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mmack/gist:3188443

Created July 27, 2012 14:45
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<--- SIP read from UDP:127.0.0.1:47475 --->
BYE sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1:5060 SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.3.186.15:47475;rport;branch=z9hG4bKOo_VR_N-UBXujgveCT6nvg..
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=54930742562
To: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1>;tag=as4aeb0764
Contact: <sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:47475>
CSeq: 2 BYE
Max-Forwards: 70
Call-ID: 447675260@10.3.186.15
<------------->
--- (9 headers 0 lines) ---
Sending to 127.0.0.1:47475 (no NAT)
Scheduling destruction of SIP dialog '447675260@10.3.186.15' in 32000 ms (Method: BYE)
<--- Transmitting (no NAT) to 127.0.0.1:47475 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.186.15:47475;branch=z9hG4bKOo_VR_N-UBXujgveCT6nvg..;received=127.0.0.1;rport=47475
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=54930742562
To: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1>;tag=as4aeb0764
Call-ID: 447675260@10.3.186.15
CSeq: 2 BYE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:127.0.0.1:47475 --->
REGISTER sip:127.0.0.1 SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.3.186.15:47475;rport;branch=z9hG4bK7pDU6Y1jLkzHeoxq3Kilfg..
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=54930742574
Expires: 0
To: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>
Contact: <sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:47475>
CSeq: 2 REGISTER
Max-Forwards: 70
Call-ID: 1540408454@10.3.186.15
<------------->
--- (10 headers 0 lines) ---
Sending to 127.0.0.1:47475 (no NAT)
<--- Transmitting (no NAT) to 127.0.0.1:47475 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.186.15:47475;branch=z9hG4bK7pDU6Y1jLkzHeoxq3Kilfg..;received=127.0.0.1;rport=47475
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=54930742574
To: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=as4f5ac798
Call-ID: 1540408454@10.3.186.15
CSeq: 2 REGISTER
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Fri, 27 Jul 2012 14:43:34 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1540408454@10.3.186.15' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:127.0.0.1:53516 --->
REGISTER sip:127.0.0.1 SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.3.186.15:53516;rport;branch=z9hG4bKbYZQGbzdSYARHd8UG-EHJA..
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=148870882968
Expires: 180
To: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>
Contact: <sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:53516>
CSeq: 1 REGISTER
Max-Forwards: 70
Call-ID: 1200852103@10.3.186.15
<------------->
--- (10 headers 0 lines) ---
Sending to 127.0.0.1:53516 (no NAT)
<--- Transmitting (no NAT) to 127.0.0.1:53516 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.186.15:53516;branch=z9hG4bKbYZQGbzdSYARHd8UG-EHJA..;received=127.0.0.1;rport=53516
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=148870882968
To: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=as5b3cab3c
Call-ID: 1200852103@10.3.186.15
CSeq: 1 REGISTER
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 180
Contact: <sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:53516>;expires=180
Date: Fri, 27 Jul 2012 14:43:36 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1200852103@10.3.186.15' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:127.0.0.1:53516 --->
INVITE sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1 SIP/2.0
Content-Length: 393
Via: SIP/2.0/UDP 10.3.186.15:53516;rport;branch=z9hG4bKHj06RmFI3hhKVmsrAnb9OA..
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=148870882996
To: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1>
Contact: <sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:53516>
CSeq: 1 INVITE
Max-Forwards: 70
Call-ID: 1958616387@10.3.186.15
Content-Type: application/sdp
v=0
o=- 1343400216 1343400216 IN IP4 1.audio.de.dozeo.net
s=-
c=IN IP4 10.3.186.15
t=0 0
m=audio 31734 RTP/AVP 96 98 0 8 101
a=rtpmap:96 speex/16000
a=rtpmap:98 speex/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
m=video 38930 RTP/AVP 99 97
a=fmtp:99 profile-level-id=420014;packetization-mode=1
a=rtpmap:99 h264/90000
a=rtpmap:97 x-flv/90000
<------------->
--- (10 headers 15 lines) ---
Sending to 127.0.0.1:53516 (no NAT)
Using INVITE request as basis request - 1958616387@10.3.186.15
No matching peer for '4521f414d4503e22672fcd85261fa7af' from '127.0.0.1:53516'
Found RTP audio format 96
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 96
Found audio description format speex for ID 98
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 RTP video format 97
Found video description format h264 for ID 99
Capabilities: us - (gsm|ulaw|alaw|speex|speex16), peer - audio=(ulaw|alaw|speex|speex16)/video=(ilbc|h264)/text=(nothing), combined - (ulaw|alaw|speex|speex16)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.3.186.15:31734
Looking for 6mzh1mp0345av4smgbjiy9ru2 in default (domain 127.0.0.1)
list_route: hop: <sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:53516>
<--- Transmitting (no NAT) to 127.0.0.1:53516 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.3.186.15:53516;branch=z9hG4bKHj06RmFI3hhKVmsrAnb9OA..;received=127.0.0.1;rport=53516
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=148870882996
To: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1>
Call-ID: 1958616387@10.3.186.15
CSeq: 1 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1:5060>
Content-Length: 0
<------------>
Audio is at 11278
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100016 (speex16) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 127.0.0.1:53516 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.186.15:53516;branch=z9hG4bKHj06RmFI3hhKVmsrAnb9OA..;received=127.0.0.1;rport=53516
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=148870882996
To: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1>;tag=as50cf2dd1
Call-ID: 1958616387@10.3.186.15
CSeq: 1 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 358
v=0
o=root 335477617 335477617 IN IP4 127.0.0.1
s=Asterisk PBX 10.5.2
c=IN IP4 127.0.0.1
t=0 0
m=audio 11278 RTP/AVP 0 8 98 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 speex/8000
a=rtpmap:96 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99 97
<------------>
<--- SIP read from UDP:127.0.0.1:53516 --->
ACK sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1:5060 SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.3.186.15:53516;rport;branch=z9hG4bKHj06RmFI3hhKVmsrAnb9OA..
From: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=148870882996
To: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1>;tag=as50cf2dd1
Contact: <sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:53516>
CSeq: 1 ACK
Max-Forwards: 70
Call-ID: 1958616387@10.3.186.15
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '447675260@10.3.186.15' Method: BYE
Really destroying SIP dialog '1540408454@10.3.186.15' Method: REGISTER
Really destroying SIP dialog '1200852103@10.3.186.15' Method: REGISTER
[Jul 27 16:44:37] NOTICE[16111]: chan_sip.c:26662 check_rtp_timeout: Disconnecting call 'SIP/127.0.0.1-00000263' for lack of RTP activity in 61 seconds
Scheduling destruction of SIP dialog '1958616387@10.3.186.15' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:53516> for address/port to send to
set_destination: set destination to 10.3.186.15:53516
Reliably Transmitting (no NAT) to 10.3.186.15:53516:
BYE sip:4521f414d4503e22672fcd85261fa7af@10.3.186.15:53516 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6c5b5a6c;rport
Max-Forwards: 70
From: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1>;tag=as50cf2dd1
To: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=148870882996
Call-ID: 1958616387@10.3.186.15
CSeq: 102 BYE
User-Agent: Asterisk PBX 10.5.2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:127.0.0.1:53516 --->
SIP/2.0 200 OK
Content-Length: 0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;branch=z9hG4bK6c5b5a6c
From: <sip:6mzh1mp0345av4smgbjiy9ru2@127.0.0.1>;tag=as50cf2dd1
To: "4521f414d4503e22672fcd85261fa7af" <sip:4521f414d4503e22672fcd85261fa7af@127.0.0.1>;tag=148870882996
CSeq: 102 BYE
Call-ID: 1958616387@10.3.186.15
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1958616387@10.3.186.15' Method: ACK
<--- SIP read from UDP:127.0.0.1:36848 --->
REGISTER sip:127.0.0.1 SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.3.186.15:36848;rport;branch=z9hG4bKT215MybM6Iv0Y_vX-2hbYw..
From: "04926a00-a8a2-012f-919c-0852dd825506" <sip:04926a00-a8a2-012f-919c-0852dd825506@127.0.0.1>;tag=210990392175
Expires: 180
To: "04926a00-a8a2-012f-919c-0852dd825506" <sip:04926a00-a8a2-012f-919c-0852dd825506@127.0.0.1>
Contact: <sip:04926a00-a8a2-012f-919c-0852dd825506@10.3.186.15:36848>
CSeq: 1 REGISTER
Max-Forwards: 70
Call-ID: 2117949954@10.3.186.15
<------------->
--- (10 headers 0 lines) ---
Sending to 127.0.0.1:36848 (no NAT)
<--- Transmitting (no NAT) to 127.0.0.1:36848 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.186.15:36848;branch=z9hG4bKT215MybM6Iv0Y_vX-2hbYw..;received=127.0.0.1;rport=36848
From: "04926a00-a8a2-012f-919c-0852dd825506" <sip:04926a00-a8a2-012f-919c-0852dd825506@127.0.0.1>;tag=210990392175
To: "04926a00-a8a2-012f-919c-0852dd825506" <sip:04926a00-a8a2-012f-919c-0852dd825506@127.0.0.1>;tag=as1b1c7138
Call-ID: 2117949954@10.3.186.15
CSeq: 1 REGISTER
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 180
Contact: <sip:04926a00-a8a2-012f-919c-0852dd825506@10.3.186.15:36848>;expires=180
Date: Fri, 27 Jul 2012 14:44:46 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '2117949954@10.3.186.15' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:127.0.0.1:36848 --->
INVITE sip:3bd8viufko0qs14eqipr2qyi2@127.0.0.1 SIP/2.0
Content-Length: 393
Via: SIP/2.0/UDP 10.3.186.15:36848;rport;branch=z9hG4bKqa1XN40Q25dG41XMMU5Pkw..
From: "04926a00-a8a2-012f-919c-0852dd825506" <sip:04926a00-a8a2-012f-919c-0852dd825506@127.0.0.1>;tag=210990392181
To: <sip:3bd8viufko0qs14eqipr2qyi2@127.0.0.1>
Contact: <sip:04926a00-a8a2-012f-919c-0852dd825506@10.3.186.15:36848>
CSeq: 1 INVITE
Max-Forwards: 70
Call-ID: 2077614715@10.3.186.15
Content-Type: application/sdp
v=0
o=- 1343400286 1343400286 IN IP4 1.audio.de.dozeo.net
s=-
c=IN IP4 10.3.186.15
t=0 0
m=audio 26444 RTP/AVP 96 98 0 8 101
a=rtpmap:96 speex/16000
a=rtpmap:98 speex/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
m=video 44542 RTP/AVP 99 97
a=fmtp:99 profile-level-id=420014;packetization-mode=1
a=rtpmap:99 h264/90000
a=rtpmap:97 x-flv/90000
<------------->
--- (10 headers 15 lines) ---
Sending to 127.0.0.1:36848 (no NAT)
Using INVITE request as basis request - 2077614715@10.3.186.15
No matching peer for '04926a00-a8a2-012f-919c-0852dd825506' from '127.0.0.1:36848'
Found RTP audio format 96
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 96
Found audio description format speex for ID 98
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 RTP video format 97
Found video description format h264 for ID 99
Capabilities: us - (gsm|ulaw|alaw|speex|speex16), peer - audio=(ulaw|alaw|speex|speex16)/video=(ilbc|h264)/text=(nothing), combined - (ulaw|alaw|speex|speex16)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.3.186.15:26444
Looking for 3bd8viufko0qs14eqipr2qyi2 in default (domain 127.0.0.1)
list_route: hop: <sip:04926a00-a8a2-012f-919c-0852dd825506@10.3.186.15:36848>
<--- Transmitting (no NAT) to 127.0.0.1:36848 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.3.186.15:36848;branch=z9hG4bKqa1XN40Q25dG41XMMU5Pkw..;received=127.0.0.1;rport=36848
From: "04926a00-a8a2-012f-919c-0852dd825506" <sip:04926a00-a8a2-012f-919c-0852dd825506@127.0.0.1>;tag=210990392181
To: <sip:3bd8viufko0qs14eqipr2qyi2@127.0.0.1>
Call-ID: 2077614715@10.3.186.15
CSeq: 1 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3bd8viufko0qs14eqipr2qyi2@127.0.0.1:5060>
Content-Length: 0
<------------>
Audio is at 13418
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100016 (speex16) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 127.0.0.1:36848 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.186.15:36848;branch=z9hG4bKqa1XN40Q25dG41XMMU5Pkw..;received=127.0.0.1;rport=36848
From: "04926a00-a8a2-012f-919c-0852dd825506" <sip:04926a00-a8a2-012f-919c-0852dd825506@127.0.0.1>;tag=210990392181
To: <sip:3bd8viufko0qs14eqipr2qyi2@127.0.0.1>;tag=as0a871826
Call-ID: 2077614715@10.3.186.15
CSeq: 1 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3bd8viufko0qs14eqipr2qyi2@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 356
v=0
o=root 95746441 95746441 IN IP4 127.0.0.1
s=Asterisk PBX 10.5.2
c=IN IP4 127.0.0.1
t=0 0
m=audio 13418 RTP/AVP 0 8 98 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 speex/8000
a=rtpmap:96 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99 97
<------------>
<--- SIP read from UDP:127.0.0.1:36848 --->
ACK sip:3bd8viufko0qs14eqipr2qyi2@127.0.0.1:5060 SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.3.186.15:36848;rport;branch=z9hG4bKqa1XN40Q25dG41XMMU5Pkw..
From: "04926a00-a8a2-012f-919c-0852dd825506" <sip:04926a00-a8a2-012f-919c-0852dd825506@127.0.0.1>;tag=210990392181
To: <sip:3bd8viufko0qs14eqipr2qyi2@127.0.0.1>;tag=as0a871826
Contact: <sip:04926a00-a8a2-012f-919c-0852dd825506@10.3.186.15:36848>
CSeq: 1 ACK
Max-Forwards: 70
Call-ID: 2077614715@10.3.186.15
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1*CLI>
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