[Sep 7 05:51:10] VERBOSE[18214] config.c: == Parsing '/etc/asterisk/logger.conf': [Sep 7 05:51:10] DEBUG[18214] config.c: Parsing /etc/asterisk/logger.conf [Sep 7 05:51:10] VERBOSE[18214] config.c: == Found [Sep 7 05:51:10] VERBOSE[18214] logger.c: Asterisk Queue Logger restarted [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: = Looking for Call ID: 1834435539@192_168_1_150 (Checking From) --From tag 3401510724 --To-tag [Sep 7 05:51:18] DEBUG[10974] acl.c: For destination '192.168.1.150', our source address is '192.168.1.1'. [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.1:5060 [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 1834435539@192_168_1_150 - REGISTER (No RTP) [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Sep 7 05:51:18] DEBUG[10974] netsock2.c: Splitting '192.168.1.150:5064' gives... [Sep 7 05:51:18] DEBUG[10974] netsock2.c: ...host '192.168.1.150' and port '5064'. [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: = Looking for Call ID: 1834435539@192_168_1_150 (Checking From) --From tag 3401510724 --To-tag [Sep 7 05:51:18] DEBUG[10974] netsock2.c: Splitting 'home.smartbyte.de' gives... [Sep 7 05:51:18] DEBUG[10974] netsock2.c: ...host 'home.smartbyte.de' and port '(null)'. [Sep 7 05:51:18] DEBUG[10974] netsock2.c: Splitting 'home.smartbyte.de' gives... [Sep 7 05:51:18] DEBUG[10974] netsock2.c: ...host 'home.smartbyte.de' and port '(null)'. [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Sep 7 05:51:18] DEBUG[10974] netsock2.c: Splitting '192.168.1.150:5064' gives... [Sep 7 05:51:18] DEBUG[10974] netsock2.c: ...host '192.168.1.150' and port '5064'. [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Sep 7 05:51:18] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:51:18] DEBUG[10957] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Sep 7 05:51:18] DEBUG[10957] chan_sip.c: Checking device state for peer 100 [Sep 7 05:51:18] DEBUG[10957] devicestate.c: Changing state for SIP/100 - state 1 (Not in use) [Sep 7 05:51:18] DEBUG[10957] devicestate.c: device 'SIP/100' state '1' [Sep 7 05:51:18] DEBUG[11003] app_queue.c: Device 'SIP/100' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 7 05:51:22] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> OPTIONS sip:100@114.77.232.125:5060 SIP/2.0 Via: SIP/2.0/UDP 80.237.160.128:5060;branch=z9hG4bK18bffd4e Max-Forwards: 70 From: "Asterisk" ;tag=as2393e9d7 To: Contact: Call-ID: 07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN--r Date: Mon, 06 Sep 2010 19:51:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 0 [ 43]: OPTIONS sip:100@114.77.232.125:5060 SIP/2.0 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 80.237.160.128:5060;branch=z9hG4bK18bffd4e [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 3 [ 61]: From: "Asterisk" ;tag=as2393e9d7 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 4 [ 33]: To: [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 5 [ 43]: Contact: [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 6 [ 61]: Call-ID: 07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX SVN--r [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 9 [ 35]: Date: Mon, 06 Sep 2010 19:51:22 GMT [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Sep 7 05:51:22] VERBOSE[10974] chan_sip.c: --- (13 headers 0 lines) --- [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: = Looking for Call ID: 07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060 (Checking From) --From tag as2393e9d7 --To-tag [Sep 7 05:51:22] DEBUG[10974] acl.c: For destination '80.237.160.128', our source address is '192.168.1.1'. [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Target address 80.237.160.128:5060 is not local, substituting externaddr [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 114.77.232.125:5060 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060 - OPTIONS (No RTP) [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Sep 7 05:51:22] VERBOSE[10974] chan_sip.c: Looking for 100 in incoming (domain 114.77.232.125:5060) [Sep 7 05:51:22] VERBOSE[10974] chan_sip.c: <--- Transmitting (NAT) to 80.237.160.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 80.237.160.128:5060;branch=z9hG4bK18bffd4e;received=80.237.160.128;rport=5060 From: "Asterisk" ;tag=as2393e9d7 To: ;tag=as02fc2e18 Call-ID: 07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 80.237.160.128:5060;branch=z9hG4bK18bffd4e;received=80.237.160.128;rport=5060 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 2 [ 61]: From: "Asterisk" ;tag=as2393e9d7 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 3 [ 48]: To: ;tag=as02fc2e18 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 4 [ 61]: Call-ID: 07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 9 [ 34]: Contact: [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Header 12 [ 0]: [Sep 7 05:51:22] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 80.237.160.128:5060 [Sep 7 05:51:22] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060' in 32000 ms (Method: OPTIONS) [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: Auto destroying SIP dialog '33e5ea6f01c2e3a7068bd56a5adf62c9@203.12.160.224' [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: Destroying SIP dialog 33e5ea6f01c2e3a7068bd56a5adf62c9@203.12.160.224 [Sep 7 05:51:32] VERBOSE[10974] chan_sip.c: Really destroying SIP dialog '33e5ea6f01c2e3a7068bd56a5adf62c9@203.12.160.224' Method: OPTIONS [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: ---------- SIP HISTORY for '33e5ea6f01c2e3a7068bd56a5adf62c9@203.12.160.224' [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: * SIP Call [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: 001. Rx OPTIONS / 102 OPTIONS / sip:s@114.77.232.125:5060 [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: 002. TxResp SIP/2.0 / 102 OPTIONS - 404 Not Found [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: 003. SchedDestroy 32000 ms [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: 004. AutoDestroy 33e5ea6f01c2e3a7068bd56a5adf62c9@203.12.160.224 [Sep 7 05:51:32] DEBUG[10974] chan_sip.c: ---------- END SIP HISTORY for '33e5ea6f01c2e3a7068bd56a5adf62c9@203.12.160.224' [Sep 7 05:51:38] NOTICE[10974] chan_sip.c: -- Re-registration for smartbyt@aphone6.tpg.com.au [Sep 7 05:51:38] VERBOSE[10974] dnsmgr.c: > doing dnsmgr_lookup for 'aphone6.tpg.com.au' [Sep 7 05:51:39] DEBUG[10974] netsock2.c: Splitting 'aphone6.tpg.com.au' gives... [Sep 7 05:51:39] DEBUG[10974] netsock2.c: ...host 'aphone6.tpg.com.au' and port '(null)'. [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 7553546773a2f5ab51c49c5c58112826@192.168.0.1 - REGISTER (No RTP) [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: SIP Registry aphone6.tpg.com.au: refcount now 3 [Sep 7 05:51:39] DEBUG[10974] acl.c: For destination '203.12.160.224', our source address is '192.168.1.1'. [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Target address 203.12.160.224:5060 is not local, substituting externaddr [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 114.77.232.125:5060 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: SIP Registry aphone6.tpg.com.au: refcount now 4 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Scheduled a registration timeout for aphone6.tpg.com.au id #16738 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: >>> Re-using Auth data for smartbyt@aphone6.tpg.com.au [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Initializing initreq for method REGISTER - callid 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 0 [ 39]: REGISTER sip:aphone6.tpg.com.au SIP/2.0 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK613c5e47;rport [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 3 [ 54]: From: ;tag=as760d47eb [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 4 [ 37]: To: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 6 [ 19]: CSeq: 1009 REGISTER [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 8 [167]: Authorization: Digest username="smartbyt", realm="asterisk", algorithm=MD5, uri="sip:aphone6.tpg.com.au", nonce="650a3199", response="3e1c0b238c3457a57a9a5d7b490af2b0" [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 10 [ 36]: Contact: [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: REGISTER 11 headers, 0 lines [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: REGISTER attempt 1 to smartbyt@aphone6.tpg.com.au [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: Reliably Transmitting (NAT) to 203.12.160.224:5060: REGISTER sip:aphone6.tpg.com.au SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK613c5e47;rport Max-Forwards: 70 From: ;tag=as760d47eb To: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 CSeq: 1009 REGISTER User-Agent: Asterisk PBX SVN-trunk-r285058 Authorization: Digest username="smartbyt", realm="asterisk", algorithm=MD5, uri="sip:aphone6.tpg.com.au", nonce="650a3199", response="3e1c0b238c3457a57a9a5d7b490af2b0" Expires: 120 Contact: Content-Length: 0 --- [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 0 [ 39]: REGISTER sip:aphone6.tpg.com.au SIP/2.0 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK613c5e47;rport [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 3 [ 54]: From: ;tag=as760d47eb [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 4 [ 37]: To: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 6 [ 19]: CSeq: 1009 REGISTER [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 8 [167]: Authorization: Digest username="smartbyt", realm="asterisk", algorithm=MD5, uri="sip:aphone6.tpg.com.au", nonce="650a3199", response="3e1c0b238c3457a57a9a5d7b490af2b0" [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 10 [ 36]: Contact: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 12 [ 0]: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16739 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 203.12.160.224:5060 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: SIP Registry aphone6.tpg.com.au: refcount now 3 [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:203.12.160.224:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK613c5e47;received=114.77.232.125;rport=5060 From: ;tag=as760d47eb To: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 CSeq: 1009 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK613c5e47;received=114.77.232.125;rport=5060 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 2 [ 54]: From: ;tag=as760d47eb [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 3 [ 37]: To: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1009 REGISTER [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 6 [ 24]: User-Agent: Asterisk PBX [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 7 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 8 [ 38]: Contact: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: = Looking for Call ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 (Checking To) --From tag as760d47eb --To-tag [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:203.12.160.224:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK613c5e47;received=114.77.232.125;rport=5060 From: ;tag=as760d47eb To: ;tag=as2da847b8 Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 CSeq: 1009 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="103414c3" Content-Length: 0 <-------------> [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK613c5e47;received=114.77.232.125;rport=5060 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 2 [ 54]: From: ;tag=as760d47eb [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 3 [ 52]: To: ;tag=as2da847b8 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1009 REGISTER [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 6 [ 24]: User-Agent: Asterisk PBX [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 7 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 8 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="103414c3" [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: = Looking for Call ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 (Checking To) --From tag as760d47eb --To-tag as2da847b8 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16739 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Stopping retransmission on '7553546773a2f5ab51c49c5c58112826@192.168.0.1' of Request 1009: Match Found [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: Responding to challenge, registration to domain/host name aphone6.tpg.com.au [Sep 7 05:51:39] VERBOSE[10974] dnsmgr.c: > doing dnsmgr_lookup for 'aphone6.tpg.com.au' [Sep 7 05:51:39] DEBUG[10974] netsock2.c: Splitting 'aphone6.tpg.com.au' gives... [Sep 7 05:51:39] DEBUG[10974] netsock2.c: ...host 'aphone6.tpg.com.au' and port '(null)'. [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Initializing already initialized SIP dialog 7553546773a2f5ab51c49c5c58112826@192.168.0.1 (presumably reinvite) [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 0 [ 39]: REGISTER sip:aphone6.tpg.com.au SIP/2.0 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK046b2533;rport [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 3 [ 54]: From: ;tag=as40b1e6c1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 4 [ 37]: To: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 6 [ 19]: CSeq: 1010 REGISTER [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 8 [167]: Authorization: Digest username="smartbyt", realm="asterisk", algorithm=MD5, uri="sip:aphone6.tpg.com.au", nonce="103414c3", response="98d12076d74e1fabb3a9d2b9a65f7b72" [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 10 [ 36]: Contact: [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: REGISTER 11 headers, 0 lines [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: REGISTER attempt 2 to smartbyt@aphone6.tpg.com.au [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: Reliably Transmitting (NAT) to 203.12.160.224:5060: REGISTER sip:aphone6.tpg.com.au SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK046b2533;rport Max-Forwards: 70 From: ;tag=as40b1e6c1 To: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 CSeq: 1010 REGISTER User-Agent: Asterisk PBX SVN-trunk-r285058 Authorization: Digest username="smartbyt", realm="asterisk", algorithm=MD5, uri="sip:aphone6.tpg.com.au", nonce="103414c3", response="98d12076d74e1fabb3a9d2b9a65f7b72" Expires: 120 Contact: Content-Length: 0 --- [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 0 [ 39]: REGISTER sip:aphone6.tpg.com.au SIP/2.0 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK046b2533;rport [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 3 [ 54]: From: ;tag=as40b1e6c1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 4 [ 37]: To: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 6 [ 19]: CSeq: 1010 REGISTER [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 8 [167]: Authorization: Digest username="smartbyt", realm="asterisk", algorithm=MD5, uri="sip:aphone6.tpg.com.au", nonce="103414c3", response="98d12076d74e1fabb3a9d2b9a65f7b72" [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 10 [ 36]: Contact: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 12 [ 0]: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16740 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 203.12.160.224:5060 [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:203.12.160.224:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK046b2533;received=114.77.232.125;rport=5060 From: ;tag=as40b1e6c1 To: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 CSeq: 1010 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK046b2533;received=114.77.232.125;rport=5060 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 2 [ 54]: From: ;tag=as40b1e6c1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 3 [ 37]: To: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1010 REGISTER [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 6 [ 24]: User-Agent: Asterisk PBX [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 7 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 8 [ 38]: Contact: [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: = Looking for Call ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 (Checking To) --From tag as40b1e6c1 --To-tag [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:203.12.160.224:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK046b2533;received=114.77.232.125;rport=5060 From: ;tag=as40b1e6c1 To: ;tag=as2da847b8 Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 CSeq: 1010 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Mon, 06 Sep 2010 19:51:39 GMT Content-Length: 0 <-------------> [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK046b2533;received=114.77.232.125;rport=5060 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 2 [ 54]: From: ;tag=as40b1e6c1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 3 [ 52]: To: ;tag=as2da847b8 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1010 REGISTER [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 6 [ 24]: User-Agent: Asterisk PBX [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 7 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 8 [ 12]: Expires: 120 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 9 [ 48]: Contact: ;expires=120 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 10 [ 35]: Date: Mon, 06 Sep 2010 19:51:39 GMT [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: --- (12 headers 0 lines) --- [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: = Looking for Call ID: 7553546773a2f5ab51c49c5c58112826@192.168.0.1 (Checking To) --From tag as40b1e6c1 --To-tag as2da847b8 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16740 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Stopping retransmission on '7553546773a2f5ab51c49c5c58112826@192.168.0.1' of Request 1010: Match Found [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Registration successful [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: Cancelling timeout 16738 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: SIP Registry aphone6.tpg.com.au: refcount now 2 [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: SIP Registry aphone6.tpg.com.au: refcount now 1 [Sep 7 05:51:39] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '7553546773a2f5ab51c49c5c58112826@192.168.0.1' in 32000 ms (Method: REGISTER) [Sep 7 05:51:39] NOTICE[10974] chan_sip.c: Outbound Registration: Expiry for aphone6.tpg.com.au is 120 sec (Scheduling reregistration in 105 s) [Sep 7 05:51:39] DEBUG[10974] chan_sip.c: SIP Registry aphone6.tpg.com.au: refcount now 2 [Sep 7 05:51:46] NOTICE[10974] chan_sip.c: -- Re-registration for 17474987674@proxy01.sipphone.com [Sep 7 05:51:46] VERBOSE[10974] dnsmgr.c: > doing dnsmgr_lookup for 'proxy01.sipphone.com' [Sep 7 05:51:46] VERBOSE[10974] srv.c: > ast_get_srv: SRV lookup for '_sip._udp.proxy01.sipphone.com' mapped to host proxy01.sipphone.com, port 5060 [Sep 7 05:51:46] DEBUG[10974] netsock2.c: Splitting 'proxy01.sipphone.com' gives... [Sep 7 05:51:46] DEBUG[10974] netsock2.c: ...host 'proxy01.sipphone.com' and port '(null)'. [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 4c76c12233a7c5d546609dc16317a534@192.168.0.1 - REGISTER (No RTP) [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: SIP Registry proxy01.sipphone.com: refcount now 3 [Sep 7 05:51:46] DEBUG[10974] acl.c: For destination '198.65.166.131', our source address is '192.168.1.1'. [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Target address 198.65.166.131:5060 is not local, substituting externaddr [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 114.77.232.125:5060 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: SIP Registry proxy01.sipphone.com: refcount now 4 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Scheduled a registration timeout for proxy01.sipphone.com id #16743 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: >>> Re-using Auth data for 17474987674@proxy01.sipphone.com [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Initializing initreq for method REGISTER - callid 4c76c12233a7c5d546609dc16317a534@192.168.0.1 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 0 [ 41]: REGISTER sip:proxy01.sipphone.com SIP/2.0 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2c13857d;rport [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 3 [ 59]: From: ;tag=as39ec3c05 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 4 [ 42]: To: [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 4c76c12233a7c5d546609dc16317a534@192.168.0.1 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 6 [ 18]: CSeq: 708 REGISTER [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 8 [216]: Authorization: Digest username="17474987674", realm="proxy01.sipphone.com", algorithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4c8546a59d9e2dacbc828f7251404506a52ba311", response="2b10af385cb0e1e0a8c3b72009f782ec" [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 10 [ 46]: Contact: [Sep 7 05:51:46] VERBOSE[10974] chan_sip.c: REGISTER 11 headers, 0 lines [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: REGISTER attempt 1 to 17474987674@proxy01.sipphone.com [Sep 7 05:51:46] VERBOSE[10974] chan_sip.c: Reliably Transmitting (NAT) to 198.65.166.131:5060: REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2c13857d;rport Max-Forwards: 70 From: ;tag=as39ec3c05 To: Call-ID: 4c76c12233a7c5d546609dc16317a534@192.168.0.1 CSeq: 708 REGISTER User-Agent: Asterisk PBX SVN-trunk-r285058 Authorization: Digest username="17474987674", realm="proxy01.sipphone.com", algorithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4c8546a59d9e2dacbc828f7251404506a52ba311", response="2b10af385cb0e1e0a8c3b72009f782ec" Expires: 120 Contact: Content-Length: 0 --- [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 0 [ 41]: REGISTER sip:proxy01.sipphone.com SIP/2.0 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2c13857d;rport [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 3 [ 59]: From: ;tag=as39ec3c05 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 4 [ 42]: To: [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 4c76c12233a7c5d546609dc16317a534@192.168.0.1 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 6 [ 18]: CSeq: 708 REGISTER [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 8 [216]: Authorization: Digest username="17474987674", realm="proxy01.sipphone.com", algorithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4c8546a59d9e2dacbc828f7251404506a52ba311", response="2b10af385cb0e1e0a8c3b72009f782ec" [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 10 [ 46]: Contact: [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 12 [ 0]: [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16744 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 198.65.166.131:5060 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: SIP Registry proxy01.sipphone.com: refcount now 3 [Sep 7 05:51:46] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:198.65.166.131:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2c13857d;rport=5060 From: ;tag=as39ec3c05 To: ;tag=92390300a369f0d75803e369c733575e.e901 Call-ID: 4c76c12233a7c5d546609dc16317a534@192.168.0.1 CSeq: 708 REGISTER Contact: ;expires=120 Content-Length: 0 <-------------> [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2c13857d;rport=5060 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 2 [ 59]: From: ;tag=as39ec3c05 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 3 [ 84]: To: ;tag=92390300a369f0d75803e369c733575e.e901 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 4c76c12233a7c5d546609dc16317a534@192.168.0.1 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 5 [ 18]: CSeq: 708 REGISTER [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 6 [ 58]: Contact: ;expires=120 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Sep 7 05:51:46] VERBOSE[10974] chan_sip.c: --- (8 headers 0 lines) --- [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: = Looking for Call ID: 4c76c12233a7c5d546609dc16317a534@192.168.0.1 (Checking To) --From tag as39ec3c05 --To-tag 92390300a369f0d75803e369c733575e.e901 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16744 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Stopping retransmission on '4c76c12233a7c5d546609dc16317a534@192.168.0.1' of Request 708: Match Found [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Registration successful [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: Cancelling timeout 16743 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: SIP Registry proxy01.sipphone.com: refcount now 2 [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: SIP Registry proxy01.sipphone.com: refcount now 1 [Sep 7 05:51:46] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '4c76c12233a7c5d546609dc16317a534@192.168.0.1' in 32000 ms (Method: REGISTER) [Sep 7 05:51:46] NOTICE[10974] chan_sip.c: Outbound Registration: Expiry for proxy01.sipphone.com is 120 sec (Scheduling reregistration in 105 s) [Sep 7 05:51:46] DEBUG[10974] chan_sip.c: SIP Registry proxy01.sipphone.com: refcount now 2 [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: Auto destroying SIP dialog '1834435539@192_168_1_150' [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: Destroying SIP dialog 1834435539@192_168_1_150 [Sep 7 05:51:50] VERBOSE[10974] chan_sip.c: Really destroying SIP dialog '1834435539@192_168_1_150' Method: REGISTER [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: ---------- SIP HISTORY for '1834435539@192_168_1_150' [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: * SIP Call [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 001. Rx REGISTER / 743 REGISTER / sip:home.smartbyte.de [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 002. TxResp SIP/2.0 / 743 REGISTER - 100 Trying [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 003. AuthChal Auth challenge sent for - nc 0 [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 004. TxResp SIP/2.0 / 743 REGISTER - 401 Unauthorized [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 005. SchedDestroy 32000 ms [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 006. RegRequest Succeeded : Account "Kati Krebs" [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 007. Rx REGISTER / 744 REGISTER / sip:home.smartbyte.de [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 008. TxResp SIP/2.0 / 744 REGISTER - 100 Trying [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 009. AuthOK Auth challenge successful for 100 [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 010. CancelDestroy [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 011. TxResp SIP/2.0 / 744 REGISTER - 200 OK [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 012. RegRequest Succeeded : Account "Kati Krebs" [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 013. SchedDestroy 32000 ms [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: 014. AutoDestroy 1834435539@192_168_1_150 [Sep 7 05:51:50] DEBUG[10974] chan_sip.c: ---------- END SIP HISTORY for '1834435539@192_168_1_150' [Sep 7 05:51:52] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.150:5064 ---> INVITE sip:19004018881000@home.smartbyte.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK4478b86e62f26e5f6d86a3f4f42a70cf;rport From: "Kati Krebs" ;tag=843715617 To: Call-ID: 3011425442@192_168_1_150 CSeq: 2 INVITE Contact: Max-Forwards: 70 User-Agent: S675IP022230000000 Supported: replaces Allow-Events: message-summary, refer Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 369 v=0 o=100 5014 6 IN IP4 192.168.1.150 s=Mapping c=IN IP4 192.168.1.150 t=0 0 m=audio 5014 RTP/AVP 9 0 8 96 97 2 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 G726-32/8000 a=rtpmap:97 AAL2-G726-32/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 0 [ 62]: INVITE sip:19004018881000@home.smartbyte.de;user=phone SIP/2.0 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK4478b86e62f26e5f6d86a3f4f42a70cf;rport [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 2 [ 60]: From: "Kati Krebs" ;tag=843715617 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 3 [ 53]: To: [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 6 [ 37]: Contact: [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 8 [ 30]: User-Agent: S675IP022230000000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 9 [ 19]: Supported: replaces [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 10 [ 36]: Allow-Events: message-summary, refer [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 11 [ 72]: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 13 [ 19]: Content-Length: 369 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 14 [ 0]: [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 1 [ 33]: o=100 5014 6 IN IP4 192.168.1.150 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 2 [ 9]: s=Mapping [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.1.150 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 5 [ 41]: m=audio 5014 RTP/AVP 9 0 8 96 97 2 18 101 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 6 [ 20]: a=rtpmap:9 G722/8000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 9 [ 24]: a=rtpmap:96 G726-32/8000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 10 [ 29]: a=rtpmap:97 AAL2-G726-32/8000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 11 [ 23]: a=rtpmap:2 G726-32/8000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 12 [ 21]: a=rtpmap:18 G729/8000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 13 [ 19]: a=fmtp:18 annexb=no [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-16 [Sep 7 05:51:52] VERBOSE[10974] chan_sip.c: --- (14 headers 16 lines) --- [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3011425442@192_168_1_150 (Checking From) --From tag 843715617 --To-tag [Sep 7 05:51:52] DEBUG[10974] acl.c: For destination '192.168.1.150', our source address is '192.168.1.1'. [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.1:5060 [Sep 7 05:51:52] VERBOSE[10974] netsock.c: == Using UDPTL TOS bits 184 [Sep 7 05:51:52] VERBOSE[10974] netsock.c: == Using UDPTL CoS mark 5 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 3011425442@192_168_1_150 - INVITE (No RTP) [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Sep 7 05:51:52] DEBUG[10974] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces" [Sep 7 05:51:52] DEBUG[10974] sip/reqresp_parser.c: Found SIP option: -replaces- [Sep 7 05:51:52] DEBUG[10974] sip/reqresp_parser.c: Matched SIP option: replaces [Sep 7 05:51:52] DEBUG[10974] netsock2.c: Splitting '192.168.1.150:5064' gives... [Sep 7 05:51:52] DEBUG[10974] netsock2.c: ...host '192.168.1.150' and port '5064'. [Sep 7 05:51:52] VERBOSE[10974] chan_sip.c: Sending to 192.168.1.150:5064 (NAT) [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Initializing initreq for method INVITE - callid 3011425442@192_168_1_150 [Sep 7 05:51:52] VERBOSE[10974] chan_sip.c: Using INVITE request as basis request - 3011425442@192_168_1_150 [Sep 7 05:51:52] VERBOSE[10974] chan_sip.c: Found peer '100' for '100' from 192.168.1.150:5064 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:51:52] VERBOSE[10974] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.1.150:5064 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK4478b86e62f26e5f6d86a3f4f42a70cf;received=192.168.1.150;rport=5064 From: "Kati Krebs" ;tag=843715617 To: ;tag=as0884cb4b Call-ID: 3011425442@192_168_1_150 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="home.smartbyte.de", nonce="7ce58b13" Content-Length: 0 <------------> [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 1 [116]: Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK4478b86e62f26e5f6d86a3f4f42a70cf;received=192.168.1.150;rport=5064 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 2 [ 60]: From: "Kati Krebs" ;tag=843715617 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 3 [ 68]: To: ;tag=as0884cb4b [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 9 [ 83]: WWW-Authenticate: Digest algorithm=MD5, realm="home.smartbyte.de", nonce="7ce58b13" [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Header 11 [ 0]: [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16747 [Sep 7 05:51:52] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:51:52] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '3011425442@192_168_1_150' in 32000 ms (Method: INVITE) [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.150:5064 ---> ACK sip:19004018881000@home.smartbyte.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK4478b86e62f26e5f6d86a3f4f42a70cf;rport From: "Kati Krebs" ;tag=843715617 To: ;tag=as0884cb4b Call-ID: 3011425442@192_168_1_150 CSeq: 2 ACK Contact: Max-Forwards: 70 User-Agent: S675IP022230000000 Content-Length: 0 <-------------> [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 0 [ 59]: ACK sip:19004018881000@home.smartbyte.de;user=phone SIP/2.0 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK4478b86e62f26e5f6d86a3f4f42a70cf;rport [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 2 [ 60]: From: "Kati Krebs" ;tag=843715617 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 3 [ 68]: To: ;tag=as0884cb4b [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 6 [ 37]: Contact: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 8 [ 30]: User-Agent: S675IP022230000000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3011425442@192_168_1_150 (Checking From) --From tag 843715617 --To-tag as0884cb4b [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16747 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Stopping retransmission on '3011425442@192_168_1_150' of Response 2: Match Found [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.150:5064 ---> INVITE sip:19004018881000@home.smartbyte.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK78c43f164b5e647a2b58d1c592cafe58;rport From: "Kati Krebs" ;tag=843715617 To: Call-ID: 3011425442@192_168_1_150 CSeq: 3 INVITE Contact: Authorization: Digest username="100", realm="home.smartbyte.de", algorithm=MD5, uri="sip:19004018881000@home.smartbyte.de;user=phone", nonce="7ce58b13", response="ec80490cee666bb1942bb19eb179f162" Max-Forwards: 70 User-Agent: S675IP022230000000 Supported: replaces Allow-Events: message-summary, refer Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 369 v=0 o=100 5014 6 IN IP4 192.168.1.150 s=Mapping c=IN IP4 192.168.1.150 t=0 0 m=audio 5014 RTP/AVP 9 0 8 96 97 2 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 G726-32/8000 a=rtpmap:97 AAL2-G726-32/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 0 [ 62]: INVITE sip:19004018881000@home.smartbyte.de;user=phone SIP/2.0 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK78c43f164b5e647a2b58d1c592cafe58;rport [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 2 [ 60]: From: "Kati Krebs" ;tag=843715617 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 3 [ 53]: To: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 5 [ 14]: CSeq: 3 INVITE [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 6 [ 37]: Contact: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 7 [196]: Authorization: Digest username="100", realm="home.smartbyte.de", algorithm=MD5, uri="sip:19004018881000@home.smartbyte.de;user=phone", nonce="7ce58b13", response="ec80490cee666bb1942bb19eb179f162" [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 9 [ 30]: User-Agent: S675IP022230000000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 10 [ 19]: Supported: replaces [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 11 [ 36]: Allow-Events: message-summary, refer [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 14 [ 19]: Content-Length: 369 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 15 [ 0]: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 1 [ 33]: o=100 5014 6 IN IP4 192.168.1.150 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 2 [ 9]: s=Mapping [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.1.150 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 5 [ 41]: m=audio 5014 RTP/AVP 9 0 8 96 97 2 18 101 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 6 [ 20]: a=rtpmap:9 G722/8000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 9 [ 24]: a=rtpmap:96 G726-32/8000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 10 [ 29]: a=rtpmap:97 AAL2-G726-32/8000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 11 [ 23]: a=rtpmap:2 G726-32/8000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 12 [ 21]: a=rtpmap:18 G729/8000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 13 [ 19]: a=fmtp:18 annexb=no [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-16 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: --- (15 headers 16 lines) --- [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3011425442@192_168_1_150 (Checking From) --From tag 843715617 --To-tag [Sep 7 05:51:53] DEBUG[10974] netsock2.c: Splitting 'home.smartbyte.de' gives... [Sep 7 05:51:53] DEBUG[10974] netsock2.c: ...host 'home.smartbyte.de' and port '(null)'. [Sep 7 05:51:53] DEBUG[10974] netsock2.c: Splitting 'home.smartbyte.de' gives... [Sep 7 05:51:53] DEBUG[10974] netsock2.c: ...host 'home.smartbyte.de' and port '(null)'. [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Sep 7 05:51:53] DEBUG[10974] netsock2.c: Splitting '192.168.1.150:5064' gives... [Sep 7 05:51:53] DEBUG[10974] netsock2.c: ...host '192.168.1.150' and port '5064'. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Sending to 192.168.1.150:5064 (NAT) [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Initializing initreq for method INVITE - callid 3011425442@192_168_1_150 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Using INVITE request as basis request - 3011425442@192_168_1_150 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found peer '100' for '100' from 192.168.1.150:5064 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x429f9170' [Sep 7 05:51:53] DEBUG[10974] res_rtp_asterisk.c: Allocated port 10060 for RTP instance '0x429f9170' [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: RTP instance '0x429f9170' is setup and ready to go [Sep 7 05:51:53] DEBUG[10974] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x429f9170' [Sep 7 05:51:53] VERBOSE[10974] netsock2.c: == Using SIP RTP TOS bits 184 [Sep 7 05:51:53] VERBOSE[10974] netsock2.c: == Using SIP RTP CoS mark 5 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Setting NAT on RTP to On [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing session-level SDP o=100 5014 6 IN IP4 192.168.1.150... UNSUPPORTED. [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing session-level SDP s=Mapping... UNSUPPORTED. [Sep 7 05:51:53] DEBUG[10974] netsock2.c: Splitting '192.168.1.150' gives... [Sep 7 05:51:53] DEBUG[10974] netsock2.c: ...host '192.168.1.150' and port '(null)'. [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.150... OK. [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found RTP audio format 9 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Setting payload 9 based on m type on 0x41a31030 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found RTP audio format 0 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Setting payload 0 based on m type on 0x41a31030 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found RTP audio format 8 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Setting payload 8 based on m type on 0x41a31030 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found RTP audio format 96 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found RTP audio format 97 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Setting payload 97 based on m type on 0x41a31030 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found RTP audio format 2 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found RTP audio format 18 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Setting payload 18 based on m type on 0x41a31030 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found RTP audio format 101 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Setting payload 101 based on m type on 0x41a31030 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found audio description format G722 for ID 9 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found audio description format PCMU for ID 0 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found audio description format PCMA for ID 8 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found audio description format G726-32 for ID 96 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-32/8000... OK. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found audio description format AAL2-G726-32 for ID 97 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 AAL2-G726-32/8000... OK. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found audio description format G726-32 for ID 2 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found audio description format G729 for ID 18 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Found audio description format telephone-event for ID 101 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Incorporating payload 0 on 0x41a31030 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Incorporating payload 2 on 0x41a31030 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Incorporating payload 8 on 0x41a31030 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Incorporating payload 9 on 0x41a31030 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Incorporating payload 18 on 0x41a31030 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Incorporating payload 96 on 0x41a31030 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Incorporating payload 97 on 0x41a31030 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Incorporating payload 101 on 0x41a31030 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Capabilities: us - 0x110e (gsm|ulaw|alaw|g729|g722), peer - audio=0x191c (ulaw|alaw|g726|g729|g726aal2|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x110c (ulaw|alaw|g729|g722) [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 7 05:51:53] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x429f9170' [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Peer audio RTP is at port 192.168.1.150:5014 [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Copying payload 0 from 0x41a31030 to 0x429f931c [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Copying payload 2 from 0x41a31030 to 0x429f931c [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Copying payload 8 from 0x41a31030 to 0x429f931c [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Copying payload 9 from 0x41a31030 to 0x429f931c [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Copying payload 18 from 0x41a31030 to 0x429f931c [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Copying payload 96 from 0x41a31030 to 0x429f931c [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Copying payload 97 from 0x41a31030 to 0x429f931c [Sep 7 05:51:53] DEBUG[10974] rtp_engine.c: Copying payload 101 from 0x41a31030 to 0x429f931c [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Peer doesn't provide T.38 UDPTL [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: We're settling with these formats: 0x110c (ulaw|alaw|g729|g722) [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Checking SIP call limits for device 100 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Updating call counter for incoming call [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: Looking for 19004018881000 in default (domain home.smartbyte.de) [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: *** Our native formats are 0x8 (alaw) [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: *** Joint capabilities are 0x110c (ulaw|alaw|g729|g722) [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: This channel will not be able to handle video. [Sep 7 05:51:53] DEBUG[10974] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Sep 7 05:51:53] DEBUG[10974] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: build_route: Contact hop: [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: list_route: hop: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: SIP/100-0000001b: New call is still down.... Trying... [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.150:5064 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK78c43f164b5e647a2b58d1c592cafe58;received=192.168.1.150;rport=5064 From: "Kati Krebs" ;tag=843715617 To: Call-ID: 3011425442@192_168_1_150 CSeq: 3 INVITE Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 1 [116]: Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK78c43f164b5e647a2b58d1c592cafe58;received=192.168.1.150;rport=5064 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 2 [ 60]: From: "Kati Krebs" ;tag=843715617 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 3 [ 53]: To: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 5 [ 14]: CSeq: 3 INVITE [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 9 [ 46]: Contact: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 11 [ 0]: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:51:53] DEBUG[10957] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Sep 7 05:51:53] DEBUG[10957] chan_sip.c: Checking device state for peer 100 [Sep 7 05:51:53] DEBUG[10957] devicestate.c: Changing state for SIP/100 - state 1 (Not in use) [Sep 7 05:51:53] DEBUG[10957] devicestate.c: device 'SIP/100' state '1' [Sep 7 05:51:53] DEBUG[11003] app_queue.c: Device 'SIP/100' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 7 05:51:53] DEBUG[18272] pbx.c: Launching 'Set' [Sep 7 05:51:53] VERBOSE[18272] pbx.c: -- Executing [19004018881000@default:1] Set("SIP/100-0000001b", "CALLERID(num)=05516345347") in new stack [Sep 7 05:51:53] DEBUG[18272] pbx.c: Result of 'EXTEN' is '19004018881000' [Sep 7 05:51:53] DEBUG[18272] pbx.c: Launching 'Set' [Sep 7 05:51:53] VERBOSE[18272] pbx.c: -- Executing [19004018881000@default:2] Set("SIP/100-0000001b", "CDR(UserField)=04018881000") in new stack [Sep 7 05:51:53] DEBUG[18272] pbx.c: Result of 'EXTEN' is '19004018881000' [Sep 7 05:51:53] DEBUG[18272] pbx.c: Launching 'Dial' [Sep 7 05:51:53] VERBOSE[18272] pbx.c: -- Executing [19004018881000@default:3] Dial("SIP/100-0000001b", "SIP/04018881000@smartbyte.de") in new stack [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Sep 7 05:51:53] VERBOSE[18272] netsock.c: == Using UDPTL TOS bits 184 [Sep 7 05:51:53] VERBOSE[18272] netsock.c: == Using UDPTL CoS mark 5 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Allocating new SIP dialog for 419a7f41628335d2565558b61ff74cbc@192.168.0.1:0 - INVITE (No RTP) [Sep 7 05:51:53] DEBUG[18272] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xe65e400' [Sep 7 05:51:53] DEBUG[18272] res_rtp_asterisk.c: Allocated port 10660 for RTP instance '0xe65e400' [Sep 7 05:51:53] DEBUG[18272] rtp_engine.c: RTP instance '0xe65e400' is setup and ready to go [Sep 7 05:51:53] DEBUG[18272] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xe66afd8' [Sep 7 05:51:53] DEBUG[18272] res_rtp_asterisk.c: Allocated port 10764 for RTP instance '0xe66afd8' [Sep 7 05:51:53] DEBUG[18272] rtp_engine.c: RTP instance '0xe66afd8' is setup and ready to go [Sep 7 05:51:53] DEBUG[18272] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xe66afd8' [Sep 7 05:51:53] DEBUG[18272] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xe65e400' [Sep 7 05:51:53] VERBOSE[18272] netsock2.c: == Using SIP RTP TOS bits 184 [Sep 7 05:51:53] VERBOSE[18272] netsock2.c: == Using SIP RTP CoS mark 5 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Setting NAT on RTP to On [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Setting NAT on VRTP to On [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Sep 7 05:51:53] DEBUG[18272] acl.c: For destination '80.237.160.128', our source address is '192.168.1.1'. [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Target address 80.237.160.128:5060 is not local, substituting externaddr [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 114.77.232.125:5060 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: *** Our native formats are 0x8 (alaw) [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: *** Our capabilities are 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719) [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Sep 7 05:51:53] DEBUG[18272] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Sep 7 05:51:53] DEBUG[18272] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Sep 7 05:51:53] DEBUG[18272] rtp_engine.c: Seeded SDP of 'SIP/smartbyte.de-0000001c' with that of 'SIP/100-0000001b' [Sep 7 05:51:53] DEBUG[18272] channel.c: Not copying variable DIALEDTIME. [Sep 7 05:51:53] DEBUG[18272] channel.c: Not copying variable ANSWEREDTIME. [Sep 7 05:51:53] DEBUG[18272] channel.c: Not copying variable DIALEDPEERNAME. [Sep 7 05:51:53] DEBUG[18272] channel.c: Not copying variable DIALEDPEERNUMBER. [Sep 7 05:51:53] DEBUG[18272] channel.c: Not copying variable DIALSTATUS. [Sep 7 05:51:53] DEBUG[18272] channel.c: Not copying variable SIPCALLID. [Sep 7 05:51:53] DEBUG[18272] channel.c: Not copying variable SIPDOMAIN. [Sep 7 05:51:53] DEBUG[18272] channel.c: Not copying variable SIPURI. [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Outgoing Call for 04018881000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Updating call counter for outgoing call [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: This call needs video offers! [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: We think we can do text [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: This call needs text offers, but there's no text support enabled ! [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: ** Our capability: 0x80020c7f9efe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|testlaw) Video flag: False Text flag: False [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Audio is at 5060 [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Video is at 114.77.232.125:5060 [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x400 (ilbc) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x80 (lpc10) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x40 (slin) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x10 (g726aal2) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x20 (adpcm) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x200 (speex) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x800 (g726) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x1000 (g722) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x8000 (slin16) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding video codec 0x10000 (jpeg) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding video codec 0x20000 (png) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding video codec 0x40000 (h261) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding video codec 0x80000 (h263) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding video codec 0x200000 (h264) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding video codec 0x400000 (mpeg4) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x200000000 (speex16) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: -- Done with adding codecs to SDP [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Done building SDP. Settling with this capability: 0x80020c7f9efe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|testlaw) [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Initializing initreq for method INVITE - callid 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 0 [ 45]: INVITE sip:04018881000@80.237.160.128 SIP/2.0 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;rport [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 3 [ 66]: From: "Kati Krebs" ;tag=as6cafe189 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 4 [ 36]: To: [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 5 [ 46]: Contact: [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 6 [ 61]: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 9 [ 35]: Date: Mon, 06 Sep 2010 19:51:53 GMT [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 12 [ 98]: Remote-Party-ID: "Kati Krebs" ;party=calling;privacy=off;screen=no [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Sep 7 05:51:53] VERBOSE[18272] chan_sip.c: Reliably Transmitting (NAT) to 80.237.160.128:5060: INVITE sip:04018881000@80.237.160.128 SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;rport Max-Forwards: 70 From: "Kati Krebs" ;tag=as6cafe189 To: Contact: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r285058 Date: Mon, 06 Sep 2010 19:51:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "Kati Krebs" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 809 v=0 o=root 1231544964 1231544964 IN IP4 114.77.232.125 s=Asterisk PBX SVN-trunk-r285058 c=IN IP4 114.77.232.125 b=CT:384 t=0 0 m=audio 10660 RTP/AVP 8 0 97 3 7 10 112 5 110 111 9 118 117 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:7 LPC/8000 a=rtpmap:10 L16/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:118 L16/16000 a=rtpmap:117 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 10764 RTP/AVP 26 31 34 98 99 104 a=rtpmap:26 JPEG/90000 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=rtpmap:104 MP4V-ES/90000 a=sendrecv --- [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 0 [ 45]: INVITE sip:04018881000@80.237.160.128 SIP/2.0 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;rport [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 3 [ 66]: From: "Kati Krebs" ;tag=as6cafe189 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 4 [ 36]: To: [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 5 [ 46]: Contact: [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 6 [ 61]: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 9 [ 35]: Date: Mon, 06 Sep 2010 19:51:53 GMT [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 12 [ 98]: Remote-Party-ID: "Kati Krebs" ;party=calling;privacy=off;screen=no [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 14 [ 19]: Content-Length: 809 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Header 15 [ 0]: [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 1 [ 50]: o=root 1231544964 1231544964 IN IP4 114.77.232.125 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 2 [ 32]: s=Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 3 [ 23]: c=IN IP4 114.77.232.125 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 4 [ 8]: b=CT:384 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 5 [ 5]: t=0 0 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 6 [ 63]: m=audio 10660 RTP/AVP 8 0 97 3 7 10 112 5 110 111 9 118 117 101 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 9 [ 21]: a=rtpmap:97 iLBC/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 10 [ 17]: a=fmtp:97 mode=30 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 11 [ 19]: a=rtpmap:3 GSM/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 12 [ 19]: a=rtpmap:7 LPC/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 13 [ 20]: a=rtpmap:10 L16/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 14 [ 30]: a=rtpmap:112 AAL2-G726-32/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 15 [ 20]: a=rtpmap:5 DVI4/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 16 [ 23]: a=rtpmap:110 speex/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 17 [ 25]: a=rtpmap:111 G726-32/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 19 [ 22]: a=rtpmap:118 L16/16000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 20 [ 24]: a=rtpmap:117 speex/16000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 21 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 22 [ 15]: a=fmtp:101 0-16 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 23 [ 10]: a=ptime:20 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 24 [ 10]: a=sendrecv [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 25 [ 40]: m=video 10764 RTP/AVP 26 31 34 98 99 104 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 26 [ 22]: a=rtpmap:26 JPEG/90000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 27 [ 22]: a=rtpmap:31 H261/90000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 28 [ 22]: a=rtpmap:34 H263/90000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 29 [ 27]: a=rtpmap:98 h263-1998/90000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 30 [ 22]: a=rtpmap:99 H264/90000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 31 [ 26]: a=rtpmap:104 MP4V-ES/90000 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Body 32 [ 10]: a=sendrecv [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16750 [Sep 7 05:51:53] DEBUG[18272] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 80.237.160.128:5060 [Sep 7 05:51:53] VERBOSE[18272] app_dial.c: -- Called 04018881000@smartbyte.de [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;received=114.77.232.125;rport=5060 From: "Kati Krebs" ;tag=as6cafe189 To: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 CSeq: 102 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;received=114.77.232.125;rport=5060 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 2 [ 66]: From: "Kati Krebs" ;tag=as6cafe189 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 3 [ 36]: To: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 4 [ 61]: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 9 [ 14]: Require: timer [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 11 [ 46]: Contact: [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Sep 7 05:51:53] VERBOSE[10974] chan_sip.c: --- (13 headers 0 lines) --- [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: = Looking for Call ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 (Checking To) --From tag as6cafe189 --To-tag [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: *** SIP TIMER: Cancelling retransmission #16750 - INVITE (got response) [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Request 102: Found [Sep 7 05:51:53] DEBUG[10974] chan_sip.c: SIP response 100 to standard invite [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: Auto destroying SIP dialog '07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060' [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: Destroying SIP dialog 07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060 [Sep 7 05:51:54] VERBOSE[10974] chan_sip.c: Really destroying SIP dialog '07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060' Method: OPTIONS [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: ---------- SIP HISTORY for '07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060' [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: * SIP Call [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: 001. Rx OPTIONS / 102 OPTIONS / sip:100@114.77.232.125:5060 [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: 002. TxResp SIP/2.0 / 102 OPTIONS - 200 OK [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: 003. SchedDestroy 32000 ms [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: 004. AutoDestroy 07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060 [Sep 7 05:51:54] DEBUG[10974] chan_sip.c: ---------- END SIP HISTORY for '07fb88400b09f55f39f4334b567022e8@80.237.160.128:5060' [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;received=114.77.232.125;rport=5060 From: "Kati Krebs" ;tag=as6cafe189 To: ;tag=as299e4ad4 Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 CSeq: 102 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 447 v=0 o=root 1673120505 1673120505 IN IP4 80.237.160.128 s=Asterisk PBX SVN--r c=IN IP4 80.237.160.128 t=0 0 m=audio 9530 RTP/AVP 8 0 3 111 97 7 10 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:7 LPC/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 0 RTP/AVP 26 31 34 98 99 104 <-------------> [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 0 [ 28]: SIP/2.0 183 Session Progress [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;received=114.77.232.125;rport=5060 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 2 [ 66]: From: "Kati Krebs" ;tag=as6cafe189 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 3 [ 51]: To: ;tag=as299e4ad4 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 4 [ 61]: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 9 [ 14]: Require: timer [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 11 [ 46]: Contact: [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 13 [ 19]: Content-Length: 447 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 14 [ 0]: [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 1 [ 50]: o=root 1673120505 1673120505 IN IP4 80.237.160.128 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 2 [ 21]: s=Asterisk PBX SVN--r [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 3 [ 23]: c=IN IP4 80.237.160.128 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 5 [ 42]: m=audio 9530 RTP/AVP 8 0 3 111 97 7 10 101 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 9 [ 25]: a=rtpmap:111 G726-32/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 10 [ 21]: a=rtpmap:97 iLBC/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 11 [ 17]: a=fmtp:97 mode=30 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 12 [ 19]: a=rtpmap:7 LPC/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 13 [ 20]: a=rtpmap:10 L16/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-16 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 16 [ 10]: a=ptime:20 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 17 [ 10]: a=sendrecv [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 18 [ 36]: m=video 0 RTP/AVP 26 31 34 98 99 104 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: --- (14 headers 19 lines) --- [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: = Looking for Call ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 (Checking To) --From tag as6cafe189 --To-tag as299e4ad4 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Request 102: Found [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: SIP response 183 to standard invite [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP o=root 1673120505 1673120505 IN IP4 80.237.160.128... UNSUPPORTED. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP s=Asterisk PBX SVN--r... UNSUPPORTED. [Sep 7 05:51:55] DEBUG[10974] netsock2.c: Splitting '80.237.160.128' gives... [Sep 7 05:51:55] DEBUG[10974] netsock2.c: ...host '80.237.160.128' and port '(null)'. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP c=IN IP4 80.237.160.128... OK. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 8 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 8 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 0 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 3 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 3 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 111 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 111 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 97 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 97 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 7 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 7 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 10 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 10 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 101 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 101 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format PCMA for ID 8 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format PCMU for ID 0 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format GSM for ID 3 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format G726-32 for ID 111 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 G726-32/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format iLBC for ID 97 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=fmtp:97 mode=30... UNSUPPORTED. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format LPC for ID 7 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:7 LPC/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format L16 for ID 10 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:10 L16/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format telephone-event for ID 101 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 26 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 26 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 31 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 31 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 34 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 34 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 98 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 98 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 99 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 99 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 104 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 104 based on m type on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 0 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 3 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 7 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 8 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 10 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 97 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 101 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 111 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 26 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 31 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 34 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 98 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 99 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 104 on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0xcce (gsm|ulaw|alaw|g726|slin|lpc10|ilbc)/video=0x7d0000 (jpeg|h261|h263|h263p|h264|mpeg4)/text=0x0 (nothing), combined - 0x7d0cce (gsm|ulaw|alaw|g726|slin|lpc10|ilbc|jpeg|h261|h263|h263p|h264|mpeg4) [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 7 05:51:55] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xe65e400' [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Peer audio RTP is at port 80.237.160.128:9530 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 0 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 3 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 7 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 8 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 10 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 97 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 101 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 111 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xe66afd8' [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Peer doesn't provide video [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Peer doesn't provide T.38 UDPTL [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: We're settling with these formats: 0x7d0cce (gsm|ulaw|alaw|g726|slin|lpc10|ilbc|jpeg|h261|h263|h263p|h264|mpeg4) [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: We have an owner, now see if we need to change this call [Sep 7 05:51:55] VERBOSE[18272] app_dial.c: -- SIP/smartbyte.de-0000001c is making progress passing it to SIP/100-0000001b [Sep 7 05:51:55] DEBUG[18272] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000001b' with that of 'SIP/smartbyte.de-0000001c' [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Setting framing from config on incoming call [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: ** Our capability: 0x110c (ulaw|alaw|g729|g722) Video flag: True Text flag: True [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Audio is at 5060 [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding codec 0x100 (g729) to SDP [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding codec 0x1000 (g722) to SDP [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: -- Done with adding codecs to SDP [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Done building SDP. Settling with this capability: 0x110c (ulaw|alaw|g729|g722) [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.150:5064 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK78c43f164b5e647a2b58d1c592cafe58;received=192.168.1.150;rport=5064 From: "Kati Krebs" ;tag=843715617 To: ;tag=as14f3f609 Call-ID: 3011425442@192_168_1_150 CSeq: 3 INVITE Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 339 v=0 o=root 1850335413 1850335413 IN IP4 192.168.1.1 s=Asterisk PBX SVN-trunk-r285058 c=IN IP4 192.168.1.1 t=0 0 m=audio 10060 RTP/AVP 8 0 18 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 0 [ 28]: SIP/2.0 183 Session Progress [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 1 [116]: Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK78c43f164b5e647a2b58d1c592cafe58;received=192.168.1.150;rport=5064 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 2 [ 60]: From: "Kati Krebs" ;tag=843715617 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 3 [ 68]: To: ;tag=as14f3f609 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 5 [ 14]: CSeq: 3 INVITE [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 9 [ 46]: Contact: [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 11 [ 19]: Content-Length: 339 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 12 [ 0]: [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 1 [ 47]: o=root 1850335413 1850335413 IN IP4 192.168.1.1 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 2 [ 32]: s=Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 3 [ 20]: c=IN IP4 192.168.1.1 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 5 [ 34]: m=audio 10060 RTP/AVP 8 0 18 9 101 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-16 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 13 [ 10]: a=ptime:20 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 14 [ 10]: a=sendrecv [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:51:55] DEBUG[18272] dsp.c: tone 1100, Ew=1.22E+04, Et=1.97E+06, s/n= 0.01 [Sep 7 05:51:55] DEBUG[18272] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Sep 7 05:51:55] DEBUG[18272] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Sep 7 05:51:55] DEBUG[18272] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x429f9170' [Sep 7 05:51:55] DEBUG[18272] dsp.c: tone 1100, Ew=1.60E+01, Et=1.64E+06, s/n= 0.00 [Sep 7 05:51:55] DEBUG[18272] dsp.c: tone 1100, Ew=1.60E+01, Et=1.64E+06, s/n= 0.00 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;received=114.77.232.125;rport=5060 From: "Kati Krebs" ;tag=as6cafe189 To: ;tag=as299e4ad4 Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 CSeq: 102 INVITE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 447 v=0 o=root 1673120505 1673120506 IN IP4 80.237.160.128 s=Asterisk PBX SVN--r c=IN IP4 80.237.160.128 t=0 0 m=audio 9530 RTP/AVP 8 0 3 111 97 7 10 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:7 LPC/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 0 RTP/AVP 26 31 34 98 99 104 <-------------> [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK654f22f5;received=114.77.232.125;rport=5060 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 2 [ 66]: From: "Kati Krebs" ;tag=as6cafe189 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 3 [ 51]: To: ;tag=as299e4ad4 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 4 [ 61]: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 9 [ 14]: Require: timer [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 11 [ 46]: Contact: [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 13 [ 19]: Content-Length: 447 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 14 [ 0]: [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 1 [ 50]: o=root 1673120505 1673120506 IN IP4 80.237.160.128 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 2 [ 21]: s=Asterisk PBX SVN--r [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 3 [ 23]: c=IN IP4 80.237.160.128 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 5 [ 42]: m=audio 9530 RTP/AVP 8 0 3 111 97 7 10 101 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 9 [ 25]: a=rtpmap:111 G726-32/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 10 [ 21]: a=rtpmap:97 iLBC/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 11 [ 17]: a=fmtp:97 mode=30 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 12 [ 19]: a=rtpmap:7 LPC/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 13 [ 20]: a=rtpmap:10 L16/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-16 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 16 [ 10]: a=ptime:20 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 17 [ 10]: a=sendrecv [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Body 18 [ 36]: m=video 0 RTP/AVP 26 31 34 98 99 104 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: --- (14 headers 19 lines) --- [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: = Looking for Call ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 (Checking To) --From tag as6cafe189 --To-tag as299e4ad4 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Acked pending invite 102 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Stopping retransmission on '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' of Request 102: Match Found [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: SIP response 200 to standard invite [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP o=root 1673120505 1673120506 IN IP4 80.237.160.128... UNSUPPORTED. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP s=Asterisk PBX SVN--r... UNSUPPORTED. [Sep 7 05:51:55] DEBUG[10974] netsock2.c: Splitting '80.237.160.128' gives... [Sep 7 05:51:55] DEBUG[10974] netsock2.c: ...host '80.237.160.128' and port '(null)'. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP c=IN IP4 80.237.160.128... OK. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 8 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 8 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 0 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 3 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 3 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 111 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 111 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 97 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 97 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 7 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 7 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 10 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 10 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP audio format 101 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 101 based on m type on 0x41a31630 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format PCMA for ID 8 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format PCMU for ID 0 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format GSM for ID 3 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format G726-32 for ID 111 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 G726-32/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format iLBC for ID 97 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=fmtp:97 mode=30... UNSUPPORTED. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format LPC for ID 7 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:7 LPC/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format L16 for ID 10 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:10 L16/8000... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found audio description format telephone-event for ID 101 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 26 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 26 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 31 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 31 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 34 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 34 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 98 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 98 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 99 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 99 based on m type on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Found RTP video format 104 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Setting payload 104 based on m type on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 0 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 3 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 7 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 8 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 10 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 97 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 101 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 111 on 0x41a31630 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 26 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 31 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 34 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 98 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 99 on 0x41a309b0 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Incorporating payload 104 on 0x41a309b0 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0xcce (gsm|ulaw|alaw|g726|slin|lpc10|ilbc)/video=0x7d0000 (jpeg|h261|h263|h263p|h264|mpeg4)/text=0x0 (nothing), combined - 0x7d0cce (gsm|ulaw|alaw|g726|slin|lpc10|ilbc|jpeg|h261|h263|h263p|h264|mpeg4) [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 7 05:51:55] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xe65e400' [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Peer audio RTP is at port 80.237.160.128:9530 [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 0 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 3 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 7 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 8 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 10 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 97 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 101 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] rtp_engine.c: Copying payload 111 from 0x41a31630 to 0xe65e5ac [Sep 7 05:51:55] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xe66afd8' [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Peer doesn't provide video [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Peer doesn't provide T.38 UDPTL [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: We're settling with these formats: 0x7d0cce (gsm|ulaw|alaw|g726|slin|lpc10|ilbc|jpeg|h261|h263|h263p|h264|mpeg4) [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: We have an owner, now see if we need to change this call [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Updating call counter for outgoing call [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: build_route: Contact hop: [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: list_route: hop: [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Session-Expires: 1800 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Refresher: UAS [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Session timer started: 16755 - 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Strict routing enforced for session 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: set_destination: Parsing for address/port to send to [Sep 7 05:51:55] DEBUG[10974] netsock2.c: Splitting '80.237.160.128:5060' gives... [Sep 7 05:51:55] DEBUG[10974] netsock2.c: ...host '80.237.160.128' and port '5060'. [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: set_destination: set destination to 80.237.160.128:5060 [Sep 7 05:51:55] VERBOSE[10974] chan_sip.c: Transmitting (NAT) to 80.237.160.128:5060: ACK sip:04018881000@80.237.160.128:5060 SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK266ddff0;rport Max-Forwards: 70 From: "Kati Krebs" ;tag=as6cafe189 To: ;tag=as299e4ad4 Contact: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r285058 Content-Length: 0 --- [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 0 [ 47]: ACK sip:04018881000@80.237.160.128:5060 SIP/2.0 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK266ddff0;rport [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 3 [ 66]: From: "Kati Krebs" ;tag=as6cafe189 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 4 [ 51]: To: ;tag=as299e4ad4 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 5 [ 46]: Contact: [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 6 [ 61]: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 8 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Header 10 [ 0]: [Sep 7 05:51:55] DEBUG[10974] chan_sip.c: Trying to put 'ACK sip:040' onto UDP socket destined for 80.237.160.128:5060 [Sep 7 05:51:55] VERBOSE[18272] app_dial.c: -- SIP/smartbyte.de-0000001c answered SIP/100-0000001b [Sep 7 05:51:55] DEBUG[18272] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000001b' with that of 'SIP/smartbyte.de-0000001c' [Sep 7 05:51:55] DEBUG[10957] devicestate.c: No provider found, checking channel drivers for SIP - smartbyte.de [Sep 7 05:51:55] DEBUG[10957] chan_sip.c: Checking device state for peer smartbyte.de [Sep 7 05:51:55] DEBUG[10957] devicestate.c: Changing state for SIP/smartbyte.de - state 1 (Not in use) [Sep 7 05:51:55] DEBUG[10957] devicestate.c: device 'SIP/smartbyte.de' state '1' [Sep 7 05:51:55] DEBUG[10957] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: SIP answering channel: SIP/100-0000001b [Sep 7 05:51:55] DEBUG[10957] chan_sip.c: Checking device state for peer 100 [Sep 7 05:51:55] DEBUG[18272] res_rtp_asterisk.c: Setting the marker bit due to a source update [Sep 7 05:51:55] DEBUG[10957] devicestate.c: Changing state for SIP/100 - state 1 (Not in use) [Sep 7 05:51:55] DEBUG[10957] devicestate.c: device 'SIP/100' state '1' [Sep 7 05:51:55] DEBUG[11003] app_queue.c: Device 'SIP/smartbyte.de' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Setting framing from config on incoming call [Sep 7 05:51:55] DEBUG[11003] app_queue.c: Device 'SIP/100' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: ** Our capability: 0x110c (ulaw|alaw|g729|g722) Video flag: True Text flag: True [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Audio is at 5060 [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding codec 0x100 (g729) to SDP [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding codec 0x1000 (g722) to SDP [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: -- Done with adding codecs to SDP [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Done building SDP. Settling with this capability: 0x110c (ulaw|alaw|g729|g722) [Sep 7 05:51:55] VERBOSE[18272] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.1.150:5064 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK78c43f164b5e647a2b58d1c592cafe58;received=192.168.1.150;rport=5064 From: "Kati Krebs" ;tag=843715617 To: ;tag=as14f3f609 Call-ID: 3011425442@192_168_1_150 CSeq: 3 INVITE Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 339 v=0 o=root 1850335413 1850335414 IN IP4 192.168.1.1 s=Asterisk PBX SVN-trunk-r285058 c=IN IP4 192.168.1.1 t=0 0 m=audio 10060 RTP/AVP 8 0 18 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 1 [116]: Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bK78c43f164b5e647a2b58d1c592cafe58;received=192.168.1.150;rport=5064 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 2 [ 60]: From: "Kati Krebs" ;tag=843715617 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 3 [ 68]: To: ;tag=as14f3f609 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 5 [ 14]: CSeq: 3 INVITE [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 9 [ 46]: Contact: [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 11 [ 19]: Content-Length: 339 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Header 12 [ 0]: [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 1 [ 47]: o=root 1850335413 1850335414 IN IP4 192.168.1.1 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 2 [ 32]: s=Asterisk PBX SVN-trunk-r285058 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 3 [ 20]: c=IN IP4 192.168.1.1 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 5 [ 34]: m=audio 10060 RTP/AVP 8 0 18 9 101 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-16 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 13 [ 10]: a=ptime:20 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Body 14 [ 10]: a=sendrecv [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16756 [Sep 7 05:51:55] DEBUG[18272] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:51:55] DEBUG[18272] features.c: bridge answer set, chan answer set [Sep 7 05:51:55] DEBUG[18272] res_rtp_asterisk.c: Changing ssrc from 1847110444 to 1873087836 due to a source change [Sep 7 05:51:55] DEBUG[18272] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Sep 7 05:51:55] VERBOSE[18272] rtp_engine.c: -- Locally bridging SIP/100-0000001b and SIP/smartbyte.de-0000001c [Sep 7 05:51:56] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.150:5064 ---> ACK sip:19004018881000@192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bKb5d8b1e6152e9a2c78504e5547931497;rport From: "Kati Krebs" ;tag=843715617 To: ;tag=as14f3f609 Call-ID: 3011425442@192_168_1_150 CSeq: 3 ACK Contact: Authorization: Digest username="100", realm="home.smartbyte.de", algorithm=MD5, uri="sip:19004018881000@home.smartbyte.de;user=phone", nonce="7ce58b13", response="ec80490cee666bb1942bb19eb179f162" Max-Forwards: 70 User-Agent: S675IP022230000000 Content-Length: 0 <-------------> [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 0 [ 47]: ACK sip:19004018881000@192.168.1.1:5060 SIP/2.0 [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 192.168.1.150:5064;branch=z9hG4bKb5d8b1e6152e9a2c78504e5547931497;rport [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 2 [ 60]: From: "Kati Krebs" ;tag=843715617 [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 3 [ 68]: To: ;tag=as14f3f609 [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 5 [ 11]: CSeq: 3 ACK [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 6 [ 37]: Contact: [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 7 [196]: Authorization: Digest username="100", realm="home.smartbyte.de", algorithm=MD5, uri="sip:19004018881000@home.smartbyte.de;user=phone", nonce="7ce58b13", response="ec80490cee666bb1942bb19eb179f162" [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 9 [ 30]: User-Agent: S675IP022230000000 [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:51:56] VERBOSE[10974] chan_sip.c: --- (11 headers 0 lines) --- [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3011425442@192_168_1_150 (Checking From) --From tag 843715617 --To-tag as14f3f609 [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16756 [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Stopping retransmission on '3011425442@192_168_1_150' of Response 3: Match Found [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:51:56] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:51:57] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:51:57] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:51:58] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:51:58] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:51:59] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:51:59] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:00] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:00] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:00] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:00] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:00] DEBUG[18272] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Sep 7 05:52:00] DEBUG[10963] res_jabber.c: XML parsing successful [Sep 7 05:52:01] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:203.12.160.224:5060 ---> OPTIONS sip:s@114.77.232.125:5060 SIP/2.0 Via: SIP/2.0/UDP 203.12.160.224:5060;branch=z9hG4bK613ecb3d;rport From: "asterisk" ;tag=as0a9ea18c To: Contact: Call-ID: 77cb11c325ade06466d4ed6f79feb4f8@203.12.160.224 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 06 Sep 2010 19:52:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 0 [ 41]: OPTIONS sip:s@114.77.232.125:5060 SIP/2.0 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 203.12.160.224:5060;branch=z9hG4bK613ecb3d;rport [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as0a9ea18c [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 3 [ 31]: To: [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 4 [ 38]: Contact: [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 5 [ 56]: Call-ID: 77cb11c325ade06466d4ed6f79feb4f8@203.12.160.224 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 7 [ 24]: User-Agent: Asterisk PBX [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 9 [ 35]: Date: Mon, 06 Sep 2010 19:52:00 GMT [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 10 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:52:01] VERBOSE[10974] chan_sip.c: --- (12 headers 0 lines) --- [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: = Looking for Call ID: 77cb11c325ade06466d4ed6f79feb4f8@203.12.160.224 (Checking From) --From tag as0a9ea18c --To-tag [Sep 7 05:52:01] DEBUG[10974] acl.c: For destination '203.12.160.224', our source address is '192.168.1.1'. [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Target address 203.12.160.224:5060 is not local, substituting externaddr [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 114.77.232.125:5060 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 77cb11c325ade06466d4ed6f79feb4f8@203.12.160.224 - OPTIONS (No RTP) [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Sep 7 05:52:01] VERBOSE[10974] chan_sip.c: Looking for s in incoming (domain 114.77.232.125:5060) [Sep 7 05:52:01] VERBOSE[10974] chan_sip.c: <--- Transmitting (NAT) to 203.12.160.224:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 203.12.160.224:5060;branch=z9hG4bK613ecb3d;received=203.12.160.224;rport=5060 From: "asterisk" ;tag=as0a9ea18c To: ;tag=as4b89ef5d Call-ID: 77cb11c325ade06466d4ed6f79feb4f8@203.12.160.224 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 203.12.160.224:5060;branch=z9hG4bK613ecb3d;received=203.12.160.224;rport=5060 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as0a9ea18c [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 3 [ 46]: To: ;tag=as4b89ef5d [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 4 [ 56]: Call-ID: 77cb11c325ade06466d4ed6f79feb4f8@203.12.160.224 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Header 11 [ 0]: [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 203.12.160.224:5060 [Sep 7 05:52:01] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '77cb11c325ade06466d4ed6f79feb4f8@203.12.160.224' in 32000 ms (Method: OPTIONS) [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: SIP message could not be handled, bad request: 77cb11c325ade06466d4ed6f79feb4f8@203.12.160.224 [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:01] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:02] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:02] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:03] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:03] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:04] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:04] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:05] DEBUG[18272] res_rtp_asterisk.c: Got RTCP report of 44 bytes [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.27:5060 ---> INVITE sip:803@home.smartbyte.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK00b720e55db8df1190e90015afdef833;rport From: ;tag=1895770013 To: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 5 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 259 v=0 o=- 972190714 0 IN IP4 192.168.1.27 s=SIPPER for PhonerLite c=IN IP4 192.168.1.27 t=0 0 m=audio 5062 RTP/AVP 9 8 0 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 0 [ 40]: INVITE sip:803@home.smartbyte.de SIP/2.0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK00b720e55db8df1190e90015afdef833;rport [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 3 [ 31]: To: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 5 [ 14]: CSeq: 5 INVITE [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 6 [ 36]: Contact: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 8 [ 71]: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 10 [ 27]: Supported: 100rel, replaces [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 11 [ 33]: User-Agent: SIPPER for PhonerLite [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 12 [ 44]: P-Preferred-Identity: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 13 [ 19]: Content-Length: 259 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 14 [ 0]: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 1 [ 35]: o=- 972190714 0 IN IP4 192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 2 [ 23]: s=SIPPER for PhonerLite [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 5 [ 30]: m=audio 5062 RTP/AVP 9 8 0 101 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 6 [ 20]: a=rtpmap:9 G722/8000 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 11 [ 10]: a=sendrecv [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: --- (14 headers 12 lines) --- [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: = Looking for Call ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 (Checking From) --From tag 1895770013 --To-tag [Sep 7 05:52:05] DEBUG[10974] acl.c: For destination '192.168.1.27', our source address is '192.168.1.1'. [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.1:5060 [Sep 7 05:52:05] VERBOSE[10974] netsock.c: == Using UDPTL TOS bits 184 [Sep 7 05:52:05] VERBOSE[10974] netsock.c: == Using UDPTL CoS mark 5 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 - INVITE (No RTP) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Sep 7 05:52:05] DEBUG[10974] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel, replaces" [Sep 7 05:52:05] DEBUG[10974] sip/reqresp_parser.c: Found SIP option: -100rel- [Sep 7 05:52:05] DEBUG[10974] sip/reqresp_parser.c: Matched SIP option: 100rel [Sep 7 05:52:05] DEBUG[10974] sip/reqresp_parser.c: Found SIP option: -replaces- [Sep 7 05:52:05] DEBUG[10974] sip/reqresp_parser.c: Matched SIP option: replaces [Sep 7 05:52:05] DEBUG[10974] netsock2.c: Splitting '192.168.1.27:5060' gives... [Sep 7 05:52:05] DEBUG[10974] netsock2.c: ...host '192.168.1.27' and port '5060'. [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Sending to 192.168.1.27:5060 (NAT) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Initializing initreq for method INVITE - callid 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Using INVITE request as basis request - 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found peer '105' for '105' from 192.168.1.27:5060 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.1.27:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK00b720e55db8df1190e90015afdef833;received=192.168.1.27;rport=5060 From: ;tag=1895770013 To: ;tag=as1a4d8d92 Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 5 INVITE Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="home.smartbyte.de", nonce="762dcaa9" Content-Length: 0 <------------> [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 1 [114]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK00b720e55db8df1190e90015afdef833;received=192.168.1.27;rport=5060 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 3 [ 46]: To: ;tag=as1a4d8d92 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 5 [ 14]: CSeq: 5 INVITE [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 9 [ 83]: WWW-Authenticate: Digest algorithm=MD5, realm="home.smartbyte.de", nonce="762dcaa9" [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 11 [ 0]: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16758 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.27:5060 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27' in 32000 ms (Method: INVITE) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.27:5060 ---> ACK sip:803@home.smartbyte.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK00b720e55db8df1190e90015afdef833;rport From: ;tag=1895770013 To: ;tag=as1a4d8d92 Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 5 ACK Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 0 [ 37]: ACK sip:803@home.smartbyte.de SIP/2.0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK00b720e55db8df1190e90015afdef833;rport [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 3 [ 46]: To: ;tag=as1a4d8d92 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 5 [ 11]: CSeq: 5 ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: --- (8 headers 0 lines) --- [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: = Looking for Call ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 (Checking From) --From tag 1895770013 --To-tag as1a4d8d92 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16758 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Stopping retransmission on '00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27' of Response 5: Match Found [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.27:5060 ---> INVITE sip:803@home.smartbyte.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK804db9e55db8df1190e90015afdef833;rport From: ;tag=1895770013 To: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 6 INVITE Contact: Authorization: Digest username="105", realm="home.smartbyte.de", nonce="762dcaa9", uri="sip:803@home.smartbyte.de", response="812dff3e6a9dc595405d66f3cd05a277", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 259 v=0 o=- 972190714 0 IN IP4 192.168.1.27 s=SIPPER for PhonerLite c=IN IP4 192.168.1.27 t=0 0 m=audio 5062 RTP/AVP 9 8 0 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 0 [ 40]: INVITE sip:803@home.smartbyte.de SIP/2.0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK804db9e55db8df1190e90015afdef833;rport [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 3 [ 31]: To: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 5 [ 14]: CSeq: 6 INVITE [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 6 [ 36]: Contact: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 7 [174]: Authorization: Digest username="105", realm="home.smartbyte.de", nonce="762dcaa9", uri="sip:803@home.smartbyte.de", response="812dff3e6a9dc595405d66f3cd05a277", algorithm=MD5 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 9 [ 71]: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 11 [ 27]: Supported: 100rel, replaces [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 12 [ 33]: User-Agent: SIPPER for PhonerLite [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 13 [ 44]: P-Preferred-Identity: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 14 [ 19]: Content-Length: 259 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 15 [ 0]: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 1 [ 35]: o=- 972190714 0 IN IP4 192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 2 [ 23]: s=SIPPER for PhonerLite [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 5 [ 30]: m=audio 5062 RTP/AVP 9 8 0 101 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 6 [ 20]: a=rtpmap:9 G722/8000 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Body 11 [ 10]: a=sendrecv [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: --- (15 headers 12 lines) --- [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: = Looking for Call ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 (Checking From) --From tag 1895770013 --To-tag [Sep 7 05:52:05] DEBUG[10974] netsock2.c: Splitting 'home.smartbyte.de' gives... [Sep 7 05:52:05] DEBUG[10974] netsock2.c: ...host 'home.smartbyte.de' and port '(null)'. [Sep 7 05:52:05] DEBUG[10974] netsock2.c: Splitting 'home.smartbyte.de' gives... [Sep 7 05:52:05] DEBUG[10974] netsock2.c: ...host 'home.smartbyte.de' and port '(null)'. [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Sep 7 05:52:05] DEBUG[10974] netsock2.c: Splitting '192.168.1.27:5060' gives... [Sep 7 05:52:05] DEBUG[10974] netsock2.c: ...host '192.168.1.27' and port '5060'. [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Sending to 192.168.1.27:5060 (NAT) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Initializing initreq for method INVITE - callid 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Using INVITE request as basis request - 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found peer '105' for '105' from 192.168.1.27:5060 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x429748d0' [Sep 7 05:52:05] DEBUG[10974] res_rtp_asterisk.c: Allocated port 10976 for RTP instance '0x429748d0' [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: RTP instance '0x429748d0' is setup and ready to go [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x429d2c58' [Sep 7 05:52:05] DEBUG[10974] res_rtp_asterisk.c: Allocated port 10328 for RTP instance '0x429d2c58' [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: RTP instance '0x429d2c58' is setup and ready to go [Sep 7 05:52:05] DEBUG[10974] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x429d2c58' [Sep 7 05:52:05] DEBUG[10974] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x429748d0' [Sep 7 05:52:05] VERBOSE[10974] netsock2.c: == Using SIP RTP TOS bits 184 [Sep 7 05:52:05] VERBOSE[10974] netsock2.c: == Using SIP RTP CoS mark 5 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Setting NAT on RTP to On [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Setting NAT on VRTP to On [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Setting NAT on UDPTL to On [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing session-level SDP o=- 972190714 0 IN IP4 192.168.1.27... UNSUPPORTED. [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing session-level SDP s=SIPPER for PhonerLite... UNSUPPORTED. [Sep 7 05:52:05] DEBUG[10974] netsock2.c: Splitting '192.168.1.27' gives... [Sep 7 05:52:05] DEBUG[10974] netsock2.c: ...host '192.168.1.27' and port '(null)'. [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.27... OK. [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found RTP audio format 9 [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Setting payload 9 based on m type on 0x41a31030 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found RTP audio format 8 [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Setting payload 8 based on m type on 0x41a31030 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found RTP audio format 0 [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Setting payload 0 based on m type on 0x41a31030 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found RTP audio format 101 [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Setting payload 101 based on m type on 0x41a31030 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found audio description format G722 for ID 9 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found audio description format PCMA for ID 8 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found audio description format PCMU for ID 0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Found audio description format telephone-event for ID 101 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Incorporating payload 0 on 0x41a31030 [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Incorporating payload 8 on 0x41a31030 [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Incorporating payload 9 on 0x41a31030 [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Incorporating payload 101 on 0x41a31030 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x100c (ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100c (ulaw|alaw|g722) [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 7 05:52:05] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x429748d0' [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Peer audio RTP is at port 192.168.1.27:5062 [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Copying payload 0 from 0x41a31030 to 0x42974a7c [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Copying payload 8 from 0x41a31030 to 0x42974a7c [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Copying payload 9 from 0x41a31030 to 0x42974a7c [Sep 7 05:52:05] DEBUG[10974] rtp_engine.c: Copying payload 101 from 0x41a31030 to 0x42974a7c [Sep 7 05:52:05] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x429d2c58' [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Peer doesn't provide video [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Peer doesn't provide T.38 UDPTL [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: We're settling with these formats: 0x100c (ulaw|alaw|g722) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Checking SIP call limits for device 105 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Updating call counter for incoming call [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: Looking for 803 in default (domain home.smartbyte.de) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: *** Our native formats are 0x8 (alaw) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: *** Joint capabilities are 0x100c (ulaw|alaw|g722) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: *** Our capabilities are 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Sep 7 05:52:05] DEBUG[10974] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Sep 7 05:52:05] DEBUG[10974] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: build_route: Contact hop: [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: list_route: hop: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: SIP/105-0000001d: New call is still down.... Trying... [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK804db9e55db8df1190e90015afdef833;received=192.168.1.27;rport=5060 From: ;tag=1895770013 To: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 6 INVITE Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 1 [114]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK804db9e55db8df1190e90015afdef833;received=192.168.1.27;rport=5060 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 3 [ 31]: To: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 5 [ 14]: CSeq: 6 INVITE [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 9 [ 35]: Contact: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 11 [ 0]: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.27:5060 [Sep 7 05:52:05] DEBUG[10957] devicestate.c: No provider found, checking channel drivers for SIP - 105 [Sep 7 05:52:05] DEBUG[10957] chan_sip.c: Checking device state for peer 105 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:05] DEBUG[10957] devicestate.c: Changing state for SIP/105 - state 1 (Not in use) [Sep 7 05:52:05] DEBUG[10957] devicestate.c: device 'SIP/105' state '1' [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:05] DEBUG[18297] pbx.c: Launching 'ChanSpy' [Sep 7 05:52:05] DEBUG[11003] app_queue.c: Device 'SIP/105' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 7 05:52:05] VERBOSE[18297] pbx.c: -- Executing [803@default:1] ChanSpy("SIP/105-0000001d", "SIP/100,q") in new stack [Sep 7 05:52:05] DEBUG[18297] channel.c: Set channel SIP/105-0000001d to write format slin [Sep 7 05:52:05] DEBUG[10957] devicestate.c: No provider found, checking channel drivers for SIP - 105 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: SIP answering channel: SIP/105-0000001d [Sep 7 05:52:05] DEBUG[10957] chan_sip.c: Checking device state for peer 105 [Sep 7 05:52:05] DEBUG[18297] res_rtp_asterisk.c: Setting the marker bit due to a source update [Sep 7 05:52:05] DEBUG[10957] devicestate.c: Changing state for SIP/105 - state 1 (Not in use) [Sep 7 05:52:05] DEBUG[10957] devicestate.c: device 'SIP/105' state '1' [Sep 7 05:52:05] DEBUG[11003] app_queue.c: Device 'SIP/105' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Setting framing from config on incoming call [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: ** Our capability: 0x100c (ulaw|alaw|g722) Video flag: True Text flag: True [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Sep 7 05:52:05] VERBOSE[18297] chan_sip.c: Audio is at 5060 [Sep 7 05:52:05] VERBOSE[18297] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Sep 7 05:52:05] VERBOSE[18297] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Sep 7 05:52:05] VERBOSE[18297] chan_sip.c: Adding codec 0x1000 (g722) to SDP [Sep 7 05:52:05] VERBOSE[18297] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: -- Done with adding codecs to SDP [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Done building SDP. Settling with this capability: 0x100c (ulaw|alaw|g722) [Sep 7 05:52:05] VERBOSE[18297] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.1.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK804db9e55db8df1190e90015afdef833;received=192.168.1.27;rport=5060 From: ;tag=1895770013 To: ;tag=as78829a27 Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 6 INVITE Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 292 v=0 o=root 1439324534 1439324534 IN IP4 192.168.1.1 s=Asterisk PBX SVN-trunk-r285058 c=IN IP4 192.168.1.1 t=0 0 m=audio 10976 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 1 [114]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK804db9e55db8df1190e90015afdef833;received=192.168.1.27;rport=5060 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 3 [ 46]: To: ;tag=as78829a27 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 5 [ 14]: CSeq: 6 INVITE [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 9 [ 35]: Contact: [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 11 [ 19]: Content-Length: 292 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Header 12 [ 0]: [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 0 [ 3]: v=0 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 1 [ 47]: o=root 1439324534 1439324534 IN IP4 192.168.1.1 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 2 [ 32]: s=Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 3 [ 20]: c=IN IP4 192.168.1.1 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 4 [ 5]: t=0 0 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 5 [ 31]: m=audio 10976 RTP/AVP 8 0 9 101 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 8 [ 20]: a=rtpmap:9 G722/8000 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Body 12 [ 10]: a=sendrecv [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16761 [Sep 7 05:52:05] DEBUG[18297] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.27:5060 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.27:5060 ---> ACK sip:803@192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK804db9e55db8df1190ea0015afdef833;rport From: ;tag=1895770013 To: ;tag=as78829a27 Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 6 ACK Contact: Authorization: Digest username="105", realm="home.smartbyte.de", nonce="762dcaa9", uri="sip:803@192.168.1.1:5060", response="ff634d4b35faf62b4ad83ed70afddde1", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 0 [ 36]: ACK sip:803@192.168.1.1:5060 SIP/2.0 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK804db9e55db8df1190ea0015afdef833;rport [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 3 [ 46]: To: ;tag=as78829a27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 5 [ 11]: CSeq: 6 ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 6 [ 36]: Contact: [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 7 [173]: Authorization: Digest username="105", realm="home.smartbyte.de", nonce="762dcaa9", uri="sip:803@192.168.1.1:5060", response="ff634d4b35faf62b4ad83ed70afddde1", algorithm=MD5 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:52:05] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: = Looking for Call ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 (Checking From) --From tag 1895770013 --To-tag as78829a27 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16761 [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Stopping retransmission on '00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27' of Response 6: Match Found [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:05] DEBUG[10974] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:05] DEBUG[18297] res_rtp_asterisk.c: Got RTCP report of 42 bytes [Sep 7 05:52:05] DEBUG[18297] res_rtp_asterisk.c: Unknown RTCP packet (pt=0) received from 192.168.1.27:5063 [Sep 7 05:52:05] DEBUG[18297] res_rtp_asterisk.c: RTCP Read too short [Sep 7 05:52:05] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:05] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:05] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:05] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] autochan.c: Created autochan 0x42c47b10 to hold channel SIP/100-0000001b (0xe670a08) [Sep 7 05:52:06] VERBOSE[18297] app_chanspy.c: == Spying on channel SIP/100-0000001b [Sep 7 05:52:06] NOTICE[18297] app_chanspy.c: Attaching SIP/105-0000001d to SIP/100-0000001b [Sep 7 05:52:06] DEBUG[18297] channel.c: Soft-Hanging up channel 'SIP/smartbyte.de-0000001c' [Sep 7 05:52:06] NOTICE[18297] app_chanspy.c: Attaching SIP/105-0000001d to SIP/100-0000001b [Sep 7 05:52:06] DEBUG[18297] channel.c: Soft-Hanging up channel 'SIP/smartbyte.de-0000001c' [Sep 7 05:52:06] DEBUG[18297] autochan.c: Created autochan 0x42c560e8 to hold channel SIP/smartbyte.de-0000001c (0xe5fc410) [Sep 7 05:52:06] NOTICE[18297] app_chanspy.c: Attaching SIP/105-0000001d to SIP/smartbyte.de-0000001c [Sep 7 05:52:06] DEBUG[18297] channel.c: Soft-Hanging up channel 'SIP/100-0000001b' [Sep 7 05:52:06] DEBUG[18272] rtp_engine.c: rtp-engine-local-bridge: Ooh, empty read... [Sep 7 05:52:06] DEBUG[18297] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Sep 7 05:52:06] VERBOSE[18272] channel.c: -- Native bridging SIP/100-0000001b and SIP/smartbyte.de-0000001c ended [Sep 7 05:52:06] DEBUG[18272] channel.c: Didn't get a frame from channel: SIP/smartbyte.de-0000001c [Sep 7 05:52:06] DEBUG[18272] channel.c: Bridge stops bridging channels SIP/100-0000001b and SIP/smartbyte.de-0000001c [Sep 7 05:52:06] DEBUG[18272] pbx.c: Result of 'DIALSTATUS' is 'ANSWER' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Launching 'Goto' [Sep 7 05:52:06] VERBOSE[18272] pbx.c: -- Executing [h@default:1] Goto("SIP/100-0000001b", "dialstatus,ANSWER,1") in new stack [Sep 7 05:52:06] VERBOSE[18272] pbx.c: -- Goto (dialstatus,ANSWER,1) [Sep 7 05:52:06] DEBUG[18272] pbx.c: Launching 'Hangup' [Sep 7 05:52:06] VERBOSE[18272] pbx.c: -- Executing [ANSWER@dialstatus:1] Hangup("SIP/100-0000001b", "") in new stack [Sep 7 05:52:06] DEBUG[18272] cdr_mysql.c: Inserting a CDR record. [Sep 7 05:52:06] DEBUG[18272] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid,userfield) VALUES ('2010-09-07 05:51:53','\"Kati Krebs\" <100>','100','19004018881000','default','SIP/100-0000001b','SIP/smartbyte.de-0000001c','Dial','SIP/04018881000@smartbyte.de','13','11','ANSWERED','3','1283802713.39','04018881000') [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is '2010-09-07 05:51:53' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is '"Kati Krebs" <100>' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is 'default' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is 'SIP/100-0000001b' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is 'SIP/smartbyte.de-0000001c' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is 'Dial' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is 'SIP/04018881000@smartbyte.de' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is '13' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is '11' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is 'ANSWERED' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is 'DOCUMENTATION' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is '(null)' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is '1283802713.39' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is '04018881000' [Sep 7 05:52:06] DEBUG[18272] pbx.c: Function result is '(null)' [Sep 7 05:52:06] DEBUG[18272] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2010-09-07 05:51:53','"Kati Krebs" <100>','default','SIP/100-0000001b','SIP/smartbyte.de-0000001c','Dial','SIP/04018881000@smartbyte.de','13','11','ANSWERED','DOCUMENTATION','','1283802713.39','04018881000','') [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18272] channel.c: Hanging up channel 'SIP/smartbyte.de-0000001c' [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Hangup call SIP/smartbyte.de-0000001c, SIP callid 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:52:06] DEBUG[18272] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xe65e400' [Sep 7 05:52:06] DEBUG[18272] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xe66afd8' [Sep 7 05:52:06] VERBOSE[18272] chan_sip.c: Scheduling destruction of SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' in 32000 ms (Method: INVITE) [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Session timer stopped: -1 - 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Strict routing enforced for session 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:52:06] VERBOSE[18272] chan_sip.c: set_destination: Parsing for address/port to send to [Sep 7 05:52:06] DEBUG[18272] netsock2.c: Splitting '80.237.160.128:5060' gives... [Sep 7 05:52:06] DEBUG[18272] netsock2.c: ...host '80.237.160.128' and port '5060'. [Sep 7 05:52:06] VERBOSE[18272] chan_sip.c: set_destination: set destination to 80.237.160.128:5060 [Sep 7 05:52:06] VERBOSE[18272] chan_sip.c: Reliably Transmitting (NAT) to 80.237.160.128:5060: BYE sip:04018881000@80.237.160.128:5060 SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK27e2ac30;rport Max-Forwards: 70 From: "Kati Krebs" ;tag=as6cafe189 To: ;tag=as299e4ad4 Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-trunk-r285058 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 0 [ 47]: BYE sip:04018881000@80.237.160.128:5060 SIP/2.0 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK27e2ac30;rport [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 3 [ 66]: From: "Kati Krebs" ;tag=as6cafe189 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 4 [ 51]: To: ;tag=as299e4ad4 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 5 [ 61]: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 6 [ 13]: CSeq: 103 BYE [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 11 [ 0]: [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16764 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Trying to put 'BYE sip:040' onto UDP socket destined for 80.237.160.128:5060 [Sep 7 05:52:06] DEBUG[18272] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Sep 7 05:52:06] DEBUG[18272] pbx.c: Spawn extension (dialstatus,19004018881000,3) exited non-zero on 'SIP/100-0000001b' [Sep 7 05:52:06] VERBOSE[18272] pbx.c: == Spawn extension (dialstatus, 19004018881000, 3) exited non-zero on 'SIP/100-0000001b' [Sep 7 05:52:06] DEBUG[18272] channel.c: Soft-Hanging up channel 'SIP/100-0000001b' [Sep 7 05:52:06] DEBUG[18272] channel.c: Hanging up channel 'SIP/100-0000001b' [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Hangup call SIP/100-0000001b, SIP callid 3011425442@192_168_1_150 [Sep 7 05:52:06] DEBUG[18272] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x429f9170' [Sep 7 05:52:06] VERBOSE[18272] chan_sip.c: Scheduling destruction of SIP dialog '3011425442@192_168_1_150' in 32000 ms (Method: ACK) [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Strict routing enforced for session 3011425442@192_168_1_150 [Sep 7 05:52:06] VERBOSE[18272] chan_sip.c: set_destination: Parsing for address/port to send to [Sep 7 05:52:06] DEBUG[18272] netsock2.c: Splitting '192.168.1.150:5064' gives... [Sep 7 05:52:06] DEBUG[18272] netsock2.c: ...host '192.168.1.150' and port '5064'. [Sep 7 05:52:06] VERBOSE[18272] chan_sip.c: set_destination: set destination to 192.168.1.150:5064 [Sep 7 05:52:06] VERBOSE[18272] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.150:5064: BYE sip:100@192.168.1.150:5064 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK0e8dad39;rport Max-Forwards: 70 From: ;tag=as14f3f609 To: "Kati Krebs" ;tag=843715617 Call-ID: 3011425442@192_168_1_150 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r285058 Proxy-Authorization: Digest username="100", realm="home.smartbyte.de", algorithm=MD5, uri="home.smartbyte.de", nonce="", response="e4373a6a098ebdc923da3694be62fc76" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 0 [ 38]: BYE sip:100@192.168.1.150:5064 SIP/2.0 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK0e8dad39;rport [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 3 [ 70]: From: ;tag=as14f3f609 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 4 [ 58]: To: "Kati Krebs" ;tag=843715617 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 5 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 8 [164]: Proxy-Authorization: Digest username="100", realm="home.smartbyte.de", algorithm=MD5, uri="home.smartbyte.de", nonce="", response="e4373a6a098ebdc923da3694be62fc76" [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Header 12 [ 0]: [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16766 [Sep 7 05:52:06] DEBUG[18272] chan_sip.c: Trying to put 'BYE sip:100' onto UDP socket destined for 192.168.1.150:5064 [Sep 7 05:52:06] DEBUG[18297] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Sep 7 05:52:06] DEBUG[18297] autochan.c: Removed autochan 0x42c560e8 from the list, about to free it [Sep 7 05:52:06] VERBOSE[18297] app_chanspy.c: == Done Spying on channel SIP/100-0000001b [Sep 7 05:52:06] DEBUG[10957] devicestate.c: No provider found, checking channel drivers for SIP - smartbyte.de [Sep 7 05:52:06] DEBUG[18297] autochan.c: Removed autochan 0x42c47b10 from the list, about to free it [Sep 7 05:52:06] DEBUG[10957] chan_sip.c: Checking device state for peer smartbyte.de [Sep 7 05:52:06] DEBUG[10957] devicestate.c: Changing state for SIP/smartbyte.de - state 1 (Not in use) [Sep 7 05:52:06] DEBUG[10957] devicestate.c: device 'SIP/smartbyte.de' state '1' [Sep 7 05:52:06] DEBUG[11003] app_queue.c: Device 'SIP/smartbyte.de' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.24E+02, Et=1.64E+06, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=8.55E+04, Et=1.64E+06, s/n= 0.06 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=9.05E+03, Et=2.77E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=5.38E+03, Et=3.56E+07, s/n= 0.00 [Sep 7 05:52:06] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.150:5064 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK0e8dad39;rport=5060 From: ;tag=as14f3f609 To: "Kati Krebs" ;tag=843715617 Call-ID: 3011425442@192_168_1_150 CSeq: 102 BYE Contact: Supported: replaces Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK0e8dad39;rport=5060 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 2 [ 70]: From: ;tag=as14f3f609 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 3 [ 58]: To: "Kati Krebs" ;tag=843715617 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 4 [ 33]: Call-ID: 3011425442@192_168_1_150 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 6 [ 37]: Contact: [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 7 [ 19]: Supported: replaces [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:52:06] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3011425442@192_168_1_150 (Checking To) --From tag as14f3f609 --To-tag 843715617 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16766 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Stopping retransmission on '3011425442@192_168_1_150' of Request 102: Match Found [Sep 7 05:52:06] VERBOSE[10974] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Destroying SIP dialog 3011425442@192_168_1_150 [Sep 7 05:52:06] VERBOSE[10974] chan_sip.c: Really destroying SIP dialog '3011425442@192_168_1_150' Method: ACK [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: ---------- SIP HISTORY for '3011425442@192_168_1_150' [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: * SIP Call [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 001. Rx INVITE / 2 INVITE / sip:19004018881000@home.smartbyte.de;user=p [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 002. AuthChal Auth challenge sent for - nc 0 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 003. TxRespRel SIP/2.0 / 2 INVITE - 401 Unauthorized [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 004. SchedDestroy 32000 ms [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 005. Rx ACK / 2 ACK / sip:19004018881000@home.smartbyte.de;user=phone [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 006. Rx INVITE / 3 INVITE / sip:19004018881000@home.smartbyte.de;user=p [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 007. CancelDestroy [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 008. Invite New call: 3011425442@192_168_1_150 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 009. AuthOK Auth challenge successful for 100 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 010. NewChan Channel SIP/100-0000001b - from 3011425442@192_168_1_150 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 011. TxResp SIP/2.0 / 3 INVITE - 100 Trying [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 012. TxResp SIP/2.0 / 3 INVITE - 183 Session Progress [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 013. TxRespRel SIP/2.0 / 3 INVITE - 200 OK [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 014. Rx ACK / 3 ACK / sip:19004018881000@192.168.1.1:5060 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 015. Hangup Cause Normal Clearing [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 016. SchedDestroy 32000 ms [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 017. RTCPaudio Quality:ssrc=1873087836;themssrc=0;lp=1;rxjitter=0.000000;rxcou [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 018. AuthResp Auth response sent for 100 in realm home.smartbyte.de - nc 1 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 019. TxReqRel BYE / 102 BYE - BYE [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 020. Rx SIP/2.0 / 102 BYE / 200 OK [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 021. NeedDestroy Setting needdestroy because transaction completed [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: ---------- END SIP HISTORY for '3011425442@192_168_1_150' [Sep 7 05:52:06] DEBUG[10974] rtp_engine.c: Destroyed RTP instance '0x429f9170' [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=8.73E+04, Et=3.92E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=6.62E+04, Et=3.83E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=7.38E+03, Et=3.56E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=2.61E+04, Et=3.95E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=3.19E+04, Et=4.36E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=1.05E+05, Et=4.41E+07, s/n= 0.00 [Sep 7 05:52:06] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK27e2ac30;received=114.77.232.125;rport=5060 From: "Kati Krebs" ;tag=as6cafe189 To: ;tag=as299e4ad4 Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 CSeq: 103 BYE Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK27e2ac30;received=114.77.232.125;rport=5060 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 2 [ 66]: From: "Kati Krebs" ;tag=as6cafe189 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 3 [ 51]: To: ;tag=as299e4ad4 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 4 [ 61]: Call-ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:52:06] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: = Looking for Call ID: 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 (Checking To) --From tag as6cafe189 --To-tag as299e4ad4 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16764 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Stopping retransmission on '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' of Request 103: Match Found [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: Destroying SIP dialog 341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060 [Sep 7 05:52:06] VERBOSE[10974] chan_sip.c: Really destroying SIP dialog '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' Method: INVITE [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: ---------- SIP HISTORY for '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: * SIP Call [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 001. NewChan Channel SIP/smartbyte.de-0000001c - from 341764e45b50a0ab3e8019 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 100 Trying [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 183 Session Progress [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 005. Rx SIP/2.0 / 102 INVITE / 200 OK [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 006. TxReq ACK / 102 ACK - ACK [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 007. Hangup Cause Normal Clearing [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 008. SchedDestroy 32000 ms [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 009. RTCPaudio Quality:ssrc=2109686368;themssrc=173267043;lp=0;rxjitter=0.0001 [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 010. RTCPvideo Quality:ssrc=2006806891;themssrc=0;lp=0;rxjitter=0.000000;rxcou [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 011. TxReqRel BYE / 103 BYE - BYE [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 012. Rx SIP/2.0 / 103 BYE / 200 OK [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: 013. NeedDestroy Setting needdestroy because received 200 response [Sep 7 05:52:06] DEBUG[10974] chan_sip.c: ---------- END SIP HISTORY for '341764e45b50a0ab3e8019f308b7a5a5@114.77.232.125:5060' [Sep 7 05:52:06] DEBUG[10974] rtp_engine.c: Destroyed RTP instance '0xe65e400' [Sep 7 05:52:06] DEBUG[10974] rtp_engine.c: Destroyed RTP instance '0xe66afd8' [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=2.09E+05, Et=5.72E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=1.10E+05, Et=5.76E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=8.47E+04, Et=7.59E+07, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=1.31E+05, Et=1.13E+08, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=2.21E+05, Et=1.23E+08, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=2.53E+06, Et=1.58E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=2.01E+06, Et=2.25E+08, s/n= 0.01 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=2.19E+05, Et=3.68E+08, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=1.81E+06, Et=3.19E+08, s/n= 0.01 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=1.61E+06, Et=3.32E+08, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=8.90E+06, Et=2.59E+08, s/n= 0.04 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=1.36E+07, Et=2.76E+08, s/n= 0.05 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.44E+06, Et=2.86E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=9.99E+06, Et=3.01E+08, s/n= 0.03 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=1.83E+07, Et=3.06E+08, s/n= 0.06 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=5.33E+06, Et=2.76E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=6.47E+06, Et=3.25E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=6.75E+06, Et=3.22E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=8.25E+05, Et=3.31E+08, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=5.81E+06, Et=2.82E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=9.03E+06, Et=2.51E+08, s/n= 0.04 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=8.14E+06, Et=3.48E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=1.56E+06, Et=2.55E+08, s/n= 0.01 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=9.22E+05, Et=2.94E+08, s/n= 0.00 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=4.67E+06, Et=2.38E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=5.63E+06, Et=2.34E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=3.31E+06, Et=2.10E+08, s/n= 0.02 [Sep 7 05:52:06] NOTICE[10974] chan_sip.c: -- Re-registration for 101@80.237.160.128 [Sep 7 05:52:06] VERBOSE[10974] dnsmgr.c: > doing dnsmgr_lookup for '80.237.160.128' [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=3.10E+06, Et=2.09E+08, s/n= 0.02 [Sep 7 05:52:06] DEBUG[18297] dsp.c: tone 1100, Ew=5.36E+06, Et=2.48E+08, s/n= 0.02 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=4.67E+05, Et=4.05E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.55E+05, Et=1.83E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=8.75E+05, Et=2.34E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=3.63E+05, Et=1.69E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=4.35E+06, Et=2.27E+08, s/n= 0.02 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.35E+06, Et=2.58E+08, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=4.71E+06, Et=2.00E+08, s/n= 0.02 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.14E+06, Et=1.96E+08, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=6.45E+06, Et=2.04E+08, s/n= 0.03 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=4.23E+06, Et=1.93E+08, s/n= 0.02 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.77E+05, Et=1.55E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.91E+06, Et=1.23E+08, s/n= 0.02 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=5.93E+05, Et=1.57E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=4.04E+04, Et=1.24E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.48E+06, Et=1.76E+08, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.84E+05, Et=1.13E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[10974] netsock2.c: Splitting '80.237.160.128' gives... [Sep 7 05:52:07] DEBUG[10974] netsock2.c: ...host '80.237.160.128' and port '(null)'. [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 3a0375611ec2831f427a14264de70562@192.168.0.1 - REGISTER (No RTP) [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: SIP Registry 80.237.160.128: refcount now 3 [Sep 7 05:52:07] DEBUG[10974] acl.c: For destination '80.237.160.128', our source address is '192.168.1.1'. [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Target address 80.237.160.128:5060 is not local, substituting externaddr [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 114.77.232.125:5060 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: SIP Registry 80.237.160.128: refcount now 4 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Scheduled a registration timeout for 80.237.160.128 id #16767 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: >>> Re-using Auth data for 101@80.237.160.128 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Initializing initreq for method REGISTER - callid 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 0 [ 35]: REGISTER sip:80.237.160.128 SIP/2.0 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2f7a64d3;rport [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 3 [ 45]: From: ;tag=as7436e77d [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 4 [ 28]: To: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 6 [ 19]: CSeq: 1002 REGISTER [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 8 [162]: Authorization: Digest username="101", realm="smartbyte.de", algorithm=MD5, uri="sip:80.237.160.128", nonce="0a7f0fb0", response="dc9a17fbdde8d709a0372938261d36b9" [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 10 [ 38]: Contact: [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: REGISTER 11 headers, 0 lines [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: REGISTER attempt 1 to 101@80.237.160.128 [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: Reliably Transmitting (NAT) to 80.237.160.128:5060: REGISTER sip:80.237.160.128 SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2f7a64d3;rport Max-Forwards: 70 From: ;tag=as7436e77d To: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1002 REGISTER User-Agent: Asterisk PBX SVN-trunk-r285058 Authorization: Digest username="101", realm="smartbyte.de", algorithm=MD5, uri="sip:80.237.160.128", nonce="0a7f0fb0", response="dc9a17fbdde8d709a0372938261d36b9" Expires: 120 Contact: Content-Length: 0 --- [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 0 [ 35]: REGISTER sip:80.237.160.128 SIP/2.0 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2f7a64d3;rport [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 3 [ 45]: From: ;tag=as7436e77d [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 4 [ 28]: To: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 6 [ 19]: CSeq: 1002 REGISTER [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 8 [162]: Authorization: Digest username="101", realm="smartbyte.de", algorithm=MD5, uri="sip:80.237.160.128", nonce="0a7f0fb0", response="dc9a17fbdde8d709a0372938261d36b9" [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 10 [ 38]: Contact: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 12 [ 0]: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16768 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 80.237.160.128:5060 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: SIP Registry 80.237.160.128: refcount now 3 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=4.71E+05, Et=2.26E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=5.33E+05, Et=1.49E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=7.49E+05, Et=8.45E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=3.71E+05, Et=1.16E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=4.01E+05, Et=1.22E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.84E+05, Et=1.20E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.11E+05, Et=6.68E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.66E+06, Et=4.89E+07, s/n= 0.04 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=6.07E+03, Et=6.57E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.61E+05, Et=1.05E+08, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.01E+05, Et=9.12E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.64E+05, Et=5.05E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=9.52E+03, Et=6.01E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.15E+05, Et=5.96E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.35E+05, Et=4.94E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.64E+05, Et=3.10E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=3.28E+05, Et=3.63E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.91E+05, Et=4.26E+07, s/n= 0.01 [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2f7a64d3;received=114.77.232.125;rport=5060 From: ;tag=as7436e77d To: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1002 REGISTER Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2f7a64d3;received=114.77.232.125;rport=5060 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 2 [ 45]: From: ;tag=as7436e77d [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 3 [ 28]: To: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1002 REGISTER [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 (Checking To) --From tag as7436e77d --To-tag [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2f7a64d3;received=114.77.232.125;rport=5060 From: ;tag=as7436e77d To: ;tag=as5e263bdf Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1002 REGISTER Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="smartbyte.de", nonce="7d6e7843" Content-Length: 0 <-------------> [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK2f7a64d3;received=114.77.232.125;rport=5060 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 2 [ 45]: From: ;tag=as7436e77d [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 3 [ 43]: To: ;tag=as5e263bdf [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1002 REGISTER [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 9 [ 78]: WWW-Authenticate: Digest algorithm=MD5, realm="smartbyte.de", nonce="7d6e7843" [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: --- (11 headers 0 lines) --- [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 (Checking To) --From tag as7436e77d --To-tag as5e263bdf [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16768 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Stopping retransmission on '3a0375611ec2831f427a14264de70562@192.168.0.1' of Request 1002: Match Found [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: Responding to challenge, registration to domain/host name 80.237.160.128 [Sep 7 05:52:07] VERBOSE[10974] dnsmgr.c: > doing dnsmgr_lookup for '80.237.160.128' [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=6.92E+05, Et=6.82E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[10974] netsock2.c: Splitting '80.237.160.128' gives... [Sep 7 05:52:07] DEBUG[10974] netsock2.c: ...host '80.237.160.128' and port '(null)'. [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Initializing already initialized SIP dialog 3a0375611ec2831f427a14264de70562@192.168.0.1 (presumably reinvite) [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 0 [ 35]: REGISTER sip:80.237.160.128 SIP/2.0 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;rport [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 3 [ 45]: From: ;tag=as4cc030a8 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 4 [ 28]: To: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 6 [ 19]: CSeq: 1003 REGISTER [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 8 [162]: Authorization: Digest username="101", realm="smartbyte.de", algorithm=MD5, uri="sip:80.237.160.128", nonce="7d6e7843", response="f7dc4d41295b94b9209607a8ab2e45bb" [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 10 [ 38]: Contact: [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: REGISTER 11 headers, 0 lines [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: REGISTER attempt 2 to 101@80.237.160.128 [Sep 7 05:52:07] VERBOSE[10974] chan_sip.c: Reliably Transmitting (NAT) to 80.237.160.128:5060: REGISTER sip:80.237.160.128 SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;rport Max-Forwards: 70 From: ;tag=as4cc030a8 To: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1003 REGISTER User-Agent: Asterisk PBX SVN-trunk-r285058 Authorization: Digest username="101", realm="smartbyte.de", algorithm=MD5, uri="sip:80.237.160.128", nonce="7d6e7843", response="f7dc4d41295b94b9209607a8ab2e45bb" Expires: 120 Contact: Content-Length: 0 --- [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 0 [ 35]: REGISTER sip:80.237.160.128 SIP/2.0 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;rport [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 3 [ 45]: From: ;tag=as4cc030a8 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 4 [ 28]: To: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 5 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 6 [ 19]: CSeq: 1003 REGISTER [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 7 [ 42]: User-Agent: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 8 [162]: Authorization: Digest username="101", realm="smartbyte.de", algorithm=MD5, uri="sip:80.237.160.128", nonce="7d6e7843", response="f7dc4d41295b94b9209607a8ab2e45bb" [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 10 [ 38]: Contact: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Header 12 [ 0]: [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16769 [Sep 7 05:52:07] DEBUG[10974] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 80.237.160.128:5060 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.37E+06, Et=5.68E+07, s/n= 0.02 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=5.10E+06, Et=5.76E+07, s/n= 0.10 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=3.17E+05, Et=3.84E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=6.29E+05, Et=5.85E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=3.60E+05, Et=7.48E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.03E+05, Et=6.41E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=2.88E+05, Et=4.92E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=4.86E+05, Et=3.41E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=3.88E+05, Et=4.83E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=5.58E+05, Et=6.27E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=3.74E+05, Et=6.03E+07, s/n= 0.01 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.35E+05, Et=8.63E+07, s/n= 0.00 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.80E+06, Et=5.99E+07, s/n= 0.03 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.16E+06, Et=4.62E+07, s/n= 0.03 [Sep 7 05:52:07] DEBUG[18297] dsp.c: tone 1100, Ew=1.32E+06, Et=4.02E+07, s/n= 0.03 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=6.33E+05, Et=7.24E+07, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=3.03E+05, Et=6.49E+07, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.14E+06, Et=7.86E+07, s/n= 0.01 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;received=114.77.232.125;rport=5060 From: ;tag=as4cc030a8 To: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1003 REGISTER Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;received=114.77.232.125;rport=5060 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 2 [ 45]: From: ;tag=as4cc030a8 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 3 [ 28]: To: [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1003 REGISTER [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 (Checking To) --From tag as4cc030a8 --To-tag [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=3.29E+05, Et=8.28E+07, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=6.67E+04, Et=6.86E+07, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.20E+05, Et=7.36E+07, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.12E+05, Et=7.87E+07, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=5.75E+05, Et=9.72E+07, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=9.18E+05, Et=8.59E+07, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=7.09E+04, Et=6.25E+07, s/n= 0.00 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: SIP TIMER: Rescheduling retransmission #16769 (1) REGISTER - 2 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #16769)) [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: Retransmitting #1 (NAT) to 80.237.160.128:5060: REGISTER sip:80.237.160.128 SIP/2.0 Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;rport Max-Forwards: 70 From: ;tag=as4cc030a8 To: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1003 REGISTER User-Agent: Asterisk PBX SVN-trunk-r285058 Authorization: Digest username="101", realm="smartbyte.de", algorithm=MD5, uri="sip:80.237.160.128", nonce="7d6e7843", response="f7dc4d41295b94b9209607a8ab2e45bb" Expires: 120 Contact: Content-Length: 0 --- [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 80.237.160.128:5060 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=5.80E+06, Et=6.48E+07, s/n= 0.10 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=3.16E+06, Et=3.73E+08, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=6.55E+07, Et=4.07E+09, s/n= 0.02 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.60E+07, Et=1.04E+09, s/n= 0.03 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.59E+06, Et=9.34E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.00E+06, Et=1.02E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.67E+06, Et=6.52E+08, s/n= 0.00 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> OPTIONS sip:100@114.77.232.125:5060 SIP/2.0 Via: SIP/2.0/UDP 80.237.160.128:5060;branch=z9hG4bK297bc4c0 Max-Forwards: 70 From: "Asterisk" ;tag=as64c30ca3 To: Contact: Call-ID: 51d888151d1ae1b932b81bd83346ac29@80.237.160.128:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN--r Date: Mon, 06 Sep 2010 19:52:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 0 [ 43]: OPTIONS sip:100@114.77.232.125:5060 SIP/2.0 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 80.237.160.128:5060;branch=z9hG4bK297bc4c0 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 3 [ 61]: From: "Asterisk" ;tag=as64c30ca3 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 4 [ 33]: To: [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 5 [ 43]: Contact: [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 6 [ 61]: Call-ID: 51d888151d1ae1b932b81bd83346ac29@80.237.160.128:5060 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX SVN--r [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 9 [ 35]: Date: Mon, 06 Sep 2010 19:52:08 GMT [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: --- (13 headers 0 lines) --- [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: = Looking for Call ID: 51d888151d1ae1b932b81bd83346ac29@80.237.160.128:5060 (Checking From) --From tag as64c30ca3 --To-tag [Sep 7 05:52:08] DEBUG[10974] acl.c: For destination '80.237.160.128', our source address is '192.168.1.1'. [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Target address 80.237.160.128:5060 is not local, substituting externaddr [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 114.77.232.125:5060 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Allocating new SIP dialog for 51d888151d1ae1b932b81bd83346ac29@80.237.160.128:5060 - OPTIONS (No RTP) [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: Looking for 100 in incoming (domain 114.77.232.125:5060) [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: <--- Transmitting (NAT) to 80.237.160.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 80.237.160.128:5060;branch=z9hG4bK297bc4c0;received=80.237.160.128;rport=5060 From: "Asterisk" ;tag=as64c30ca3 To: ;tag=as032ec101 Call-ID: 51d888151d1ae1b932b81bd83346ac29@80.237.160.128:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 80.237.160.128:5060;branch=z9hG4bK297bc4c0;received=80.237.160.128;rport=5060 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 2 [ 61]: From: "Asterisk" ;tag=as64c30ca3 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 3 [ 48]: To: ;tag=as032ec101 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 4 [ 61]: Call-ID: 51d888151d1ae1b932b81bd83346ac29@80.237.160.128:5060 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 9 [ 34]: Contact: [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 12 [ 0]: [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 80.237.160.128:5060 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '51d888151d1ae1b932b81bd83346ac29@80.237.160.128:5060' in 32000 ms (Method: OPTIONS) [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;received=114.77.232.125;rport=5060 From: ;tag=as4cc030a8 To: ;tag=as5e263bdf Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1003 REGISTER Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 06 Sep 2010 19:52:08 GMT Content-Length: 0 <-------------> [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;received=114.77.232.125;rport=5060 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 2 [ 45]: From: ;tag=as4cc030a8 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 3 [ 43]: To: ;tag=as5e263bdf [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1003 REGISTER [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 9 [ 12]: Expires: 120 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 10 [ 50]: Contact: ;expires=120 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 11 [ 35]: Date: Mon, 06 Sep 2010 19:52:08 GMT [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: --- (13 headers 0 lines) --- [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 (Checking To) --From tag as4cc030a8 --To-tag as5e263bdf [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16769 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Stopping retransmission on '3a0375611ec2831f427a14264de70562@192.168.0.1' of Request 1003: Match Found [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Registration successful [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Cancelling timeout 16767 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: SIP Registry 80.237.160.128: refcount now 2 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: SIP Registry 80.237.160.128: refcount now 1 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '3a0375611ec2831f427a14264de70562@192.168.0.1' in 32000 ms (Method: REGISTER) [Sep 7 05:52:08] NOTICE[10974] chan_sip.c: Outbound Registration: Expiry for 80.237.160.128 is 120 sec (Scheduling reregistration in 105 s) [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: SIP Registry 80.237.160.128: refcount now 2 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=7.62E+06, Et=8.24E+08, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=5.29E+07, Et=1.89E+09, s/n= 0.03 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=6.43E+07, Et=1.80E+09, s/n= 0.04 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=9.14E+06, Et=1.40E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=9.98E+06, Et=1.82E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.65E+07, Et=4.07E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=4.59E+07, Et=3.44E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=5.69E+07, Et=5.26E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.14E+07, Et=4.71E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.09E+08, Et=5.19E+09, s/n= 0.04 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=3.11E+07, Et=8.76E+09, s/n= 0.00 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;received=114.77.232.125;rport=5060 From: ;tag=as4cc030a8 To: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1003 REGISTER Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;received=114.77.232.125;rport=5060 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 2 [ 45]: From: ;tag=as4cc030a8 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 3 [ 28]: To: [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1003 REGISTER [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: --- (10 headers 0 lines) --- [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 (Checking To) --From tag as4cc030a8 --To-tag [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:80.237.160.128:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;received=114.77.232.125;rport=5060 From: ;tag=as4cc030a8 To: ;tag=as5e263bdf Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 CSeq: 1003 REGISTER Server: Asterisk PBX SVN--r Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="smartbyte.de", nonce="0c50098e", stale=true Content-Length: 0 <-------------> [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 114.77.232.125:5060;branch=z9hG4bK6d745ac8;received=114.77.232.125;rport=5060 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 2 [ 45]: From: ;tag=as4cc030a8 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 3 [ 43]: To: ;tag=as5e263bdf [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 4 [ 53]: Call-ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 5 [ 19]: CSeq: 1003 REGISTER [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 6 [ 27]: Server: Asterisk PBX SVN--r [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 9 [ 90]: WWW-Authenticate: Digest algorithm=MD5, realm="smartbyte.de", nonce="0c50098e", stale=true [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:52:08] VERBOSE[10974] chan_sip.c: --- (11 headers 0 lines) --- [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: = Looking for Call ID: 3a0375611ec2831f427a14264de70562@192.168.0.1 (Checking To) --From tag as4cc030a8 --To-tag as5e263bdf [Sep 7 05:52:08] DEBUG[10974] chan_sip.c: Stopping retransmission on '3a0375611ec2831f427a14264de70562@192.168.0.1' of Request 1003: Match Not Found [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=4.18E+07, Et=7.68E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=4.64E+07, Et=5.08E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.18E+08, Et=9.65E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=5.41E+08, Et=7.45E+09, s/n= 0.08 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.32E+07, Et=7.14E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.28E+07, Et=5.58E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.48E+07, Et=4.90E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=4.57E+04, Et=5.47E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.06E+08, Et=5.33E+09, s/n= 0.02 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.53E+07, Et=3.00E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=7.43E+07, Et=4.52E+09, s/n= 0.02 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=4.57E+05, Et=2.64E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.92E+07, Et=2.20E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=6.36E+06, Et=1.78E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.40E+07, Et=1.84E+09, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.72E+06, Et=1.74E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=4.71E+06, Et=1.35E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.23E+07, Et=1.04E+09, s/n= 0.02 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.58E+06, Et=1.15E+09, s/n= 0.00 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=1.15E+07, Et=8.36E+08, s/n= 0.01 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=3.09E+07, Et=9.82E+08, s/n= 0.03 [Sep 7 05:52:08] DEBUG[18297] dsp.c: tone 1100, Ew=2.28E+07, Et=1.01E+09, s/n= 0.02 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.64E+06, Et=1.44E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.13E+06, Et=8.52E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.53E+07, Et=1.26E+09, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.11E+07, Et=8.78E+08, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=7.91E+05, Et=8.01E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=8.06E+06, Et=7.41E+08, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=6.35E+06, Et=6.14E+08, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.99E+07, Et=7.98E+08, s/n= 0.03 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=2.23E+06, Et=1.05E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=7.48E+06, Et=4.51E+08, s/n= 0.02 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=5.96E+06, Et=8.61E+08, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.02E+07, Et=7.78E+08, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=2.55E+06, Et=7.34E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.70E+06, Et=5.26E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.87E+06, Et=4.43E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=4.21E+05, Et=3.92E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=9.86E+05, Et=4.28E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.20E+07, Et=4.75E+08, s/n= 0.03 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.09E+07, Et=5.14E+08, s/n= 0.02 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.62E+06, Et=1.26E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=8.01E+05, Et=1.88E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.54E+06, Et=7.61E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=6.10E+06, Et=4.54E+08, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=8.18E+06, Et=5.00E+08, s/n= 0.02 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=2.39E+05, Et=4.54E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.80E+06, Et=9.26E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.25E+07, Et=1.35E+09, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=6.21E+08, Et=2.19E+10, s/n= 0.03 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.28E+07, Et=3.06E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.92E+07, Et=1.67E+09, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.09E+08, Et=1.07E+10, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=6.49E+07, Et=1.92E+09, s/n= 0.04 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=2.84E+06, Et=1.73E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=4.14E+07, Et=1.94E+10, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=7.40E+06, Et=2.27E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.22E+07, Et=1.25E+09, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=4.82E+07, Et=1.31E+09, s/n= 0.04 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=2.60E+06, Et=1.31E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=2.24E+06, Et=1.91E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=3.16E+05, Et=8.89E+08, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=2.23E+06, Et=1.19E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=5.64E+06, Et=1.35E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.12E+07, Et=1.24E+09, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=3.54E+08, Et=1.19E+10, s/n= 0.03 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=6.46E+06, Et=1.67E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.34E+07, Et=1.37E+09, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=4.77E+06, Et=1.26E+09, s/n= 0.00 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=1.07E+07, Et=9.93E+08, s/n= 0.01 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=2.14E+07, Et=5.56E+08, s/n= 0.04 [Sep 7 05:52:09] DEBUG[18297] dsp.c: tone 1100, Ew=3.08E+06, Et=7.81E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=3.85E+06, Et=7.20E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=3.82E+06, Et=1.29E+09, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=4.83E+06, Et=7.94E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=5.11E+07, Et=1.33E+09, s/n= 0.04 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.24E+06, Et=1.20E+09, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.82E+06, Et=8.93E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=3.12E+07, Et=8.09E+08, s/n= 0.04 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=5.85E+06, Et=7.84E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.31E+07, Et=8.63E+08, s/n= 0.03 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.95E+07, Et=7.06E+08, s/n= 0.04 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=8.42E+05, Et=6.04E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.46E+07, Et=5.60E+08, s/n= 0.05 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.85E+07, Et=6.31E+08, s/n= 0.05 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.28E+07, Et=5.54E+08, s/n= 0.02 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.80E+07, Et=4.58E+08, s/n= 0.04 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.02E+06, Et=3.49E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.75E+07, Et=3.67E+08, s/n= 0.08 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=3.10E+05, Et=4.87E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.25E+06, Et=3.06E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=3.22E+06, Et=3.17E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.10E+06, Et=2.51E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=6.02E+06, Et=2.04E+08, s/n= 0.03 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=7.91E+06, Et=3.49E+08, s/n= 0.02 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=5.21E+07, Et=2.30E+08, s/n= 0.29 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=9.71E+06, Et=2.61E+08, s/n= 0.04 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=6.91E+05, Et=2.96E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=3.57E+06, Et=4.05E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.70E+06, Et=3.22E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.42E+06, Et=3.55E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=9.87E+06, Et=2.11E+08, s/n= 0.05 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.52E+06, Et=1.57E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.28E+05, Et=1.82E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.87E+06, Et=1.51E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=9.40E+04, Et=1.57E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.00E+05, Et=1.27E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.55E+06, Et=1.03E+08, s/n= 0.03 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=4.64E+06, Et=1.59E+08, s/n= 0.03 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=3.31E+06, Et=1.10E+08, s/n= 0.03 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=7.82E+05, Et=1.53E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.78E+06, Et=1.99E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.15E+07, Et=2.37E+08, s/n= 0.05 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=9.36E+06, Et=2.71E+08, s/n= 0.04 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.01E+07, Et=2.21E+08, s/n= 0.05 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.25E+07, Et=2.90E+08, s/n= 0.05 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=4.06E+06, Et=1.59E+08, s/n= 0.03 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.52E+06, Et=1.39E+08, s/n= 0.02 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.10E+05, Et=3.37E+07, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=1.40E+06, Et=1.89E+08, s/n= 0.01 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=5.35E+05, Et=3.82E+08, s/n= 0.00 [Sep 7 05:52:10] DEBUG[18297] dsp.c: tone 1100, Ew=2.09E+06, Et=5.50E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.68E+06, Et=4.75E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.29E+06, Et=4.06E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.82E+06, Et=4.29E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.28E+06, Et=7.09E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=4.07E+05, Et=5.35E+09, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=4.73E+05, Et=8.59E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=5.92E+05, Et=4.65E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=4.81E+06, Et=7.53E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=9.13E+06, Et=1.21E+09, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.06E+07, Et=1.57E+10, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=8.34E+05, Et=6.67E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=8.61E+06, Et=4.40E+08, s/n= 0.02 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=5.82E+06, Et=3.71E+08, s/n= 0.02 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.28E+06, Et=3.37E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.16E+06, Et=3.48E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.98E+05, Et=8.27E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.52E+06, Et=2.53E+09, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.50E+07, Et=2.37E+09, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.81E+05, Et=1.85E+09, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.94E+06, Et=9.95E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=6.37E+06, Et=7.34E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: Auto destroying SIP dialog '7553546773a2f5ab51c49c5c58112826@192.168.0.1' [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: Destroying SIP dialog 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:52:11] VERBOSE[10974] chan_sip.c: Really destroying SIP dialog '7553546773a2f5ab51c49c5c58112826@192.168.0.1' Method: REGISTER [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: ---------- SIP HISTORY for '7553546773a2f5ab51c49c5c58112826@192.168.0.1' [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: * SIP Call [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 001. RegistryInit Account: smartbyt@aphone6.tpg.com.au [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 002. AuthResp Auth response sent for smartbyt in realm asterisk - nc 2 [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 003. TxReqRel REGISTER / 1009 REGISTER - REGISTER [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 004. Rx SIP/2.0 / 1009 REGISTER / 100 Trying [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 005. Ignore Ignoring this retransmit [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 006. Rx SIP/2.0 / 1009 REGISTER / 401 Unauthorized [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 007. AuthResp Auth response sent for smartbyt in realm asterisk - nc 1 [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 008. RegistryAuth Try: 1 [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 009. TxReqRel REGISTER / 1010 REGISTER - REGISTER [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 010. Rx SIP/2.0 / 1010 REGISTER / 100 Trying [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 011. Ignore Ignoring this retransmit [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 012. Rx SIP/2.0 / 1010 REGISTER / 200 OK [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 013. SchedDestroy 32000 ms [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: 014. AutoDestroy 7553546773a2f5ab51c49c5c58112826@192.168.0.1 [Sep 7 05:52:11] DEBUG[10974] chan_sip.c: ---------- END SIP HISTORY for '7553546773a2f5ab51c49c5c58112826@192.168.0.1' [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.27E+06, Et=4.78E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=9.22E+05, Et=2.50E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.06E+06, Et=2.93E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.06E+07, Et=4.16E+08, s/n= 0.05 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.23E+06, Et=1.09E+09, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.18E+07, Et=3.44E+09, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=8.41E+05, Et=1.01E+09, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.18E+06, Et=8.85E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.04E+06, Et=4.10E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.25E+05, Et=2.58E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=4.70E+06, Et=1.70E+08, s/n= 0.03 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.71E+05, Et=1.36E+08, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=7.89E+06, Et=1.34E+08, s/n= 0.06 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.61E+06, Et=1.26E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.19E+06, Et=1.09E+08, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.55E+06, Et=1.01E+08, s/n= 0.02 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.20E+05, Et=8.54E+07, s/n= 0.00 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.03E+06, Et=7.03E+07, s/n= 0.03 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=4.18E+05, Et=6.52E+07, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.20E+06, Et=7.36E+07, s/n= 0.02 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=7.62E+05, Et=5.15E+07, s/n= 0.02 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=6.52E+05, Et=5.00E+07, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.05E+05, Et=5.14E+07, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=2.98E+05, Et=4.56E+07, s/n= 0.01 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=6.19E+05, Et=3.50E+07, s/n= 0.02 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.48E+06, Et=4.19E+07, s/n= 0.04 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=1.38E+06, Et=3.69E+07, s/n= 0.04 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=7.41E+05, Et=4.45E+07, s/n= 0.02 [Sep 7 05:52:11] DEBUG[18297] dsp.c: tone 1100, Ew=3.21E+05, Et=3.96E+07, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=3.58E+05, Et=5.60E+07, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.67E+05, Et=5.13E+07, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.79E+05, Et=4.75E+07, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=6.85E+05, Et=3.54E+07, s/n= 0.02 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=8.24E+03, Et=4.38E+07, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.51E+05, Et=3.83E+07, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=6.94E+05, Et=4.39E+07, s/n= 0.02 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.65E+05, Et=7.14E+07, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=9.17E+04, Et=3.88E+07, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=5.27E+05, Et=5.95E+07, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=8.29E+04, Et=6.12E+07, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.01E+05, Et=5.88E+07, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.51E+05, Et=1.18E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=6.03E+05, Et=2.71E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.02E+05, Et=5.56E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=5.18E+06, Et=7.55E+08, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=5.90E+06, Et=4.56E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=3.10E+05, Et=3.36E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.65E+06, Et=1.69E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.80E+06, Et=1.08E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=5.22E+06, Et=1.64E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=4.57E+05, Et=1.97E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.07E+06, Et=3.75E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=8.78E+06, Et=3.05E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.02E+07, Et=1.64E+09, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.63E+06, Et=2.36E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=5.97E+06, Et=2.65E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.30E+07, Et=2.83E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=9.53E+06, Et=3.55E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.89E+07, Et=2.70E+09, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.64E+06, Et=2.40E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.25E+06, Et=2.74E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=3.06E+06, Et=1.69E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.67E+05, Et=1.03E+09, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.18E+06, Et=6.63E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.03E+06, Et=3.14E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.26E+05, Et=4.20E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=3.95E+05, Et=2.58E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.06E+06, Et=1.77E+08, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=7.00E+05, Et=1.94E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=4.75E+05, Et=2.01E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.96E+05, Et=1.28E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=3.78E+05, Et=1.16E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=4.10E+05, Et=1.18E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=3.64E+05, Et=1.22E+08, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=7.62E+05, Et=1.50E+08, s/n= 0.01 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.83E+05, Et=6.11E+07, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=3.98E+05, Et=9.37E+07, s/n= 0.00 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=2.50E+06, Et=8.56E+07, s/n= 0.03 [Sep 7 05:52:12] DEBUG[18297] dsp.c: tone 1100, Ew=1.47E+06, Et=9.09E+07, s/n= 0.02 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=5.24E+05, Et=7.72E+07, s/n= 0.01 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=4.70E+05, Et=6.33E+07, s/n= 0.01 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=1.93E+05, Et=5.52E+07, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=1.28E+05, Et=4.70E+07, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=5.64E+04, Et=4.30E+07, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=4.81E+05, Et=7.23E+07, s/n= 0.01 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=2.08E+06, Et=9.18E+07, s/n= 0.02 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=6.65E+05, Et=2.59E+08, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=4.12E+05, Et=5.35E+07, s/n= 0.01 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=1.21E+07, Et=4.04E+08, s/n= 0.03 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=7.10E+08, Et=1.74E+11, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=2.58E+07, Et=1.35E+11, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=2.89E+06, Et=1.24E+10, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=1.66E+05, Et=1.28E+09, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=8.59E+04, Et=3.14E+08, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=1.12E+06, Et=1.00E+09, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=2.28E+05, Et=2.35E+08, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=5.74E+06, Et=1.46E+08, s/n= 0.04 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=2.79E+06, Et=1.18E+08, s/n= 0.02 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=5.99E+05, Et=2.76E+08, s/n= 0.00 [Sep 7 05:52:13] DEBUG[18297] dsp.c: tone 1100, Ew=1.33E+06, Et=2.64E+08, s/n= 0.01 [Sep 7 05:52:13] VERBOSE[10974] chan_sip.c: <--- SIP read from UDP:192.168.1.27:5060 ---> BYE sip:803@192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK80017eea5db8df1190ea0015afdef833;rport From: ;tag=1895770013 To: ;tag=as78829a27 Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 7 BYE Contact: Authorization: Digest username="105", realm="home.smartbyte.de", nonce="762dcaa9", uri="sip:803@192.168.1.1:5060", response="0584fa987bc66725dc601f7f4b89cdb7", algorithm=MD5 Max-Forwards: 70 User-Agent: SIPPER for PhonerLite Content-Length: 0 <-------------> [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 0 [ 36]: BYE sip:803@192.168.1.1:5060 SIP/2.0 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK80017eea5db8df1190ea0015afdef833;rport [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 3 [ 46]: To: ;tag=as78829a27 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 5 [ 11]: CSeq: 7 BYE [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 6 [ 36]: Contact: [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 7 [173]: Authorization: Digest username="105", realm="home.smartbyte.de", nonce="762dcaa9", uri="sip:803@192.168.1.1:5060", response="0584fa987bc66725dc601f7f4b89cdb7", algorithm=MD5 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 9 [ 33]: User-Agent: SIPPER for PhonerLite [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Sep 7 05:52:13] VERBOSE[10974] chan_sip.c: --- (11 headers 0 lines) --- [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: = Looking for Call ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 (Checking From) --From tag 1895770013 --To-tag as78829a27 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Initializing initreq for method BYE - callid 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:13] DEBUG[10974] netsock2.c: Splitting '192.168.1.27:5060' gives... [Sep 7 05:52:13] DEBUG[10974] netsock2.c: ...host '192.168.1.27' and port '5060'. [Sep 7 05:52:13] VERBOSE[10974] chan_sip.c: Sending to 192.168.1.27:5060 (NAT) [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Setting SIP_ALREADYGONE on dialog 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:13] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x429748d0' [Sep 7 05:52:13] DEBUG[10974] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x429d2c58' [Sep 7 05:52:13] VERBOSE[10974] chan_sip.c: Scheduling destruction of SIP dialog '00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27' in 32000 ms (Method: BYE) [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Received bye, issuing owner hangup [Sep 7 05:52:13] VERBOSE[10974] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK80017eea5db8df1190ea0015afdef833;received=192.168.1.27;rport=5060 From: ;tag=1895770013 To: ;tag=as78829a27 Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 CSeq: 7 BYE Server: Asterisk PBX SVN-trunk-r285058 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 1 [114]: Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK80017eea5db8df1190ea0015afdef833;received=192.168.1.27;rport=5060 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 2 [ 43]: From: ;tag=1895770013 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 3 [ 46]: To: ;tag=as78829a27 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 4 [ 58]: Call-ID: 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 5 [ 11]: CSeq: 7 BYE [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 6 [ 38]: Server: Asterisk PBX SVN-trunk-r285058 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Header 10 [ 0]: [Sep 7 05:52:13] DEBUG[10974] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.27:5060 [Sep 7 05:52:13] NOTICE[18297] chan_sip.c: Unknown option: 5 [Sep 7 05:52:13] DEBUG[18297] channel.c: Set channel SIP/105-0000001d to write format alaw [Sep 7 05:52:13] DEBUG[18297] pbx.c: Spawn extension (default,803,1) exited non-zero on 'SIP/105-0000001d' [Sep 7 05:52:13] VERBOSE[18297] pbx.c: == Spawn extension (default, 803, 1) exited non-zero on 'SIP/105-0000001d' [Sep 7 05:52:13] DEBUG[18297] channel.c: Soft-Hanging up channel 'SIP/105-0000001d' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Result of 'DIALSTATUS' is NULL [Sep 7 05:52:13] DEBUG[18297] pbx.c: Launching 'Goto' [Sep 7 05:52:13] VERBOSE[18297] pbx.c: -- Executing [h@default:1] Goto("SIP/105-0000001d", "dialstatus,,1") in new stack [Sep 7 05:52:13] VERBOSE[18297] pbx.c: -- Goto (dialstatus,h,1) [Sep 7 05:52:13] DEBUG[18297] channel.c: Hanging up channel 'SIP/105-0000001d' [Sep 7 05:52:13] DEBUG[18297] chan_sip.c: Hangup call SIP/105-0000001d, SIP callid 00B720E5-5DB8-DF11-90E8-0015AFDEF833@192.168.1.27 [Sep 7 05:52:13] DEBUG[18297] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x429748d0' [Sep 7 05:52:13] DEBUG[18297] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x429d2c58' [Sep 7 05:52:13] DEBUG[18297] cdr_mysql.c: Inserting a CDR record. [Sep 7 05:52:13] DEBUG[18297] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid) VALUES ('2010-09-07 05:52:05','105','105','803','default','SIP/105-0000001d','ChanSpy','SIP/100,q','8','8','ANSWERED','3','1283802725.41') [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '2010-09-07 05:52:05' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '105' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is 'default' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is 'SIP/105-0000001d' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '(null)' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is 'ChanSpy' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is 'SIP/100,q' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '8' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '8' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is 'ANSWERED' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is 'DOCUMENTATION' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '(null)' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '1283802725.41' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '(null)' [Sep 7 05:52:13] DEBUG[18297] pbx.c: Function result is '(null)' [Sep 7 05:52:13] DEBUG[18297] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2010-09-07 05:52:05','105','default','SIP/105-0000001d','','ChanSpy','SIP/100,q','8','8','ANSWERED','DOCUMENTATION','','1283802725.41','','') [Sep 7 05:52:13] DEBUG[10957] devicestate.c: No provider found, checking channel drivers for SIP - 105 [Sep 7 05:52:13] DEBUG[10957] chan_sip.c: Checking device state for peer 105 [Sep 7 05:52:13] DEBUG[10957] devicestate.c: Changing state for SIP/105 - state 1 (Not in use) [Sep 7 05:52:13] DEBUG[10957] devicestate.c: device 'SIP/105' state '1' [Sep 7 05:52:13] DEBUG[11003] app_queue.c: Device 'SIP/105' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: Auto destroying SIP dialog '4c76c12233a7c5d546609dc16317a534@192.168.0.1' [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: Destroying SIP dialog 4c76c12233a7c5d546609dc16317a534@192.168.0.1 [Sep 7 05:52:18] VERBOSE[10974] chan_sip.c: Really destroying SIP dialog '4c76c12233a7c5d546609dc16317a534@192.168.0.1' Method: REGISTER [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: ---------- SIP HISTORY for '4c76c12233a7c5d546609dc16317a534@192.168.0.1' [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: * SIP Call [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: 001. RegistryInit Account: 17474987674@proxy01.sipphone.com [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: 002. AuthResp Auth response sent for 17474987674 in realm proxy01.sipphone.co [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: 003. TxReqRel REGISTER / 708 REGISTER - REGISTER [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: 004. Rx SIP/2.0 / 708 REGISTER / 200 OK [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: 005. SchedDestroy 32000 ms [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: 006. AutoDestroy 4c76c12233a7c5d546609dc16317a534@192.168.0.1 [Sep 7 05:52:18] DEBUG[10974] chan_sip.c: ---------- END SIP HISTORY for '4c76c12233a7c5d546609dc16317a534@192.168.0.1' [Sep 7 05:52:22] NOTICE[18214] res_stun_monitor.c: STUN monitor stopped [Sep 7 05:52:22] VERBOSE[18214] res_snmp.c: Unloading [Sub]Agent Module [Sep 7 05:52:22] VERBOSE[18214] res_security_log.c: -- Security Logging Disabled [Sep 7 05:52:22] VERBOSE[18214] rtp_engine.c: == Unregistered RTP engine 'multicast' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered custom function PP_EACH_USER [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered custom function PP_EACH_EXTENSION [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered custom function MUTEAUDIO [Sep 7 05:52:22] VERBOSE[18214] manager.c: == Manager unregistered action MuteAudio [Sep 7 05:52:22] VERBOSE[18214] res_musiconhold.c: == Destroying musiconhold processes [Sep 7 05:52:22] DEBUG[18214] res_musiconhold.c: Destroying MOH class 'native-random' [Sep 7 05:52:22] DEBUG[18214] res_musiconhold.c: Destroying MOH class 'default' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'MusicOnHold' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'WaitMusicOnHold' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'SetMusicOnHold' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'StartMusicOnHold' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'StopMusicOnHold' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'Monitor' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'StopMonitor' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'ChangeMonitor' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'PauseMonitor' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'UnpauseMonitor' [Sep 7 05:52:22] VERBOSE[18214] manager.c: == Manager unregistered action Monitor [Sep 7 05:52:22] VERBOSE[18214] manager.c: == Manager unregistered action StopMonitor [Sep 7 05:52:22] VERBOSE[18214] manager.c: == Manager unregistered action ChangeMonitor [Sep 7 05:52:22] VERBOSE[18214] manager.c: == Manager unregistered action PauseMonitor [Sep 7 05:52:22] VERBOSE[18214] manager.c: == Manager unregistered action UnpauseMonitor [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'JabberSend' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'JabberSendGroup' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'JabberStatus' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'JabberJoin' [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered application 'JabberLeave' [Sep 7 05:52:22] VERBOSE[18214] manager.c: == Manager unregistered action JabberSend [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered custom function JABBER_STATUS [Sep 7 05:52:22] VERBOSE[18214] pbx.c: == Unregistered custom function JABBER_RECEIVE [Sep 7 05:52:22] DEBUG[18214] res_jabber.c: JABBER: Releasing and disconnecting client: joern.krebs.asterisk [Sep 7 05:52:22] VERBOSE[18214] res_jabber.c: > JABBER: Disconnecting