is at 13998
Video is at 10.10.1.201:19236
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=z9hG4bK0adf25b7;rport
Max-Forwards: 70
From: <sip:301 at 10.10.1.201>;tag=as2e0a7bb2
To: <sip:306 at 10.10.1.145:5062>
Contact: <sip:301 at 10.10.1.201:5060>
Call-ID: 3c7241696275b34c664111f736759783 at 10.10.1.201:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:01:20 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 763823955 763823955 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 13998 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 19236 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=z9hG4bK0adf25b7;rport=5060
From: <sip:301 at 10.10.1.201>;tag=as2e0a7bb2
To: <sip:306 at 10.10.1.145:5062>
Call-ID: 3c7241696275b34c664111f736759783 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=z9hG4bK0adf25b7;rport=5060
From: <sip:301 at 10.10.1.201>;tag=as2e0a7bb2
To: <sip:306 at 10.10.1.145:5062>;tag=177996708
Call-ID: 3c7241696275b34c664111f736759783 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/306-00000009 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.~izIMUo0u;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=WSFA2GLK9
To: sip:306 at 10.10.1.201;tag=as0ae1acfc
Call-ID: 64TsxOvORf
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 --->
<------------->
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=z9hG4bK4b77810b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as7bef7d1c
To: <sip:306 at 10.10.1.145:5062>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 28e06368307769e702a73e524062623a at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:01:25 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=z9hG4bK4b77810b;rport=5060
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as7bef7d1c
To: <sip:306 at 10.10.1.145:5062>;tag=637705810
Call-ID: 28e06368307769e702a73e524062623a 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 '28e06368307769e702a73e524062623a at 10.10.1.201:5060' Method: OPTIONS
<--- SIP read from UDP:10.10.1.145:5062 --->
<------------->
<--- SIP read from UDP:10.10.1.145:5062 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK0adf25b7;rport=5060
From: <sip:301 at 10.10.1.201>;tag=as2e0a7bb2
To: <sip:306 at 10.10.1.145:5062>;tag=177996708
Call-ID: 3c7241696275b34c664111f736759783 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 43870 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 31652 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) ---
list_route: route/path hop: <sip:306 at 10.10.1.145:5062>
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 10.10.1.145:43870
Peer video RTP is at port 10.10.1.145:31652
Peer doesn't provide T.140
-- SIP/306-00000009 is making progress passing it to SIP/301-00000008
Audio is at 16968
Video is at 10.10.1.201:16576
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
<--- Transmitting (NAT) to 10.10.1.144:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=WSFA2GLK9
To: sip:306 at 10.10.1.201;tag=as0ae1acfc
Call-ID: 64TsxOvORf
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: 436
v=0
o=root 120783996 120783996 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 16968 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 16576 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=sendrecv
<------------>
> 0x7f84dc026930 -- Probation passed - setting RTP source address to 10.10.1.144:9078
> 0x7f84dc026930 -- Probation passed - setting RTP source address to 10.10.1.144:9078
> 0x7f84dc023590 -- Probation passed - setting RTP source address to 10.10.1.144:7078
> 0x7f84dc023590 -- Probation passed - setting RTP source address to 10.10.1.144:7078
<--- SIP read from UDP:10.10.1.144:5060 --->
<------------->
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=z9hG4bK6c349143;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as0db0840e
To: <sip:301 at 10.10.1.144>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 588ac3e618f7bcc458801e4f111b21e9 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:01:34 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=z9hG4bK6c349143;rport
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as0db0840e
To: <sip:301 at 10.10.1.144>;tag=O6KBR
Call-ID: 588ac3e618f7bcc458801e4f111b21e9 at 10.10.1.201:5060
CSeq: 102 OPTIONS
<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '588ac3e618f7bcc458801e4f111b21e9 at 10.10.1.201:5060' Method: OPTIONS
<--- SIP read from UDP:10.10.1.145:5062 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK0adf25b7;rport=5060
From: <sip:301 at 10.10.1.201>;tag=as2e0a7bb2
To: <sip:306 at 10.10.1.145:5062>;tag=177996708
Call-ID: 3c7241696275b34c664111f736759783 at 10.10.1.201:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3175v2 1.0.1.55
Warning: 399 GS "The call is rejected"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
-- Got SIP response 486 "Busy Here" back from 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=z9hG4bK0adf25b7;rport
Max-Forwards: 70
From: <sip:301 at 10.10.1.201>;tag=as2e0a7bb2
To: <sip:306 at 10.10.1.145:5062>;tag=177996708
Contact: <sip:301 at 10.10.1.201:5060>
Call-ID: 3c7241696275b34c664111f736759783 at 10.10.1.201:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.2.0-rc1
Content-Length: 0
---
-- SIP/306-00000009 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Auto fallthrough, channel 'SIP/301-00000008' status is 'BUSY'
<--- Reliably Transmitting (NAT) to 10.10.1.144:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;received=10.10.1.144;rport=5060
From: <sip:301 at 10.10.1.201>;tag=WSFA2GLK9
To: sip:306 at 10.10.1.201;tag=as0ae1acfc
Call-ID: 64TsxOvORf
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
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0
<------------>
Really destroying SIP dialog '3c7241696275b34c664111f736759783 at 10.10.1.201:5060' Method: INVITE
<--- SIP read from UDP:10.10.1.144:5060 --->
ACK sip:306 at 10.10.1.201 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.~izIMUo0u;rport
Call-ID: 64TsxOvORf
From: <sip:301 at 10.10.1.201>;tag=WSFA2GLK9
To: <sip:306 at 10.10.1.201>;tag=as0ae1acfc
Contact: <sip:301 at 10.10.1.144>;+sip.instance="<urn:uuid:c03a376f-325c-4764-a9c0-827fb2a23a79>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '64TsxOvORf' 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.sq11cyw3l;rport
From: <sip:301 at 10.10.1.201>;tag=TOdTkgRTc
To: sip:301 at 10.10.1.201
CSeq: 26 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.sq11cyw3l;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=as22c12a03
Call-ID: LErqEXzaBt
CSeq: 26 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 --->
<------------->
<--- SIP read from UDP:10.10.1.145:5062 --->
<------------->
Reliably Transmitting (NAT) to 10.10.1.113:42055:
OPTIONS sip:402 at 10.10.1.113:42055;ob SIP/2.0
Via: SIP/2.0/UDP 10.10.1.201:5060;branch=z9hG4bK20f2fbcb;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as5a859160
To: <sip:402 at 10.10.1.113:42055;ob>
Contact: <sip:asterisk at 10.10.1.201:5060>
Call-ID: 01e706c701493e44456abe8551c383a3 at 10.10.1.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.2.0-rc1
Date: Wed, 07 May 2014 12:01:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.10.1.113:42055 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.201:5060;rport=5060;received=10.10.1.201;branch=z9hG4bK20f2fbcb
Call-ID: 01e706c701493e44456abe8551c383a3 at 10.10.1.201:5060
From: "asterisk" <sip:asterisk at 10.10.1.201>;tag=as5a859160
To: <sip:402 at 10.10.1.113;ob>;tag=z9hG4bK20f2fbcb
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_hammerhead-19/r2398
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3608452942 3608452942 IN IP4 10.10.1.113
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 9 0 8 101
c=IN IP4 10.10.1.113
a=sendrecv
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
<------------->
--- (13 headers 12 lines) ---
Really destroying SIP dialog '01e706c701493e44456abe8551c383a3 at 10.10.1.201:5060' Method: OPTIONS
[May 7 14:01:51] WARNING[1820]: res_musiconhold.c:744 monmp3thread: poll() failed: Interrupted system call
<--- SIP read from UDP:10.10.1.144:5060 --->
<------------->
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