rendered paste body[Oct 13 18:13:44] VERBOSE[16820] config.c: == Parsing '/tested/asterisk-trunk/live/etc/asterisk/logger.conf': [Oct 13 18:13:44] VERBOSE[16820] config.c: == Found
[Oct 13 18:13:44] VERBOSE[16820] logger.c: Asterisk Queue Logger restarted
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26794 --->
INVITE sip:8888@192.168.6.23 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-b02eb04ab803457b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@192.168.6.111:26794>
To: "8888"<sip:8888@192.168.6.23>
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 483
v=0
o=- 1 2 IN IP4 192.168.6.111
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.6.111
t=0 0
m=audio 8338 RTP/SAVP 107 0 8 18 101
a=alt:1 1 : h+gdtF5l DQtrVraR 192.168.6.111 8338
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SB7JgYI4zFk7uvjW/g2OJCrXqzRlwNmoMvSMwLx3
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:QPiuNwVn+t0LkuU72YFrY+gGDv+qoaEbjbkXg2e/
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: --- (12 headers 15 lines) ---
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Sending to 192.168.6.111:26794 (no NAT)
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Using INVITE request as basis request - YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found peer 'test' for 'test' from 192.168.6.111:26794
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.6.111:26794 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-b02eb04ab803457b-1---d8754z-;received=192.168.6.111;rport=26794
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
To: "8888"<sip:8888@192.168.6.23>;tag=as028b1012
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 1 INVITE
Server: Asterisk PBX SVN-trunk-r291361
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40634acf"
Content-Length: 0
<------------>
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Scheduling destruction of SIP dialog 'YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.' in 32000 ms (Method: INVITE)
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26794 --->
ACK sip:8888@192.168.6.23 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-b02eb04ab803457b-1---d8754z-;rport
To: "8888"<sip:8888@192.168.6.23>;tag=as028b1012
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 1 ACK
Content-Length: 0
<------------->
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: --- (7 headers 0 lines) ---
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26794 --->
INVITE sip:8888@192.168.6.23 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-db4b3f23e37d4a3e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@192.168.6.111:26794>
To: "8888"<sip:8888@192.168.6.23>
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102u stamp 52345
Authorization: Digest username="test",realm="asterisk",nonce="40634acf",uri="sip:8888@192.168.6.23",response="c6e6e275fef07e439eeb627330772f85",algorithm=MD5
Content-Length: 483
v=0
o=- 1 2 IN IP4 192.168.6.111
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.6.111
t=0 0
m=audio 8338 RTP/SAVP 107 0 8 18 101
a=alt:1 1 : h+gdtF5l DQtrVraR 192.168.6.111 8338
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SB7JgYI4zFk7uvjW/g2OJCrXqzRlwNmoMvSMwLx3
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:QPiuNwVn+t0LkuU72YFrY+gGDv+qoaEbjbkXg2e/
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: --- (13 headers 15 lines) ---
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Sending to 192.168.6.111:26794 (no NAT)
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Using INVITE request as basis request - YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found peer 'test' for 'test' from 192.168.6.111:26794
[Oct 13 18:14:06] VERBOSE[16800] netsock2.c: == Using SIP RTP CoS mark 5
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found RTP audio format 107
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found RTP audio format 0
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found RTP audio format 8
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found RTP audio format 18
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found RTP audio format 101
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found audio description format BV32 for ID 107
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found audio description format G729 for ID 18
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Peer audio RTP is at port 192.168.6.111:8338
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: Looking for 8888 in default (domain 192.168.6.23)
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: list_route: hop: <sip:test@192.168.6.111:26794>
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.6.111:26794 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-db4b3f23e37d4a3e-1---d8754z-;received=192.168.6.111;rport=26794
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
To: "8888"<sip:8888@192.168.6.23>
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-trunk-r291361
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8888@192.168.6.23:5060>
Content-Length: 0
<------------>
[Oct 13 18:14:06] VERBOSE[16843] pbx.c: -- Executing [8888@default:1] NoOp("SIP/test-0000000c", "TEST CALL") in new stack
[Oct 13 18:14:06] VERBOSE[16843] pbx.c: -- Executing [8888@default:2] Set("SIP/test-0000000c", "_SIPSRTP_CRYPTO=enable") in new stack
[Oct 13 18:14:06] VERBOSE[16843] pbx.c: -- Executing [8888@default:3] Dial("SIP/test-0000000c", "SIP/test2") in new stack
[Oct 13 18:14:06] VERBOSE[16843] netsock2.c: == Using SIP RTP CoS mark 5
[Oct 13 18:14:06] VERBOSE[16843] chan_sip.c: Audio is at 5060
[Oct 13 18:14:06] VERBOSE[16843] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 13 18:14:06] VERBOSE[16843] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Oct 13 18:14:06] VERBOSE[16843] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 13 18:14:06] VERBOSE[16843] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP
[Oct 13 18:14:06] VERBOSE[16843] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 13 18:14:06] VERBOSE[16843] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.6.111:26992:
INVITE sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.23:5060;branch=z9hG4bK442ad8fc
Max-Forwards: 70
From: "test" <sip:test@192.168.6.23>;tag=as6680e861
To: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>
Contact: <sip:test@192.168.6.23:5060>
Call-ID: 7fdfb63a39f0a8e011a434c84a96a216@192.168.6.23:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r291361
Date: Wed, 13 Oct 2010 15:14:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 293
v=0
o=root 1193097017 1193097017 IN IP4 192.168.6.23
s=Asterisk PBX SVN-trunk-r291361
c=IN IP4 192.168.6.23
t=0 0
m=audio 19798 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Oct 13 18:14:06] VERBOSE[16843] app_dial.c: -- Called test2
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26992 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.6.23:5060;branch=z9hG4bK442ad8fc
Contact: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>
To: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>;tag=12197249
From: "test"<sip:test@192.168.6.23>;tag=as6680e861
Call-ID: 7fdfb63a39f0a8e011a434c84a96a216@192.168.6.23:5060
CSeq: 102 INVITE
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 0
<------------->
[Oct 13 18:14:06] VERBOSE[16800] chan_sip.c: --- (9 headers 0 lines) ---
[Oct 13 18:14:06] VERBOSE[16843] app_dial.c: -- SIP/test2-0000000d is ringing
[Oct 13 18:14:06] VERBOSE[16843] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.6.111:26794 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-db4b3f23e37d4a3e-1---d8754z-;received=192.168.6.111;rport=26794
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
To: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-trunk-r291361
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8888@192.168.6.23:5060>
Content-Length: 0
<------------>
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26992 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.6.23:5060;branch=z9hG4bK442ad8fc
Contact: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>
To: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>;tag=12197249
From: "test"<sip:test@192.168.6.23>;tag=as6680e861
Call-ID: 7fdfb63a39f0a8e011a434c84a96a216@192.168.6.23:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 188
v=0
o=- 6 2 IN IP4 192.168.6.111
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.6.111
t=0 0
m=audio 22100 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: --- (11 headers 9 lines) ---
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Found RTP audio format 0
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Found RTP audio format 8
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Found RTP audio format 101
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Peer audio RTP is at port 192.168.6.111:22100
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: list_route: hop: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: set_destination: Parsing <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943> for address/port to send to
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: set_destination: set destination to 192.168.6.111:26992
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Transmitting (no NAT) to 192.168.6.111:26992:
ACK sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.23:5060;branch=z9hG4bK6a2c47b5
Max-Forwards: 70
From: "test" <sip:test@192.168.6.23>;tag=as6680e861
To: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>;tag=12197249
Contact: <sip:test@192.168.6.23:5060>
Call-ID: 7fdfb63a39f0a8e011a434c84a96a216@192.168.6.23:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r291361
Content-Length: 0
---
[Oct 13 18:14:08] VERBOSE[16843] app_dial.c: -- SIP/test2-0000000d answered SIP/test-0000000c
[Oct 13 18:14:08] VERBOSE[16843] chan_sip.c: Audio is at 5060
[Oct 13 18:14:08] VERBOSE[16843] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 13 18:14:08] VERBOSE[16843] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 13 18:14:08] VERBOSE[16843] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 13 18:14:08] VERBOSE[16843] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.6.111:26794 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-db4b3f23e37d4a3e-1---d8754z-;received=192.168.6.111;rport=26794
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
To: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-trunk-r291361
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8888@192.168.6.23:5060>
Content-Type: application/sdp
Content-Length: 355
v=0
o=root 1424231712 1424231712 IN IP4 192.168.6.23
s=Asterisk PBX SVN-trunk-r291361
c=IN IP4 192.168.6.23
t=0 0
m=audio 15012 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:a4c4dEDZBt3hJNUIZsPGDRhXdvU71J7N0OKjpnGB
<------------>
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26794 --->
ACK sip:8888@192.168.6.23:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-ce09a372e975b87a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@192.168.6.111:26794>
To: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 2 ACK
User-Agent: eyeBeam release 1102u stamp 52345
Authorization: Digest username="test",realm="asterisk",nonce="40634acf",uri="sip:8888@192.168.6.23",response="c6e6e275fef07e439eeb627330772f85",algorithm=MD5
Content-Length: 0
<------------->
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26794 --->
INVITE sip:8888@192.168.6.23:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-dc396c71f90f8939-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@192.168.6.111:26794>
To: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102u stamp 52345
Authorization: Digest username="test",realm="asterisk",nonce="40634acf",uri="sip:8888@192.168.6.23:5060",response="362402f6133c907a633cabcb46316bb0",algorithm=MD5
Content-Length: 350
v=0
o=- 1 3 IN IP4 192.168.6.111
s=CounterPath eyeBeam 1.5
c=IN IP4 0.0.0.0
t=0 0
m=audio 8338 RTP/SAVP 0 8 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SB7JgYI4zFk7uvjW/g2OJCrXqzRlwNmoMvSMwLx3
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:QPiuNwVn+t0LkuU72YFrY+gGDv+qoaEbjbkXg2e/
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly
<------------->
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: --- (13 headers 11 lines) ---
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Sending to 192.168.6.111:26794 (no NAT)
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Found RTP audio format 0
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Found RTP audio format 8
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Found RTP audio format 101
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Peer audio RTP is at port 0.0.0.0:8338
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.6.111:26794 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-dc396c71f90f8939-1---d8754z-;received=192.168.6.111;rport=26794
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
To: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 3 INVITE
Server: Asterisk PBX SVN-trunk-r291361
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8888@192.168.6.23:5060>
Content-Length: 0
<------------>
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Audio is at 5060
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.6.111:26794 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-dc396c71f90f8939-1---d8754z-;received=192.168.6.111;rport=26794
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
To: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 3 INVITE
Server: Asterisk PBX SVN-trunk-r291361
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8888@192.168.6.23:5060>
Content-Type: application/sdp
Content-Length: 355
v=0
o=root 1424231712 1424231713 IN IP4 192.168.6.23
s=Asterisk PBX SVN-trunk-r291361
c=IN IP4 192.168.6.23
t=0 0
m=audio 15012 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:a4c4dEDZBt3hJNUIZsPGDRhXdvU71J7N0OKjpnGB
<------------>
[Oct 13 18:14:08] VERBOSE[16843] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/test2-0000000d
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26794 --->
ACK sip:8888@192.168.6.23:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.111:26794;branch=z9hG4bK-d8754z-ad3b0200d634c727-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@192.168.6.111:26794>
To: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
From: "test"<sip:test@192.168.6.23>;tag=035fb70a
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 3 ACK
User-Agent: eyeBeam release 1102u stamp 52345
Authorization: Digest username="test",realm="asterisk",nonce="40634acf",uri="sip:8888@192.168.6.23:5060",response="362402f6133c907a633cabcb46316bb0",algorithm=MD5
Content-Length: 0
<------------->
[Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 13 18:14:10] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26992 --->
BYE sip:test@192.168.6.23:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.111:26992;branch=z9hG4bK-d8754z-523eba339d2d627e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>
To: "test"<sip:test@192.168.6.23>;tag=as6680e861
From: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>;tag=12197249
Call-ID: 7fdfb63a39f0a8e011a434c84a96a216@192.168.6.23:5060
CSeq: 2 BYE
User-Agent: eyeBeam release 1102u stamp 52345
Reason: SIP;description="User Hung Up"
Content-Length: 0
<------------->
[Oct 13 18:14:10] VERBOSE[16800] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 13 18:14:10] VERBOSE[16800] chan_sip.c: Sending to 192.168.6.111:26992 (no NAT)
[Oct 13 18:14:10] VERBOSE[16800] chan_sip.c: Scheduling destruction of SIP dialog '7fdfb63a39f0a8e011a434c84a96a216@192.168.6.23:5060' in 32000 ms (Method: BYE)
[Oct 13 18:14:10] VERBOSE[16800] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.6.111:26992 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.6.111:26992;branch=z9hG4bK-d8754z-523eba339d2d627e-1---d8754z-;received=192.168.6.111;rport=26992
From: <sip:test2@192.168.6.111:26992;rinstance=d07a71d3c83ce943>;tag=12197249
To: "test"<sip:test@192.168.6.23>;tag=as6680e861
Call-ID: 7fdfb63a39f0a8e011a434c84a96a216@192.168.6.23:5060
CSeq: 2 BYE
Server: Asterisk PBX SVN-trunk-r291361
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Oct 13 18:14:10] VERBOSE[16843] res_musiconhold.c: -- Stopped music on hold on SIP/test2-0000000d
[Oct 13 18:14:10] VERBOSE[16843] pbx.c: == Spawn extension (default, 8888, 3) exited non-zero on 'SIP/test-0000000c'
[Oct 13 18:14:10] VERBOSE[16843] chan_sip.c: Scheduling destruction of SIP dialog 'YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.' in 32000 ms (Method: ACK)
[Oct 13 18:14:10] VERBOSE[16843] chan_sip.c: set_destination: Parsing <sip:test@192.168.6.111:26794> for address/port to send to
[Oct 13 18:14:10] VERBOSE[16843] chan_sip.c: set_destination: set destination to 192.168.6.111:26794
[Oct 13 18:14:10] VERBOSE[16843] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.6.111:26794:
BYE sip:test@192.168.6.111:26794 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.23:5060;branch=z9hG4bK58d5941c;rport
Max-Forwards: 70
From: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
To: "test"<sip:test@192.168.6.23>;tag=035fb70a
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 102 BYE
User-Agent: Asterisk PBX SVN-trunk-r291361
Proxy-Authorization: Digest username="test", realm="asterisk", algorithm=MD5, uri="192.168.6.23", nonce="", response="52a2e8a8da97478bc6365baaacbbf168"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Oct 13 18:14:11] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26794 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.6.23:5060;branch=z9hG4bK58d5941c;rport=5060
Contact: <sip:test@192.168.6.111:26794>
To: "test"<sip:test@192.168.6.23>;tag=035fb70a
From: "8888"<sip:8888@192.168.6.23>;tag=as0f7c83ca
Call-ID: YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.
CSeq: 102 BYE
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 0
<------------->
[Oct 13 18:14:11] VERBOSE[16800] chan_sip.c: --- (9 headers 0 lines) ---
[Oct 13 18:14:11] VERBOSE[16800] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[Oct 13 18:14:11] VERBOSE[16800] chan_sip.c: Really destroying SIP dialog 'YTJkZmJhZGMwZmRhZWNhMGUxNTQzYzEwYmY3MjAyNzE.' Method: ACK
[Oct 13 18:14:11] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26992 --->
<------------->
[Oct 13 18:14:12] VERBOSE[16800] chan_sip.c:
<--- SIP read from UDP:192.168.6.111:26794 --->
<------------->
[Oct 13 18:14:41] NOTICE[16800] chan_sip.c: Received SIP subscribe for peer without mailbox: test2
[Oct 13 18:14:42] NOTICE[16800] chan_sip.c: Received SIP subscribe for peer without mailbox: test
[Oct 13 18:17:41] NOTICE[16800] chan_sip.c: Received SIP subscribe for peer without mailbox: test2
[Oct 13 18:17:42] NOTICE[16800] chan_sip.c: Received SIP subscribe for peer without mailbox: test