[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: To: "8888" From: "test";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";tag=035fb70a To: "8888";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";tag=as028b1012 From: "test";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: To: "8888" From: "test";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: [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";tag=035fb70a To: "8888" 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: 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" ;tag=as6680e861 To: Contact: 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: To: ;tag=12197249 From: "test";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";tag=035fb70a To: "8888";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: 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: To: ;tag=12197249 From: "test";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: [Oct 13 18:14:08] VERBOSE[16800] chan_sip.c: set_destination: Parsing 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" ;tag=as6680e861 To: ;tag=12197249 Contact: 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";tag=035fb70a To: "8888";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: 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: To: "8888";tag=as0f7c83ca From: "test";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: To: "8888";tag=as0f7c83ca From: "test";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";tag=035fb70a To: "8888";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: 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";tag=035fb70a To: "8888";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: 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: To: "8888";tag=as0f7c83ca From: "test";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: To: "test";tag=as6680e861 From: ;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: ;tag=12197249 To: "test";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 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";tag=as0f7c83ca To: "test";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: To: "test";tag=035fb70a From: "8888";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