[May 23 13:11:59] VERBOSE[19457] config.c: == Parsing '/etc/asterisk-2011-05-03/logger.conf': [May 23 13:11:59] DEBUG[19457] config.c: Parsing /etc/asterisk-2011-05-03/logger.conf [May 23 13:11:59] VERBOSE[19457] config.c: == Found [May 23 13:11:59] VERBOSE[19457] logger.c: Asterisk Queue Logger restarted [May 23 13:11:59] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 5701dc5818ba94e900d96f822be98c7e@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:11:59] DEBUG[13067] acl.c: For destination '64.19.145.20', our source address is '64.19.145.13'. [May 23 13:11:59] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:11:59] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060 [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.20 SIP/2.0 [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1ff7052b [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as622089bb [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060 [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:11:59 GMT [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:11:59] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:11:59] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1ff7052b Max-Forwards: 70 From: "unknown" ;tag=as622089bb To: Contact: Call-ID: 068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:11:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:11:59] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042568 [May 23 13:11:59] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:00] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-229b26b2 From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113874 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-229b26b2 [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113874 NOTIFY [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:00] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:00] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:00] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:00] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:00] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 113874, ours 113874) [May 23 13:12:00] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:00] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-229b26b2;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113874 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:00] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:00] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:00] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042568:OPTIONS (Method 3) (No timer T1) [May 23 13:12:00] VERBOSE[13067] chan_sip.c: Retransmitting #1 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1ff7052b Max-Forwards: 70 From: "unknown" ;tag=as622089bb To: Contact: Call-ID: 068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:11:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:00] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:01] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042568:OPTIONS (Method 3) (No timer T1) [May 23 13:12:01] VERBOSE[13067] chan_sip.c: Retransmitting #2 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1ff7052b Max-Forwards: 70 From: "unknown" ;tag=as622089bb To: Contact: Call-ID: 068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:11:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:01] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:01] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> NOTIFY sip:64.19.145.13 SIP/2.0 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 28173 NOTIFY Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-dbacdc47 Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Contact: "SPA303 Cisco" Event: keep-alive Content-Length: 0 <-------------> [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 2 [ 22]: To: [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 3 [ 38]: Call-ID: 5d07fe66-394bec48@10.0.15.101 [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 4 [ 18]: CSeq: 28173 NOTIFY [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 5 [ 60]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-dbacdc47 [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 7 [ 30]: User-Agent: Cisco/SPA303-7.4.6 [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 8 [ 82]: Contact: "SPA303 Cisco" [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Event: keep-alive [May 23 13:12:01] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:01] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:01] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5d07fe66-394bec48@10.0.15.101 (Checking From) --From tag c7d0e91e95d40f0o0 --To-tag [May 23 13:12:01] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:01] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:01] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-dbacdc47;received=209.191.39.117 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: ;tag=as1046500e Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 28173 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:01] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:01] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '5d07fe66-394bec48@10.0.15.101' in 32000 ms (Method: NOTIFY) [May 23 13:12:02] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042568:OPTIONS (Method 3) (No timer T1) [May 23 13:12:02] VERBOSE[13067] chan_sip.c: Retransmitting #3 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1ff7052b Max-Forwards: 70 From: "unknown" ;tag=as622089bb To: Contact: Call-ID: 068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:11:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:02] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:03] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042568:OPTIONS (Method 3) (No timer T1) [May 23 13:12:03] VERBOSE[13067] chan_sip.c: Retransmitting #4 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1ff7052b Max-Forwards: 70 From: "unknown" ;tag=as622089bb To: Contact: Call-ID: 068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:11:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:03] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:03] DEBUG[13067] chan_sip.c: Destroying SIP dialog 068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060 [May 23 13:12:03] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '068822ec5041e7677b2a3aa34a18074a@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:04] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-229b26b2 From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113874 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-229b26b2 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113874 NOTIFY [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:04] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:04] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:04] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:04] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 113874, ours 113874) [May 23 13:12:04] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:04] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-229b26b2;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113874 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:04] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:04] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:04] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> REGISTER sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-63caf9c9 From: ;tag=5e35c995200173e1o3 To: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94304 REGISTER Max-Forwards: 70 Authorization: Digest username="175-eng",realm="asterisk",nonce="268cbfa2",uri="sip:64.19.145.13",algorithm=MD5,response="150399ff726429e7b67e644f71bf6755" Contact: ;expires=3600 User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 0 [ 33]: REGISTER sip:64.19.145.13 SIP/2.0 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-63caf9c9 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 3 [ 30]: To: [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 5 [ 20]: CSeq: 94304 REGISTER [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 7 [155]: Authorization: Digest username="175-eng",realm="asterisk",nonce="268cbfa2",uri="sip:64.19.145.13",algorithm=MD5,response="150399ff726429e7b67e644f71bf6755" [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 8 [ 55]: Contact: ;expires=3600 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 11 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 12 [ 19]: Supported: replaces [May 23 13:12:04] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:04] DEBUG[13067] chan_sip.c: = Looking for Call ID: b7b02bc6-e8d28b72@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:04] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:04] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:04] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:04] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:04] DEBUG[13067] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [May 23 13:12:04] DEBUG[13067] chan_sip.c: Initializing initreq for method REGISTER - callid b7b02bc6-e8d28b72@192.168.15.187 [May 23 13:12:04] DEBUG[13067] netsock2.c: Splitting '192.168.15.187:5063' gives... [May 23 13:12:04] DEBUG[13067] netsock2.c: ...host '192.168.15.187' and port '5063'. [May 23 13:12:04] VERBOSE[13067] chan_sip.c: Sending to 209.191.13.243:17616 (NAT) [May 23 13:12:04] NOTICE[13067] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=5e35c995200173e1o3' [May 23 13:12:04] VERBOSE[13067] chan_sip.c: <--- Transmitting (NAT) to 209.191.13.243:17616 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-63caf9c9;received=209.191.13.243;rport=17616 From: ;tag=5e35c995200173e1o3 To: ;tag=as1f6b40e0 Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94304 REGISTER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d5fbf24", stale=true Content-Length: 0 <------------> [May 23 13:12:04] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 209.191.13.243:17616 [May 23 13:12:04] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'b7b02bc6-e8d28b72@192.168.15.187' in 32000 ms (Method: REGISTER) [May 23 13:12:04] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> REGISTER sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-a472f8fa From: ;tag=5e35c995200173e1o3 To: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94305 REGISTER Max-Forwards: 70 Authorization: Digest username="175-eng",realm="asterisk",nonce="2d5fbf24",uri="sip:64.19.145.13",algorithm=MD5,response="6b3ef8867b6e7df7dae858c862a04f42" Contact: ;expires=3600 User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 0 [ 33]: REGISTER sip:64.19.145.13 SIP/2.0 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-a472f8fa [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 3 [ 30]: To: [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 5 [ 20]: CSeq: 94305 REGISTER [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 7 [155]: Authorization: Digest username="175-eng",realm="asterisk",nonce="2d5fbf24",uri="sip:64.19.145.13",algorithm=MD5,response="6b3ef8867b6e7df7dae858c862a04f42" [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 8 [ 55]: Contact: ;expires=3600 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 11 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [May 23 13:12:04] DEBUG[13067] chan_sip.c: Header 12 [ 19]: Supported: replaces [May 23 13:12:04] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:04] DEBUG[13067] chan_sip.c: = Looking for Call ID: b7b02bc6-e8d28b72@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:04] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:04] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:04] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:04] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:04] DEBUG[13067] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [May 23 13:12:04] DEBUG[13067] chan_sip.c: Initializing initreq for method REGISTER - callid b7b02bc6-e8d28b72@192.168.15.187 [May 23 13:12:04] DEBUG[13067] netsock2.c: Splitting '192.168.15.187:5063' gives... [May 23 13:12:04] DEBUG[13067] netsock2.c: ...host '192.168.15.187' and port '5063'. [May 23 13:12:04] VERBOSE[13067] chan_sip.c: Sending to 209.191.13.243:17616 (NAT) [May 23 13:12:04] DEBUG[13067] chan_sip.c: Store REGISTER's src-IP:port for call routing. [May 23 13:12:04] DEBUG[13109] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/175-eng PeerStatus: Registered Address: 209.191.13.243:17616 [May 23 13:12:04] VERBOSE[13067] chan_sip.c: <--- Transmitting (NAT) to 209.191.13.243:17616 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-a472f8fa;received=209.191.13.243;rport=17616 From: ;tag=5e35c995200173e1o3 To: ;tag=as1f6b40e0 Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94305 REGISTER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Mon, 23 May 2011 17:12:04 GMT Content-Length: 0 <------------> [May 23 13:12:04] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:17616 [May 23 13:12:04] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 175-eng [May 23 13:12:04] DEBUG[13069] chan_sip.c: Checking device state for peer 175-eng [May 23 13:12:04] DEBUG[13069] devicestate.c: Changing state for SIP/175-eng - state 1 (Not in use) [May 23 13:12:04] DEBUG[13069] devicestate.c: device 'SIP/175-eng' state '1' [May 23 13:12:04] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'b7b02bc6-e8d28b72@192.168.15.187' in 32000 ms (Method: REGISTER) [May 23 13:12:04] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/175-eng MemberName: SIP/175-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:04] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: supporthotline-eng Location: SIP/175-eng MemberName: SIP/175-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:04] DEBUG[13094] app_queue.c: Device 'SIP/175-eng' changed to state '1' (Not in use) [May 23 13:12:05] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 3798a7fb38fddb936a9b9f4005266c8b@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:05] DEBUG[13067] acl.c: For destination '209.191.13.243', our source address is '64.19.145.13'. [May 23 13:12:05] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:05] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060 [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 0 [ 46]: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59377900;rport [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as226fa9bc [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 4 [ 36]: To: [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060 [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:05 GMT [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:05] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:05] VERBOSE[13067] chan_sip.c: Reliably Transmitting (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59377900;rport Max-Forwards: 70 From: "unknown" ;tag=as226fa9bc To: Contact: Call-ID: 435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:05] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042577 [May 23 13:12:05] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:06] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042577:OPTIONS (Method 3) (No timer T1) [May 23 13:12:06] VERBOSE[13067] chan_sip.c: Retransmitting #1 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59377900;rport Max-Forwards: 70 From: "unknown" ;tag=as226fa9bc To: Contact: Call-ID: 435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:06] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:06] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.44.130:5060 ---> OPTIONS sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK4a77e0af;rport Max-Forwards: 70 From: "asterisk" ;tag=as1b59f9d2 To: Contact: Call-ID: 1a4ec89b671a3caa051af79e59e921e6@209.191.44.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 17:12:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK4a77e0af;rport [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as1b59f9d2 [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 5 [ 38]: Contact: [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 6 [ 56]: Call-ID: 1a4ec89b671a3caa051af79e59e921e6@209.191.44.130 [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:06 GMT [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 13:12:06] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 13:12:06] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:06] DEBUG[13067] chan_sip.c: = Looking for Call ID: 1a4ec89b671a3caa051af79e59e921e6@209.191.44.130 (Checking From) --From tag as1b59f9d2 --To-tag [May 23 13:12:06] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 13:12:06] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:06] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 1a4ec89b671a3caa051af79e59e921e6@209.191.44.130 - OPTIONS (No RTP) [May 23 13:12:06] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 13:12:06] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 13:12:06] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.44.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK4a77e0af;rport;received=209.191.44.130 From: "asterisk" ;tag=as1b59f9d2 To: ;tag=as29d25ea0 Call-ID: 1a4ec89b671a3caa051af79e59e921e6@209.191.44.130 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> [May 23 13:12:06] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 13:12:06] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '1a4ec89b671a3caa051af79e59e921e6@209.191.44.130' in 32000 ms (Method: OPTIONS) [May 23 13:12:07] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 089833ef052a097b7899482c1c833c28@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:07] DEBUG[13067] acl.c: For destination '64.19.145.18', our source address is '64.19.145.13'. [May 23 13:12:07] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.18 SIP/2.0 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK05085263 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as6fe34302 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:07 GMT [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.18:5060: OPTIONS sip:64.19.145.18 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK05085263 Max-Forwards: 70 From: "unknown" ;tag=as6fe34302 To: Contact: Call-ID: 28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042580 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.18:5060 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.18:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK05085263;received=64.19.145.13 From: "unknown" ;tag=as6fe34302 To: ;tag=as35ce4ff3 Call-ID: 28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK05085263;received=64.19.145.13 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as6fe34302 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as35ce4ff3 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: = Looking for Call ID: 28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060 (Checking To) --From tag as6fe34302 --To-tag as35ce4ff3 [May 23 13:12:07] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042580 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Stopping retransmission on '28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:07] DEBUG[13067] chan_sip.c: Destroying SIP dialog 28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '28a0c2fa4a6526fd238a97a00bbcbe23@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042577:OPTIONS (Method 3) (No timer T1) [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Retransmitting #2 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59377900;rport Max-Forwards: 70 From: "unknown" ;tag=as226fa9bc To: Contact: Call-ID: 435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 234d6bd30b71cb4739b386fd3188277d@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:07] DEBUG[13067] acl.c: For destination '64.19.145.15', our source address is '64.19.145.13'. [May 23 13:12:07] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.15 SIP/2.0 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK63cf5e2f [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as6c964664 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:07 GMT [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.15:5060: OPTIONS sip:64.19.145.15 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK63cf5e2f Max-Forwards: 70 From: "unknown" ;tag=as6c964664 To: Contact: Call-ID: 0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042583 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.15:5060 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.15:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK63cf5e2f;received=64.19.145.13 From: "unknown" ;tag=as6c964664 To: ;tag=as77e93415 Call-ID: 0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK63cf5e2f;received=64.19.145.13 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as6c964664 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as77e93415 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060 (Checking To) --From tag as6c964664 --To-tag as77e93415 [May 23 13:12:07] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042583 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Stopping retransmission on '0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:07] DEBUG[13067] chan_sip.c: Destroying SIP dialog 0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '0d5f4d82253e11ae74e795131a29831f@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 6cfcc9f0187bf0ed27ce509904d0cf02@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:07] DEBUG[13067] acl.c: For destination '64.19.145.7', our source address is '64.19.145.13'. [May 23 13:12:07] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 0 [ 31]: OPTIONS sip:64.19.145.7 SIP/2.0 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1bd8b64b [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as1089013a [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 4 [ 21]: To: [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:07 GMT [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: OPTIONS sip:64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1bd8b64b Max-Forwards: 70 From: "unknown" ;tag=as1089013a To: Contact: Call-ID: 2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042586 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1bd8b64b;received=64.19.145.13 From: "unknown" ;tag=as1089013a To: ;tag=as138a0f44 Call-ID: 2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1bd8b64b;received=64.19.145.13 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as1089013a [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 3 [ 36]: To: ;tag=as138a0f44 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: = Looking for Call ID: 2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060 (Checking To) --From tag as1089013a --To-tag as138a0f44 [May 23 13:12:07] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042586 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Stopping retransmission on '2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:07] DEBUG[13067] chan_sip.c: Destroying SIP dialog 2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '2e6bd4831afd4a67081d34111993ea3a@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 2ef8672d4bf4020b6d8a47c727233c13@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:07] DEBUG[13067] acl.c: For destination '64.19.145.12', our source address is '64.19.145.13'. [May 23 13:12:07] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.12 SIP/2.0 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK317acad1 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as4813913d [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:07 GMT [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.12:5060: OPTIONS sip:64.19.145.12 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK317acad1 Max-Forwards: 70 From: "unknown" ;tag=as4813913d To: Contact: Call-ID: 47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042589 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.12:5060 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.12:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK317acad1;received=64.19.145.13 From: "unknown" ;tag=as4813913d To: ;tag=as069da0b8 Call-ID: 47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK317acad1;received=64.19.145.13 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as4813913d [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as069da0b8 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:07] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:07] DEBUG[13067] chan_sip.c: = Looking for Call ID: 47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060 (Checking To) --From tag as4813913d --To-tag as069da0b8 [May 23 13:12:07] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042589 [May 23 13:12:07] DEBUG[13067] chan_sip.c: Stopping retransmission on '47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:07] DEBUG[13067] chan_sip.c: Destroying SIP dialog 47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060 [May 23 13:12:07] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '47c8201c67d51119456730ca01bc59d0@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:08] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 470f7aae18fe2dbd05b1688c7e643b35@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:08] DEBUG[13067] acl.c: For destination '64.19.145.11', our source address is '64.19.145.13'. [May 23 13:12:08] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.11 SIP/2.0 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4eefa2b7 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as25d35c91 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:08 GMT [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:08] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.11:5060: OPTIONS sip:64.19.145.11 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4eefa2b7 Max-Forwards: 70 From: "unknown" ;tag=as25d35c91 To: Contact: Call-ID: 3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:08] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042592 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.11:5060 [May 23 13:12:08] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.11:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4eefa2b7;received=64.19.145.13 From: "unknown" ;tag=as25d35c91 To: ;tag=as14111c2b Call-ID: 3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4eefa2b7;received=64.19.145.13 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as25d35c91 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as14111c2b [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:08] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:08] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:08] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060 (Checking To) --From tag as25d35c91 --To-tag as14111c2b [May 23 13:12:08] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042592 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Stopping retransmission on '3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:08] DEBUG[13067] chan_sip.c: Destroying SIP dialog 3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060 [May 23 13:12:08] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '3ad6c6510227baed0aead8b0147f3ce6@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:08] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042577:OPTIONS (Method 3) (No timer T1) [May 23 13:12:08] VERBOSE[13067] chan_sip.c: Retransmitting #3 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59377900;rport Max-Forwards: 70 From: "unknown" ;tag=as226fa9bc To: Contact: Call-ID: 435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:08] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:08] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '339300464dd2d84555c13cf67b38cf76@209.191.44.130' [May 23 13:12:08] DEBUG[13067] chan_sip.c: Destroying SIP dialog 339300464dd2d84555c13cf67b38cf76@209.191.44.130 [May 23 13:12:08] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '339300464dd2d84555c13cf67b38cf76@209.191.44.130' Method: OPTIONS [May 23 13:12:08] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '4415d19f2e11fea57b3e4eff63b528bc@127.0.0.1' [May 23 13:12:08] DEBUG[13067] chan_sip.c: Destroying SIP dialog 4415d19f2e11fea57b3e4eff63b528bc@127.0.0.1 [May 23 13:12:08] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '4415d19f2e11fea57b3e4eff63b528bc@127.0.0.1' Method: REGISTER [May 23 13:12:09] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042577:OPTIONS (Method 3) (No timer T1) [May 23 13:12:09] VERBOSE[13067] chan_sip.c: Retransmitting #4 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59377900;rport Max-Forwards: 70 From: "unknown" ;tag=as226fa9bc To: Contact: Call-ID: 435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:09] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:09] DEBUG[13067] chan_sip.c: Destroying SIP dialog 435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060 [May 23 13:12:09] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '435aaf4611fe976f295dacfc17b2f3a4@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:10] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.4:5060 ---> OPTIONS sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK63bd0b9f;rport From: "asterisk" ;tag=as5d0a6ecb To: Contact: Call-ID: 1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 23 May 2011 17:12:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <-------------> [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK63bd0b9f;rport [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 2 [ 58]: From: "asterisk" ;tag=as5d0a6ecb [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 4 [ 35]: Contact: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 5 [ 53]: Call-ID: 1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 7 [ 24]: User-Agent: Asterisk PBX [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:10 GMT [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 10 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:10] DEBUG[13067] chan_sip.c: = Looking for Call ID: 1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4 (Checking From) --From tag as5d0a6ecb --To-tag [May 23 13:12:10] DEBUG[13067] acl.c: For destination '64.19.145.4', our source address is '64.19.145.13'. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4 - OPTIONS (No RTP) [May 23 13:12:10] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 13:12:10] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 64.19.145.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK63bd0b9f;rport;received=64.19.145.4 From: "asterisk" ;tag=as5d0a6ecb To: ;tag=as760c3825 Call-ID: 1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> [May 23 13:12:10] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 64.19.145.4:5060 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4' in 32000 ms (Method: OPTIONS) [May 23 13:12:10] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> INVITE sip:7327049020@64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK39094eb7;rport Max-Forwards: 70 From: "Anonymous" ;tag=as6423a45f To: Contact: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 17:12:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 223 v=0 o=root 1302972891 1302972891 IN IP4 64.19.145.7 s=Asterisk PBX SVN-branch-1.6.1-r230383M c=IN IP4 64.19.145.7 t=0 0 m=audio 12008 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 0 [ 42]: INVITE sip:7327049020@64.19.145.13 SIP/2.0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK39094eb7;rport [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as6423a45f [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 4 [ 33]: To: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 5 [ 36]: Contact: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 6 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:10 GMT [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 13 [ 19]: Content-Length: 223 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 14 [ 0]: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=root 1302972891 1302972891 IN IP4 64.19.145.7 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 2 [ 40]: s=Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 3 [ 20]: c=IN IP4 64.19.145.7 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 12008 RTP/AVP 0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 7 [ 25]: a=silenceSupp:off - - - - [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 8 [ 10]: a=ptime:20 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=sendrecv [May 23 13:12:10] VERBOSE[13067] chan_sip.c: --- (14 headers 10 lines) --- [May 23 13:12:10] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking From) --From tag as6423a45f --To-tag [May 23 13:12:10] DEBUG[13067] acl.c: For destination '64.19.145.7', our source address is '64.19.145.13'. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 4eff848341deec190001f2470396b9ea@64.19.145.7 - INVITE (No RTP) [May 23 13:12:10] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 13:12:10] DEBUG[13067] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer" [May 23 13:12:10] DEBUG[13067] sip/reqresp_parser.c: Found SIP option: -replaces- [May 23 13:12:10] DEBUG[13067] sip/reqresp_parser.c: Matched SIP option: replaces [May 23 13:12:10] DEBUG[13067] sip/reqresp_parser.c: Found SIP option: -timer- [May 23 13:12:10] DEBUG[13067] sip/reqresp_parser.c: Matched SIP option: timer [May 23 13:12:10] DEBUG[13067] netsock2.c: Splitting '64.19.145.7:5060' gives... [May 23 13:12:10] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '5060'. [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Sending to 64.19.145.7:5060 (no NAT) [May 23 13:12:10] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Using INVITE request as basis request - 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Found peer 'mg2' for 'Anonymous' from 64.19.145.7:5060 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: <--- Reliably Transmitting (no NAT) to 64.19.145.7:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK39094eb7;received=64.19.145.7;rport=5060 From: "Anonymous" ;tag=as6423a45f To: ;tag=as36c15e4e Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63632ae0" Content-Length: 0 <------------> [May 23 13:12:10] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042597 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' in 6400 ms (Method: INVITE) [May 23 13:12:10] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> ACK sip:7327049020@64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK39094eb7;rport Max-Forwards: 70 From: "Anonymous" ;tag=as6423a45f To: ;tag=as36c15e4e Contact: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Content-Length: 0 <-------------> [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 0 [ 39]: ACK sip:7327049020@64.19.145.13 SIP/2.0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK39094eb7;rport [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as6423a45f [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 4 [ 48]: To: ;tag=as36c15e4e [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 5 [ 36]: Contact: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 6 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: --- (10 headers 0 lines) --- [May 23 13:12:10] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking From) --From tag as6423a45f --To-tag as36c15e4e [May 23 13:12:10] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 13:12:10] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042597 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Stopping retransmission on '4eff848341deec190001f2470396b9ea@64.19.145.7' of Response 102: Match Found [May 23 13:12:10] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> INVITE sip:7327049020@64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK466c24bd;rport Max-Forwards: 70 From: "Anonymous" ;tag=as6423a45f To: Contact: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Authorization: Digest username="mg2", realm="asterisk", algorithm=MD5, uri="sip:7327049020@64.19.145.13", nonce="63632ae0", response="e885b9baf9afa2a85386e0a2b3fba4fe" Date: Mon, 23 May 2011 17:12:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 223 v=0 o=root 1302972891 1302972892 IN IP4 64.19.145.7 s=Asterisk PBX SVN-branch-1.6.1-r230383M c=IN IP4 64.19.145.7 t=0 0 m=audio 12008 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 0 [ 42]: INVITE sip:7327049020@64.19.145.13 SIP/2.0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK466c24bd;rport [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as6423a45f [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 4 [ 33]: To: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 5 [ 36]: Contact: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 6 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 9 [167]: Authorization: Digest username="mg2", realm="asterisk", algorithm=MD5, uri="sip:7327049020@64.19.145.13", nonce="63632ae0", response="e885b9baf9afa2a85386e0a2b3fba4fe" [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 10 [ 35]: Date: Mon, 23 May 2011 17:12:10 GMT [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 11 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 14 [ 19]: Content-Length: 223 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Header 15 [ 0]: [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=root 1302972891 1302972892 IN IP4 64.19.145.7 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 2 [ 40]: s=Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 3 [ 20]: c=IN IP4 64.19.145.7 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 12008 RTP/AVP 0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 7 [ 25]: a=silenceSupp:off - - - - [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 8 [ 10]: a=ptime:20 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=sendrecv [May 23 13:12:10] VERBOSE[13067] chan_sip.c: --- (15 headers 10 lines) --- [May 23 13:12:10] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking From) --From tag as6423a45f --To-tag [May 23 13:12:10] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:10] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:10] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:10] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:10] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 13:12:10] DEBUG[13067] netsock2.c: Splitting '64.19.145.7:5060' gives... [May 23 13:12:10] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '5060'. [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Sending to 64.19.145.7:5060 (no NAT) [May 23 13:12:10] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Using INVITE request as basis request - 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Found peer 'mg2' for 'Anonymous' from 64.19.145.7:5060 [May 23 13:12:10] DEBUG[13067] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb71129c8' [May 23 13:12:10] DEBUG[13067] res_rtp_asterisk.c: Allocated port 12964 for RTP instance '0xb71129c8' [May 23 13:12:10] DEBUG[13067] rtp_engine.c: RTP instance '0xb71129c8' is setup and ready to go [May 23 13:12:10] DEBUG[13067] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb71129c8' [May 23 13:12:10] VERBOSE[13067] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Setting NAT on RTP to Off [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing session-level SDP o=root 1302972891 1302972892 IN IP4 64.19.145.7... UNSUPPORTED. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Asterisk PBX SVN-branch-1.6.1-r230383M... UNSUPPORTED. [May 23 13:12:10] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:10] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 64.19.145.7... OK. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:10] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd39c [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 13:12:10] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 13:12:10] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd39c [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 13:12:10] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb71129c8' [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 64.19.145.7:12008 [May 23 13:12:10] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd39c to 0xb7112b74 [May 23 13:12:10] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 13:12:10] DEBUG[13067] chan_sip.c: Checking SIP call limits for device fsdev-mg2 [May 23 13:12:10] DEBUG[13067] chan_sip.c: Updating call counter for incoming call [May 23 13:12:10] VERBOSE[13067] chan_sip.c: Looking for 7327049020 in from-outside (domain 64.19.145.13) [May 23 13:12:10] DEBUG[13067] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 13:12:10] DEBUG[13067] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 13:12:10] DEBUG[13067] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [May 23 13:12:10] DEBUG[13067] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 13:12:10] DEBUG[13067] chan_sip.c: This channel will not be able to handle video. [May 23 13:12:10] DEBUG[13067] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 13:12:10] DEBUG[13067] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 13:12:10] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/mg2-00000015 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: Anonymous CallerIDName: Anonymous AccountCode: Exten: 7327049020 Context: from-outside Uniqueid: 1306170730.21 [May 23 13:12:10] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: SIPURI Value: sip:Anonymous@64.19.145.7 Uniqueid: 1306170730.21 [May 23 13:12:10] DEBUG[13067] chan_sip.c: build_route: Contact hop: [May 23 13:12:10] VERBOSE[13067] chan_sip.c: list_route: hop: [May 23 13:12:10] DEBUG[13067] chan_sip.c: SIP/mg2-00000015: New call is still down.... Trying... [May 23 13:12:10] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 64.19.145.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK466c24bd;received=64.19.145.7;rport=5060 From: "Anonymous" ;tag=as6423a45f To: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [May 23 13:12:10] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:10] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - mg2 [May 23 13:12:10] DEBUG[13069] chan_sip.c: Checking device state for peer mg2 [May 23 13:12:10] DEBUG[13069] devicestate.c: Changing state for SIP/mg2 - state 1 (Not in use) [May 23 13:12:10] DEBUG[13069] devicestate.c: device 'SIP/mg2' state '1' [May 23 13:12:10] DEBUG[19459] pbx.c: Launching 'Wait' [May 23 13:12:10] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside:1] Wait("SIP/mg2-00000015", "1") in new stack [May 23 13:12:10] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: SIPDOMAIN Value: 64.19.145.13 Uniqueid: 1306170730.21 [May 23 13:12:10] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: SIPCALLID Value: 4eff848341deec190001f2470396b9ea@64.19.145.7 Uniqueid: 1306170730.21 [May 23 13:12:10] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/mg2-00000015 Uniqueid: 1306170730.21 Channeltype: SIP SIPcallid: 4eff848341deec190001f2470396b9ea@64.19.145.7 SIPfullcontact: [May 23 13:12:10] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/mg2-00000015 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: Anonymous CallerIDName: Anonymous Uniqueid: 1306170730.21 [May 23 13:12:10] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside Extension: 7327049020 Priority: 1 Application: Wait AppData: 1 Uniqueid: 1306170730.21 [May 23 13:12:10] DEBUG[13094] app_queue.c: Device 'SIP/mg2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 23 13:12:11] DEBUG[19459] pbx.c: Function result is 'Anonymous' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside:2] Set("SIP/mg2-00000015", "__INCOMINGCLI=Anonymous") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside Extension: 7327049020 Priority: 2 Application: Set AppData: __INCOMINGCLI=Anonymous Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: __INCOMINGCLI Value: Anonymous Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'EXTEN' is '7327049020' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Goto' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside:3] Goto("SIP/mg2-00000015", "from-outside-redir,7327049020,1") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside Extension: 7327049020 Priority: 3 Application: Goto AppData: from-outside-redir,7327049020,1 Uniqueid: 1306170730.21 [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Goto (from-outside-redir,7327049020,1) [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'EXTEN' is '7327049020' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-redir:1] Set("SIP/mg2-00000015", "DIALED_PUBLIC_NUMBER=7327049020") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-redir Extension: 7327049020 Priority: 1 Application: Set AppData: DIALED_PUBLIC_NUMBER=7327049020 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALED_PUBLIC_NUMBER Value: 7327049020 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'EXTEN' is '7327049020' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-redir:2] Set("SIP/mg2-00000015", "DIALED_NUMBER=7327049020") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-redir Extension: 7327049020 Priority: 2 Application: Set AppData: DIALED_NUMBER=7327049020 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALED_NUMBER Value: 7327049020 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Function result is '1' [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DB_RESULT Value: 1 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-redir:3] Set("SIP/mg2-00000015", "status=1") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-redir Extension: 7327049020 Priority: 3 Application: Set AppData: status=1 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: status Value: 1 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'status' is '1' [May 23 13:12:11] DEBUG[19459] pbx.c: Expression result is '1' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'GotoIf' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-redir:4] GotoIf("SIP/mg2-00000015", "1?7") in new stack [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Goto (from-outside-redir,7327049020,7) [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-redir Extension: 7327049020 Priority: 4 Application: GotoIf AppData: 1?7 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'TL_ENABLE_MAXCALLS_CHECK' is '1' [May 23 13:12:11] DEBUG[19459] pbx.c: Expression result is '0' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'GotoIf' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-redir:7] GotoIf("SIP/mg2-00000015", "0?16") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-redir Extension: 7327049020 Priority: 7 Application: GotoIf AppData: 0?16 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Not taking any branch [May 23 13:12:11] DEBUG[19459] pbx.c: Function result is '' [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DB_RESULT Value: Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-redir:8] Set("SIP/mg2-00000015", "MAXCALLS=") in new stack [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'MAXCALLS' is '' [May 23 13:12:11] DEBUG[19459] pbx.c: Expression result is '1' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'GotoIf' [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-redir Extension: 7327049020 Priority: 8 Application: Set AppData: MAXCALLS= Uniqueid: 1306170730.21 [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-redir:9] GotoIf("SIP/mg2-00000015", "1?16") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MAXCALLS Value: Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-redir Extension: 7327049020 Priority: 9 Application: GotoIf AppData: 1?16 Uniqueid: 1306170730.21 [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Goto (from-outside-redir,7327049020,16) [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'EXTEN' is '7327049020' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'GotoIfTime' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-redir:16] GotoIfTime("SIP/mg2-00000015", "*,*,*,*?from-outside-7327049020-tl-allhours-eng,7327049020,1") in new stack [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Goto (from-outside-7327049020-tl-allhours-eng,7327049020,1) [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-redir Extension: 7327049020 Priority: 16 Application: GotoIfTime AppData: *,*,*,*?from-outside-7327049020-tl-allhours-eng,7327049020,1 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:1] Set("SIP/mg2-00000015", "__tenant=eng") in new stack [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-7327049020-tl-allhours-eng Extension: 7327049020 Priority: 1 Application: Set AppData: __tenant=eng Uniqueid: 1306170730.21 [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:2] Set("SIP/mg2-00000015", "CDR(userfield)=eng") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: __tenant Value: eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-7327049020-tl-allhours-eng Extension: 7327049020 Priority: 2 Application: Set AppData: CDR(userfield)=eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:3] Set("SIP/mg2-00000015", "CDR(accountcode)=eng") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-7327049020-tl-allhours-eng Extension: 7327049020 Priority: 3 Application: Set AppData: CDR(accountcode)=eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/mg2-00000015 Uniqueid: 1306170730.21 AccountCode: eng OldAccountCode: [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'TL_DASH' is '-' [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:11] DEBUG[19459] pbx.c: Function result is 'default-eng' [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DB_RESULT Value: default-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:4] Set("SIP/mg2-00000015", "MOH=default-eng") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-7327049020-tl-allhours-eng Extension: 7327049020 Priority: 4 Application: Set AppData: MOH=default-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MOH Value: default-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'MOH' is 'default-eng' [May 23 13:12:11] DEBUG[19459] pbx.c: Expression result is '0' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'GotoIf' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:5] GotoIf("SIP/mg2-00000015", "0?nomoh") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-7327049020-tl-allhours-eng Extension: 7327049020 Priority: 5 Application: GotoIf AppData: 0?nomoh Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Not taking any branch [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'MOH' is 'default-eng' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:6] Set("SIP/mg2-00000015", "CHANNEL(musicclass)=default-eng") in new stack [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Macro' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:7] Macro("SIP/mg2-00000015", "tl-huntlist,engtest-eng") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-7327049020-tl-allhours-eng Extension: 7327049020 Priority: 6 Application: Set AppData: CHANNEL(musicclass)=default-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-outside-7327049020-tl-allhours-eng Extension: 7327049020 Priority: 7 Application: Macro AppData: tl-huntlist,engtest-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_EXTEN Value: 7327049020 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_CONTEXT Value: from-outside-7327049020-tl-allhours-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_PRIORITY Value: 7 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ARG1 Value: engtest-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'ARG2' is NULL [May 23 13:12:11] DEBUG[19459] pbx.c: Function result is 'Anonymous' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Set' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [s@macro-tl-huntlist:1] Set("SIP/mg2-00000015", "CALLERID(name)=Anonymous") in new stack [May 23 13:12:11] DEBUG[19459] app_macro.c: Executed application: Set [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: macro-tl-huntlist Extension: s Priority: 1 Application: Set AppData: CALLERID(name)=Anonymous Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/mg2-00000015 CallerIDNum: Anonymous CallerIDName: Anonymous Uniqueid: 1306170730.21 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Result of 'ARG1' is 'engtest-eng' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Goto' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [s@macro-tl-huntlist:2] Goto("SIP/mg2-00000015", "engtest-eng,s,1") in new stack [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Goto (engtest-eng,s,1) [May 23 13:12:11] DEBUG[19459] app_macro.c: Executed application: Goto [May 23 13:12:11] VERBOSE[19459] app_macro.c: == Channel 'SIP/mg2-00000015' jumping out of macro 'tl-huntlist' [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'NoOp' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [s@engtest-eng:1] NoOp("SIP/mg2-00000015", "engtest-eng") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: macro-tl-huntlist Extension: s Priority: 2 Application: Goto AppData: engtest-eng,s,1 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_DEPTH Value: 0 Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: engtest-eng Extension: s Priority: 1 Application: NoOp AppData: engtest-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] pbx.c: Launching 'Dial' [May 23 13:12:11] VERBOSE[19459] pbx.c: -- Executing [s@engtest-eng:2] Dial("SIP/mg2-00000015", "SIP/322-eng&SIP/312-eng") in new stack [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: engtest-eng Extension: s Priority: 2 Application: Dial AppData: SIP/322-eng&SIP/312-eng Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALSTATUS Value: Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALEDPEERNUMBER Value: Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALEDPEERNAME Value: Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ANSWEREDTIME Value: Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALEDTIME Value: Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: Allocating new SIP dialog for 4e3070027fb6194e289409d2461804d5@127.0.0.1:0 - INVITE (No RTP) [May 23 13:12:11] DEBUG[19459] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6646b68' [May 23 13:12:11] DEBUG[19459] res_rtp_asterisk.c: Allocated port 14698 for RTP instance '0xb6646b68' [May 23 13:12:11] DEBUG[19459] rtp_engine.c: RTP instance '0xb6646b68' is setup and ready to go [May 23 13:12:11] DEBUG[19459] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6646b68' [May 23 13:12:11] VERBOSE[19459] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Setting NAT on RTP to Off [May 23 13:12:11] DEBUG[19459] chan_sip.c: OBPROXY: Not applying OBproxy to this call [May 23 13:12:11] DEBUG[19459] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 13:12:11] DEBUG[19459] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** Our capabilities are 0x404 (ulaw|ilbc) [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/322-eng-00000016 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 322 CallerIDName: Poly_test ENG AccountCode: eng Exten: Context: from-inside-eng Uniqueid: 1306170731.22 [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: This channel will not be able to handle video. [May 23 13:12:11] DEBUG[19459] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 13:12:11] DEBUG[19459] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: SIPCALLID Value: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 Uniqueid: 1306170731.22 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/322-eng-00000016 Uniqueid: 1306170731.22 Channeltype: SIP SIPcallid: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 SIPfullcontact: sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/322-eng-00000016 Channeltype: SIP SIPcallid: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 SIPfullcontact: sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 Peername: 322-eng [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: DIALEDPEERNUMBER Value: 322-eng Uniqueid: 1306170731.22 [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALEDTIME. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable ANSWEREDTIME. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALEDPEERNAME. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALEDPEERNUMBER. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALSTATUS. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable MACRO_DEPTH. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable MOH. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DB_RESULT. [May 23 13:12:11] DEBUG[19459] channel.c: Copying hard-transferable variable tenant. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable MAXCALLS. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable status. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALED_NUMBER. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALED_PUBLIC_NUMBER. [May 23 13:12:11] DEBUG[19459] channel.c: Copying hard-transferable variable INCOMINGCLI. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable SIPCALLID. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable SIPDOMAIN. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable SIPURI. [May 23 13:12:11] DEBUG[19459] chan_sip.c: Outgoing Call for 322-eng [May 23 13:12:11] DEBUG[19459] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:11] DEBUG[19459] chan_sip.c: Call to peer '322-eng' is 1 out of 2147483647 [May 23 13:12:11] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:11] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:11] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 6 (Ringing) [May 23 13:12:11] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '6' [May 23 13:12:11] DEBUG[19459] chan_sip.c: ** Our capability: 0x404 (ulaw|ilbc) Video flag: False Text flag: False [May 23 13:12:11] DEBUG[19459] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 13:12:11] VERBOSE[19459] chan_sip.c: Audio is at 5060 [May 23 13:12:11] VERBOSE[19459] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:11] VERBOSE[19459] chan_sip.c: Adding codec 0x400 (ilbc) to SDP [May 23 13:12:11] DEBUG[19459] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:11] DEBUG[19459] chan_sip.c: Done building SDP. Settling with this capability: 0x404 (ulaw|ilbc) [May 23 13:12:11] DEBUG[19459] chan_sip.c: Initializing initreq for method INVITE - callid 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 0 [ 72]: INVITE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5bb2fe1c [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 3 [ 61]: From: "Anonymous" ;tag=as552f30c6 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 4 [ 63]: To: [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 5 [ 42]: Contact: [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 6 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:11 GMT [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 13:12:11] VERBOSE[19459] chan_sip.c: Reliably Transmitting (no NAT) to 209.191.39.117:5060: INVITE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5bb2fe1c Max-Forwards: 70 From: "Anonymous" ;tag=as552f30c6 To: Contact: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 236 v=0 o=root 41473503 41473503 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.13 t=0 0 m=audio 14698 RTP/AVP 0 97 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=ptime:20 a=sendrecv --- [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042600 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:11] VERBOSE[19459] app_dial.c: -- Called SIP/322-eng [May 23 13:12:11] DEBUG[19459] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: Allocating new SIP dialog for 0aaa540b2101d6a875130bcf0a4a5ba2@127.0.0.1:0 - INVITE (No RTP) [May 23 13:12:11] DEBUG[19459] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa0c8ee8' [May 23 13:12:11] DEBUG[19459] res_rtp_asterisk.c: Allocated port 10138 for RTP instance '0xa0c8ee8' [May 23 13:12:11] DEBUG[19459] rtp_engine.c: RTP instance '0xa0c8ee8' is setup and ready to go [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/mg2-00000015 Destination: SIP/322-eng-00000016 CallerIDNum: Anonymous CallerIDName: Anonymous UniqueID: 1306170730.21 DestUniqueID: 1306170731.22 Dialstring: 322-eng [May 23 13:12:11] DEBUG[19459] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa0c8ee8' [May 23 13:12:11] VERBOSE[19459] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Setting NAT on RTP to On [May 23 13:12:11] DEBUG[19459] chan_sip.c: OBPROXY: Not applying OBproxy to this call [May 23 13:12:11] DEBUG[19459] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 13:12:11] DEBUG[19459] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/312-eng-00000017 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 312 CallerIDName: SPA303 Cisco AccountCode: eng Exten: Context: from-inside-eng Uniqueid: 1306170731.23 [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: This channel will not be able to handle video. [May 23 13:12:11] DEBUG[19459] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 13:12:11] DEBUG[19459] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000017 Variable: SIPCALLID Value: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 Uniqueid: 1306170731.23 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-00000017 Uniqueid: 1306170731.23 Channeltype: SIP SIPcallid: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-00000017 Channeltype: SIP SIPcallid: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 Peername: 312-eng [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000017 Variable: DIALEDPEERNUMBER Value: 312-eng Uniqueid: 1306170731.23 [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALEDTIME. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable ANSWEREDTIME. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALEDPEERNAME. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALEDPEERNUMBER. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALSTATUS. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable MACRO_DEPTH. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable MOH. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DB_RESULT. [May 23 13:12:11] DEBUG[19459] channel.c: Copying hard-transferable variable tenant. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable MAXCALLS. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable status. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALED_NUMBER. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable DIALED_PUBLIC_NUMBER. [May 23 13:12:11] DEBUG[19459] channel.c: Copying hard-transferable variable INCOMINGCLI. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable SIPCALLID. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable SIPDOMAIN. [May 23 13:12:11] DEBUG[19459] channel.c: Not copying variable SIPURI. [May 23 13:12:11] DEBUG[19459] chan_sip.c: Outgoing Call for 312-eng [May 23 13:12:11] DEBUG[19459] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:11] DEBUG[19459] chan_sip.c: Call to peer '312-eng' is 1 out of 2147483647 [May 23 13:12:11] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:11] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:11] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 6 (Ringing) [May 23 13:12:11] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '6' [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 6 Paused: 0 [May 23 13:12:11] DEBUG[19459] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [May 23 13:12:11] DEBUG[19459] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 13:12:11] VERBOSE[19459] chan_sip.c: Audio is at 5060 [May 23 13:12:11] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '6' (Ringing) [May 23 13:12:11] VERBOSE[19459] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:11] DEBUG[19459] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 6 Paused: 0 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 13:12:11] DEBUG[19459] chan_sip.c: Initializing initreq for method INVITE - callid 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 0 [ 72]: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK0f6a3596;rport [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:11] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '6' (Ringing) [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 3 [ 61]: From: "Anonymous" ;tag=as191e5801 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 4 [ 63]: To: [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 5 [ 42]: Contact: [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 6 [ 59]: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:11 GMT [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 12 [ 93]: Remote-Party-ID: "Anonymous" ;party=calling;privacy=off;screen=no [May 23 13:12:11] DEBUG[19459] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [May 23 13:12:11] VERBOSE[19459] chan_sip.c: Reliably Transmitting (NAT) to 209.191.39.117:5060: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK0f6a3596;rport Max-Forwards: 70 From: "Anonymous" ;tag=as191e5801 To: Contact: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "Anonymous" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 193 v=0 o=root 907077690 907077690 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.13 t=0 0 m=audio 10138 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 13:12:11] DEBUG[19459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042602 [May 23 13:12:11] DEBUG[19459] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/mg2-00000015 Destination: SIP/312-eng-00000017 CallerIDNum: Anonymous CallerIDName: Anonymous UniqueID: 1306170730.21 DestUniqueID: 1306170731.23 Dialstring: 312-eng [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 322 Context: local-extensions-eng Hint: SIP/322-eng Status: 8 [May 23 13:12:11] VERBOSE[19459] app_dial.c: -- Called SIP/312-eng [May 23 13:12:11] VERBOSE[19459] app_dial.c: -- SIP/322-eng-00000016 connected line has changed. Saving it until answer for SIP/mg2-00000015 [May 23 13:12:11] VERBOSE[19459] app_dial.c: -- SIP/312-eng-00000017 connected line has changed. Saving it until answer for SIP/mg2-00000015 [May 23 13:12:11] DEBUG[13070] app_queue.c: Extension '322@local-extensions-eng' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 312 Context: local-extensions-eng Hint: SIP/312-eng Status: 8 [May 23 13:12:11] DEBUG[13070] app_queue.c: Extension '312@local-extensions-eng' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [May 23 13:12:11] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 100 Trying From: "Anonymous";tag=as552f30c6 To: "Poly_test ENG";tag=C5E6782D-50C4FB20 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5bb2fe1c Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Length: 0 <-------------> [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as552f30c6 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5bb2fe1c [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [May 23 13:12:11] VERBOSE[13067] chan_sip.c: --- (10 headers 0 lines) --- [May 23 13:12:11] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking To) --From tag as552f30c6 --To-tag C5E6782D-50C4FB20 [May 23 13:12:11] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1042600 - INVITE (got response) [May 23 13:12:11] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Request 102: Found [May 23 13:12:11] DEBUG[13067] chan_sip.c: SIP response 100 to standard invite [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000016~ Value: SIP 100 Trying Uniqueid: 1306170730.21 [May 23 13:12:11] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 100 Trying From: "Anonymous";tag=as191e5801 To: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0f6a3596 Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as191e5801 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 2 [ 63]: To: [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0f6a3596 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 6 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [May 23 13:12:11] VERBOSE[13067] chan_sip.c: --- (8 headers 0 lines) --- [May 23 13:12:11] DEBUG[13067] chan_sip.c: = Looking for Call ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 (Checking To) --From tag as191e5801 --To-tag [May 23 13:12:11] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1042602 - INVITE (got response) [May 23 13:12:11] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060' Request 102: Found [May 23 13:12:11] DEBUG[13067] chan_sip.c: SIP response 100 to standard invite [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-00000017~ Value: SIP 100 Trying Uniqueid: 1306170730.21 [May 23 13:12:11] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 180 Ringing From: "Anonymous";tag=as191e5801 To: ;tag=97bdf59e2d1676fi0 Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0f6a3596 Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as191e5801 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 2 [ 85]: To: ;tag=97bdf59e2d1676fi0 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0f6a3596 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 6 [ 82]: Contact: "SPA303 Cisco" [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:11] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [May 23 13:12:11] VERBOSE[13067] chan_sip.c: --- (9 headers 0 lines) --- [May 23 13:12:11] DEBUG[13067] chan_sip.c: = Looking for Call ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 (Checking To) --From tag as191e5801 --To-tag 97bdf59e2d1676fi0 [May 23 13:12:11] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060' Request 102: Found [May 23 13:12:11] DEBUG[13067] chan_sip.c: SIP response 180 to standard invite [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/312-eng-00000017 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 312 CallerIDName: SPA303 Cisco Uniqueid: 1306170731.23 [May 23 13:12:11] VERBOSE[19459] app_dial.c: -- SIP/312-eng-00000017 is ringing [May 23 13:12:11] VERBOSE[19459] chan_sip.c: <--- Transmitting (no NAT) to 64.19.145.7:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK466c24bd;received=64.19.145.7;rport=5060 From: "Anonymous" ;tag=as6423a45f To: ;tag=as60db63bb Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [May 23 13:12:11] DEBUG[19459] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:11] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:11] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:11] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 6 (Ringing) [May 23 13:12:11] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '6' [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-00000017~ Value: SIP 180 Ringing Uniqueid: 1306170730.21 [May 23 13:12:11] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 6 Paused: 0 [May 23 13:12:11] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '6' (Ringing) [May 23 13:12:12] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 180 Ringing From: "Anonymous";tag=as552f30c6 To: "Poly_test ENG";tag=C5E6782D-50C4FB20 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5bb2fe1c Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Allow-Events: talk,hold,conference Content-Length: 0 <-------------> [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as552f30c6 [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5bb2fe1c [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 9 [ 34]: Allow-Events: talk,hold,conference [May 23 13:12:12] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:12] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:12] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking To) --From tag as552f30c6 --To-tag C5E6782D-50C4FB20 [May 23 13:12:12] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Request 102: Found [May 23 13:12:12] DEBUG[13067] chan_sip.c: SIP response 180 to standard invite [May 23 13:12:12] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/322-eng-00000016 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 322 CallerIDName: Poly_test ENG Uniqueid: 1306170731.22 [May 23 13:12:12] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:12] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000016~ Value: SIP 180 Ringing Uniqueid: 1306170730.21 [May 23 13:12:12] VERBOSE[19459] app_dial.c: -- SIP/322-eng-00000016 is ringing [May 23 13:12:12] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:12] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 6 (Ringing) [May 23 13:12:12] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '6' [May 23 13:12:12] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 6 Paused: 0 [May 23 13:12:12] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '6' (Ringing) [May 23 13:12:13] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 348d5f2e672008673ce4f6922134b43c@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:13] DEBUG[13067] acl.c: For destination '64.19.145.20', our source address is '64.19.145.13'. [May 23 13:12:13] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:13] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060 [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.20 SIP/2.0 [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4941072c [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as2e08a8f2 [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060 [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:13 GMT [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:13] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:13] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4941072c Max-Forwards: 70 From: "unknown" ;tag=as2e08a8f2 To: Contact: Call-ID: 6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:13] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042605 [May 23 13:12:13] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Anonymous";tag=as552f30c6 To: "Poly_test ENG";tag=C5E6782D-50C4FB20 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5bb2fe1c Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Supported: 100rel Supported: replaces User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Type: application/SDP Content-Length: 165 v=0 o=- 1306170710 1306170710 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 m=audio 51818 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as552f30c6 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5bb2fe1c [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Supported: 100rel [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Supported: replaces [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 10 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Accept-Language: en [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 12 [ 29]: Content-Type: application/SDP [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 13 [ 19]: Content-Length: 165 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 14 [ 0]: [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306170710 1306170710 IN IP4 209.191.39.117 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 51818 RTP/AVP 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 6 [ 10]: a=sendrecv [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: --- (14 headers 8 lines) --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking To) --From tag as552f30c6 --To-tag C5E6782D-50C4FB20 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Acked pending invite 102 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Stopping retransmission on '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:14] DEBUG[13067] chan_sip.c: SIP response 200 to standard invite [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 1306170710 1306170710 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 13:12:14] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6646b68' [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51818 [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xb6646d14 [May 23 13:12:14] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 13:12:14] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 13:12:14] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:14] DEBUG[13067] chan_sip.c: build_route: Contact hop: [May 23 13:12:14] VERBOSE[13067] chan_sip.c: list_route: hop: [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Strict routing enforced for session 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/322-eng-00000016 Channeltype: SIP Uniqueid: 1306170731.22 SIPcallid: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 SIPfullcontact: sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 Peername: 322-eng [May 23 13:12:14] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Transmitting (no NAT) to 209.191.39.117:5060: ACK sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK28c74b88 Max-Forwards: 70 From: "Anonymous" ;tag=as552f30c6 To: ;tag=C5E6782D-50C4FB20 Contact: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:322' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000016~ Value: SIP 200 OK Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:14] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:14] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 2 (In use) [May 23 13:12:14] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '2' [May 23 13:12:14] VERBOSE[19459] app_dial.c: -- SIP/322-eng-00000016 connected line has changed. Saving it until answer for SIP/mg2-00000015 [May 23 13:12:14] VERBOSE[19459] app_dial.c: -- SIP/322-eng-00000016 answered SIP/mg2-00000015 [May 23 13:12:14] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:14] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:14] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 2 (In use) [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/322-eng-00000016 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 322 CallerIDName: Poly_test ENG Uniqueid: 1306170731.22 [May 23 13:12:14] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '2' [May 23 13:12:14] DEBUG[19459] rtp_engine.c: Setting early bridge SDP of 'SIP/mg2-00000015' with that of 'SIP/322-eng-00000016' [May 23 13:12:14] DEBUG[19459] channel.c: Hanging up channel 'SIP/312-eng-00000017' [May 23 13:12:14] DEBUG[19459] chan_sip.c: This call was answered elsewhere[May 23 13:12:14] DEBUG[19459] chan_sip.c: ####### It's the cause code, buddy. The cause code!!! [May 23 13:12:14] DEBUG[19459] chan_sip.c: Hangup call SIP/312-eng-00000017, SIP callid 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:14] DEBUG[19459] chan_sip.c: update_call_counter(312-eng) - decrement call limit counter on hangup [May 23 13:12:14] DEBUG[19459] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:14] DEBUG[19459] chan_sip.c: Call to peer '312-eng' removed from call limit 2147483647 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Hanging up channel in state Ringing (not UP) [May 23 13:12:14] DEBUG[19459] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa0c8ee8' [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Scheduling destruction of SIP dialog '307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060' in 32000 ms (Method: INVITE) [May 23 13:12:14] DEBUG[19459] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060' Request 102: Found [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Reliably Transmitting (NAT) to 209.191.39.117:5060: CANCEL sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK0f6a3596;rport Max-Forwards: 70 From: "Anonymous" ;tag=as191e5801 To: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 --- [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[19459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042608 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Scheduling destruction of SIP dialog '307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060' in 32000 ms (Method: INVITE) [May 23 13:12:14] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:14] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:14] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 1 (Not in use) [May 23 13:12:14] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '1' [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/312-eng-00000017 Uniqueid: 1306170731.23 CallerIDNum: 312 CallerIDName: SPA303 Cisco Cause: 26 Cause-txt: Unknown [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALEDPEERNAME Value: SIP/322-eng-00000016 Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:14] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:14] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 1 (Not in use) [May 23 13:12:14] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '1' [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALEDPEERNUMBER Value: 322-eng Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: BRIDGEPEER Value: SIP/322-eng-00000016 Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '2' (In use) [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 2 Paused: 0 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: BRIDGEPEER Value: SIP/mg2-00000015 Uniqueid: 1306170731.22 [May 23 13:12:14] DEBUG[19459] chan_sip.c: SIP answering channel: SIP/mg2-00000015 [May 23 13:12:14] DEBUG[19459] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:14] DEBUG[19459] chan_sip.c: Setting framing from config on incoming call [May 23 13:12:14] DEBUG[19459] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 13:12:14] DEBUG[19459] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Audio is at 5060 [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:14] DEBUG[19459] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:14] DEBUG[19459] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 13:12:14] VERBOSE[19459] chan_sip.c: <--- Reliably Transmitting (no NAT) to 64.19.145.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK466c24bd;received=64.19.145.7;rport=5060 From: "Anonymous" ;tag=as6423a45f To: ;tag=as60db63bb Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 193 v=0 o=root 272259394 272259394 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.13 t=0 0 m=audio 12964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv <------------> [May 23 13:12:14] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - mg2 [May 23 13:12:14] DEBUG[13069] chan_sip.c: Checking device state for peer mg2 [May 23 13:12:14] DEBUG[13069] devicestate.c: Changing state for SIP/mg2 - state 1 (Not in use) [May 23 13:12:14] DEBUG[13069] devicestate.c: device 'SIP/mg2' state '1' [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/mg2-00000015 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: Anonymous CallerIDName: Anonymous Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[19459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042610 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:14] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '2' (In use) [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 2 Paused: 0 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> ACK sip:7327049020@64.19.145.13:5060 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK12b12ea5;rport Max-Forwards: 70 From: "Anonymous" ;tag=as6423a45f To: ;tag=as60db63bb Contact: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Content-Length: 0 <-------------> [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 0 [ 44]: ACK sip:7327049020@64.19.145.13:5060 SIP/2.0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK12b12ea5;rport [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as6423a45f [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 4 [ 48]: To: ;tag=as60db63bb [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 5 [ 36]: Contact: [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 6 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: --- (10 headers 0 lines) --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking From) --From tag as6423a45f --To-tag as60db63bb [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/322-eng-00000016 Uniqueid: 1306170731.22 AccountCode: eng OldAccountCode: eng [May 23 13:12:14] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 13:12:14] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042610 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Stopping retransmission on '4eff848341deec190001f2470396b9ea@64.19.145.7' of Response 103: Match Found [May 23 13:12:14] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '1' (Not in use) [May 23 13:12:14] DEBUG[19459] features.c: bridge answer set, chan answer set [May 23 13:12:14] DEBUG[19459] features.c: Removing dialed interfaces datastore on SIP/322-eng-00000016 since we're bridging [May 23 13:12:14] DEBUG[19459] channel.c: setting peeraccount to eng for SIP/mg2-00000015 from data on channel SIP/322-eng-00000016 [May 23 13:12:14] DEBUG[19459] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:14] DEBUG[19459] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/mg2-00000015 Channel2: SIP/322-eng-00000016 Uniqueid1: 1306170730.21 Uniqueid2: 1306170731.22 CallerID1: Anonymous CallerID2: 322 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 322 Context: local-extensions-eng Hint: SIP/322-eng Status: 1 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: BRIDGEPEER Value: SIP/322-eng-00000016 Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: BRIDGEPVTCALLID Value: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: BRIDGEPEER Value: SIP/mg2-00000015 Uniqueid: 1306170731.22 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: BRIDGEPVTCALLID Value: 4eff848341deec190001f2470396b9ea@64.19.145.7 Uniqueid: 1306170731.22 [May 23 13:12:14] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '1' (Not in use) [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:14] VERBOSE[19459] rtp_engine.c: -- Remotely bridging SIP/mg2-00000015 and SIP/322-eng-00000016 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Sending reinvite on SIP '4eff848341deec190001f2470396b9ea@64.19.145.7' - It's audio soon redirected to IP 209.191.39.117:51818 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Strict routing enforced for session 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:14] VERBOSE[19459] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:14] DEBUG[19459] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:14] DEBUG[19459] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:14] VERBOSE[19459] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 13:12:14] DEBUG[19459] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 13:12:14] DEBUG[19459] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 13:12:14] DEBUG[19459] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Audio is at 5060 [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:14] DEBUG[19459] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:14] DEBUG[19459] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 13:12:14] DEBUG[19459] chan_sip.c: Initializing already initialized SIP dialog 4eff848341deec190001f2470396b9ea@64.19.145.7 (presumably reinvite) [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 0 [ 40]: INVITE sip:Anonymous@64.19.145.7 SIP/2.0 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK04156fd7;rport [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 3 [ 50]: From: ;tag=as60db63bb [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 4 [ 64]: To: "Anonymous" ;tag=as6423a45f [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 5 [ 43]: Contact: [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 6 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 10 [ 19]: Supported: replaces [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: INVITE sip:Anonymous@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK04156fd7;rport Max-Forwards: 70 From: ;tag=as60db63bb To: "Anonymous" ;tag=as6423a45f Contact: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 197 v=0 o=root 272259394 272259395 IN IP4 209.191.39.117 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 209.191.39.117 t=0 0 m=audio 51818 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 13:12:14] DEBUG[19459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042611 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Sending reinvite on SIP '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' - It's audio soon redirected to IP 64.19.145.7:12008 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Strict routing enforced for session 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:14] VERBOSE[19459] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:14] DEBUG[19459] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:14] DEBUG[19459] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:14] VERBOSE[19459] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:14] DEBUG[19459] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 13:12:14] DEBUG[19459] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 13:12:14] DEBUG[19459] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Audio is at 5060 [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:14] DEBUG[19459] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:14] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK04156fd7;received=64.19.145.13;rport=5060 From: ;tag=as60db63bb To: "Anonymous" ;tag=as6423a45f Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <-------------> [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 13:12:14] DEBUG[19459] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK04156fd7;received=64.19.145.13;rport=5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as60db63bb [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 3 [ 64]: To: "Anonymous" ;tag=as6423a45f [May 23 13:12:14] DEBUG[19459] chan_sip.c: Initializing already initialized SIP dialog 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (presumably reinvite) [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 0 [ 72]: INVITE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59e42ce8 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [May 23 13:12:14] DEBUG[13070] app_queue.c: Extension '322@local-extensions-eng' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 9 [ 36]: Contact: [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking To) --From tag as60db63bb --To-tag as6423a45f [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 3 [ 61]: From: "Anonymous" ;tag=as552f30c6 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 4 [ 85]: To: ;tag=C5E6782D-50C4FB20 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 5 [ 42]: Contact: [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 6 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 10 [ 19]: Supported: replaces [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 23 13:12:14] DEBUG[19459] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 13:12:14] VERBOSE[19459] chan_sip.c: Reliably Transmitting (no NAT) to 209.191.39.117:5060: INVITE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59e42ce8 Max-Forwards: 70 From: "Anonymous" ;tag=as552f30c6 To: ;tag=C5E6782D-50C4FB20 Contact: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 189 v=0 o=root 41473503 41473504 IN IP4 64.19.145.7 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.7 t=0 0 m=audio 12008 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 13:12:14] DEBUG[19459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042612 [May 23 13:12:14] DEBUG[19459] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1042611 - INVITE (got response) [May 23 13:12:14] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4eff848341deec190001f2470396b9ea@64.19.145.7' Request 102: Found [May 23 13:12:14] DEBUG[13067] chan_sip.c: SIP response 100 to RE-invite on outgoing call 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:14] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK04156fd7;received=64.19.145.13;rport=5060 From: ;tag=as60db63bb To: "Anonymous" ;tag=as6423a45f Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 223 v=0 o=root 1302972891 1302972893 IN IP4 64.19.145.7 s=Asterisk PBX SVN-branch-1.6.1-r230383M c=IN IP4 64.19.145.7 t=0 0 m=audio 12008 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK04156fd7;received=64.19.145.13;rport=5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as60db63bb [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 3 [ 64]: To: "Anonymous" ;tag=as6423a45f [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 9 [ 36]: Contact: [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Content-Length: 223 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 12 [ 0]: [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=root 1302972891 1302972893 IN IP4 64.19.145.7 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 2 [ 40]: s=Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 3 [ 20]: c=IN IP4 64.19.145.7 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 12008 RTP/AVP 0 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/mg2-00000015~ Value: SIP 100 Trying Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 7 [ 25]: a=silenceSupp:off - - - - [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 8 [ 10]: a=ptime:20 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=sendrecv [May 23 13:12:14] VERBOSE[13067] chan_sip.c: --- (12 headers 10 lines) --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking To) --From tag as60db63bb --To-tag as6423a45f [May 23 13:12:14] DEBUG[13067] chan_sip.c: Acked pending invite 102 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Stopping retransmission on '4eff848341deec190001f2470396b9ea@64.19.145.7' of Request 102: Match Found [May 23 13:12:14] DEBUG[13067] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP o=root 1302972891 1302972893 IN IP4 64.19.145.7... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Asterisk PBX SVN-branch-1.6.1-r230383M... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 64.19.145.7... OK. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 13:12:14] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb71129c8' [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 64.19.145.7:12008 [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xb7112b74 [May 23 13:12:14] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 13:12:14] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 13:12:14] DEBUG[13067] chan_sip.c: Updating call counter for incoming call [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Strict routing enforced for session 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Transmitting (no NAT) to 64.19.145.7:5060: ACK sip:Anonymous@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK6fbb13c3;rport Max-Forwards: 70 From: ;tag=as60db63bb To: "Anonymous" ;tag=as6423a45f Contact: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:Ano' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/mg2-00000015~ Value: SIP 200 OK Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13094] app_queue.c: Device 'SIP/mg2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 312 Context: local-extensions-eng Hint: SIP/312-eng Status: 0 [May 23 13:12:14] DEBUG[13070] app_queue.c: Extension '312@local-extensions-eng' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 487 Request Terminated From: "Anonymous";tag=as191e5801 To: ;tag=97bdf59e2d1676fi0 Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0f6a3596 Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 0 [ 30]: SIP/2.0 487 Request Terminated [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as191e5801 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 2 [ 85]: To: ;tag=97bdf59e2d1676fi0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0f6a3596 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 6 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: --- (8 headers 0 lines) --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: = Looking for Call ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 (Checking To) --From tag as191e5801 --To-tag 97bdf59e2d1676fi0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Acked pending invite 102 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Stopping retransmission on '307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:14] DEBUG[13067] chan_sip.c: SIP response 487 to standard invite [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Transmitting (NAT) to 209.191.39.117:5060: ACK sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK0f6a3596;rport Max-Forwards: 70 From: "Anonymous" ;tag=as191e5801 To: ;tag=97bdf59e2d1676fi0 Contact: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:312' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:14] DEBUG[13067] chan_sip.c: Call to peer '312-eng' removed from call limit 2147483647 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Setting SIP_ALREADYGONE on dialog 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:14] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:14] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:14] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 1 (Not in use) [May 23 13:12:14] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '1' [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:14] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '1' (Not in use) [May 23 13:12:14] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Anonymous";tag=as191e5801 To: ;tag=97bdf59e2d1676fi0 Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 CSeq: 102 CANCEL Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0f6a3596 Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as191e5801 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 2 [ 85]: To: ;tag=97bdf59e2d1676fi0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 CANCEL [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0f6a3596 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 6 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: --- (8 headers 0 lines) --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: = Looking for Call ID: 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 (Checking To) --From tag as191e5801 --To-tag 97bdf59e2d1676fi0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042608 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Stopping retransmission on '307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Destroying SIP dialog 307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '307c70186d836d1827e8e9c5427a75a0@64.19.145.13:5060' Method: INVITE [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Destroyed RTP instance '0xa0c8ee8' [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:14] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Anonymous";tag=as552f30c6 To: "Poly_test ENG";tag=C5E6782D-50C4FB20 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59e42ce8 Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Supported: 100rel Supported: replaces User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Type: application/SDP Content-Length: 165 v=0 o=- 1306170710 1306170711 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 m=audio 51818 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as552f30c6 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK59e42ce8 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Supported: 100rel [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Supported: replaces [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 10 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Accept-Language: en [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 12 [ 29]: Content-Type: application/SDP [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 13 [ 19]: Content-Length: 165 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Header 14 [ 0]: [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306170710 1306170711 IN IP4 209.191.39.117 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 51818 RTP/AVP 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 6 [ 10]: a=sendrecv [May 23 13:12:14] DEBUG[13067] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: --- (14 headers 8 lines) --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking To) --From tag as552f30c6 --To-tag C5E6782D-50C4FB20 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Acked pending invite 103 [May 23 13:12:14] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042612 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Stopping retransmission on '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' of Request 103: Match Found [May 23 13:12:14] DEBUG[13067] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 1306170710 1306170711 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 13:12:14] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6646b68' [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51818 [May 23 13:12:14] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xb6646d14 [May 23 13:12:14] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 13:12:14] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 13:12:14] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:14] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:14] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:14] DEBUG[13067] chan_sip.c: Strict routing enforced for session 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:14] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 2 (In use) [May 23 13:12:14] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '2' [May 23 13:12:14] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:14] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:14] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:14] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Transmitting (no NAT) to 209.191.39.117:5060: ACK sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK6071dbdb Max-Forwards: 70 From: "Anonymous" ;tag=as552f30c6 To: ;tag=C5E6782D-50C4FB20 Contact: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:322' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000016~ Value: SIP 200 OK Uniqueid: 1306170730.21 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:14] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 2 Paused: 0 [May 23 13:12:14] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '2' (In use) [May 23 13:12:14] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042605:OPTIONS (Method 3) (No timer T1) [May 23 13:12:14] VERBOSE[13067] chan_sip.c: Retransmitting #1 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4941072c Max-Forwards: 70 From: "unknown" ;tag=as2e08a8f2 To: Contact: Call-ID: 6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:14] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:14] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:15] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042605:OPTIONS (Method 3) (No timer T1) [May 23 13:12:15] VERBOSE[13067] chan_sip.c: Retransmitting #2 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4941072c Max-Forwards: 70 From: "unknown" ;tag=as2e08a8f2 To: Contact: Call-ID: 6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:15] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:15] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:15] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:16] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.44.130:5060 ---> OPTIONS sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK76e3fa0c;rport Max-Forwards: 70 From: "asterisk" ;tag=as1664597a To: Contact: Call-ID: 3cae8f2579381430056cf7be37ccb991@209.191.44.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 17:12:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK76e3fa0c;rport [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as1664597a [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 5 [ 38]: Contact: [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 6 [ 56]: Call-ID: 3cae8f2579381430056cf7be37ccb991@209.191.44.130 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:16 GMT [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 13:12:16] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:16] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3cae8f2579381430056cf7be37ccb991@209.191.44.130 (Checking From) --From tag as1664597a --To-tag [May 23 13:12:16] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 13:12:16] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 3cae8f2579381430056cf7be37ccb991@209.191.44.130 - OPTIONS (No RTP) [May 23 13:12:16] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 13:12:16] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 13:12:16] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.44.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK76e3fa0c;rport;received=209.191.44.130 From: "asterisk" ;tag=as1664597a To: ;tag=as0c54bfa2 Call-ID: 3cae8f2579381430056cf7be37ccb991@209.191.44.130 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> [May 23 13:12:16] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 13:12:16] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '3cae8f2579381430056cf7be37ccb991@209.191.44.130' in 32000 ms (Method: OPTIONS) [May 23 13:12:16] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:16] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:16] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042605:OPTIONS (Method 3) (No timer T1) [May 23 13:12:16] VERBOSE[13067] chan_sip.c: Retransmitting #3 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4941072c Max-Forwards: 70 From: "unknown" ;tag=as2e08a8f2 To: Contact: Call-ID: 6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:16] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:16] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:16] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> NOTIFY sip:64.19.145.13 SIP/2.0 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 28174 NOTIFY Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-a2c5f065 Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Contact: "SPA303 Cisco" Event: keep-alive Content-Length: 0 <-------------> [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 2 [ 22]: To: [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 3 [ 38]: Call-ID: 5d07fe66-394bec48@10.0.15.101 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 4 [ 18]: CSeq: 28174 NOTIFY [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 5 [ 60]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-a2c5f065 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 7 [ 30]: User-Agent: Cisco/SPA303-7.4.6 [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 8 [ 82]: Contact: "SPA303 Cisco" [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Event: keep-alive [May 23 13:12:16] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:16] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:16] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5d07fe66-394bec48@10.0.15.101 (Checking From) --From tag c7d0e91e95d40f0o0 --To-tag [May 23 13:12:16] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:16] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:16] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-a2c5f065;received=209.191.39.117 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: ;tag=as1046500e Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 28174 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:16] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:16] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '5d07fe66-394bec48@10.0.15.101' in 32000 ms (Method: NOTIFY) [May 23 13:12:16] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:16] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 7fca07167f33d9d33f5baf5747fcbc2f@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:17] DEBUG[13067] acl.c: For destination '64.19.145.18', our source address is '64.19.145.13'. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.18 SIP/2.0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2d19dc5c [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as15c07c92 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:17 GMT [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.18:5060: OPTIONS sip:64.19.145.18 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2d19dc5c Max-Forwards: 70 From: "unknown" ;tag=as15c07c92 To: Contact: Call-ID: 00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042615 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.18:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.18:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2d19dc5c;received=64.19.145.13 From: "unknown" ;tag=as15c07c92 To: ;tag=as4178e9eb Call-ID: 00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2d19dc5c;received=64.19.145.13 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as15c07c92 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as4178e9eb [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060 (Checking To) --From tag as15c07c92 --To-tag as4178e9eb [May 23 13:12:17] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042615 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Destroying SIP dialog 00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '00656b024ba6ca7473193719355ba7b5@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 489fe74a0060228c68c5a8cf77625786@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:17] DEBUG[13067] acl.c: For destination '64.19.145.15', our source address is '64.19.145.13'. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.15 SIP/2.0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK61979c32 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as1b17da49 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:17 GMT [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.15:5060: OPTIONS sip:64.19.145.15 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK61979c32 Max-Forwards: 70 From: "unknown" ;tag=as1b17da49 To: Contact: Call-ID: 1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042618 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.15:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.15:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK61979c32;received=64.19.145.13 From: "unknown" ;tag=as1b17da49 To: ;tag=as3cbab0d2 Call-ID: 1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK61979c32;received=64.19.145.13 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as1b17da49 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as3cbab0d2 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060 (Checking To) --From tag as1b17da49 --To-tag as3cbab0d2 [May 23 13:12:17] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042618 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Destroying SIP dialog 1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '1fa416c7788846f86e07fa333263d9eb@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042605:OPTIONS (Method 3) (No timer T1) [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Retransmitting #4 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4941072c Max-Forwards: 70 From: "unknown" ;tag=as2e08a8f2 To: Contact: Call-ID: 6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Destroying SIP dialog 6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '6beeae4927235cd72fb29a3d4ea52d49@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 30e882c060f46da76f4c127677a3f0ae@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:17] DEBUG[13067] acl.c: For destination '64.19.145.7', our source address is '64.19.145.13'. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 31]: OPTIONS sip:64.19.145.7 SIP/2.0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK239d4e21 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as14bfa72b [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 21]: To: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:17 GMT [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: OPTIONS sip:64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK239d4e21 Max-Forwards: 70 From: "unknown" ;tag=as14bfa72b To: Contact: Call-ID: 184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042622 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK239d4e21;received=64.19.145.13 From: "unknown" ;tag=as14bfa72b To: ;tag=as61ed1d18 Call-ID: 184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK239d4e21;received=64.19.145.13 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as14bfa72b [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 36]: To: ;tag=as61ed1d18 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060 (Checking To) --From tag as14bfa72b --To-tag as61ed1d18 [May 23 13:12:17] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042622 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Destroying SIP dialog 184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '184c96af5bdfc4121cb7ad604227c499@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 6dc78985106501476fbc024905dad82e@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:17] DEBUG[13067] acl.c: For destination '64.19.145.12', our source address is '64.19.145.13'. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.12 SIP/2.0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK72f888e2 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as5c9b3308 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:17 GMT [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.12:5060: OPTIONS sip:64.19.145.12 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK72f888e2 Max-Forwards: 70 From: "unknown" ;tag=as5c9b3308 To: Contact: Call-ID: 5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042625 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.12:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.12:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK72f888e2;received=64.19.145.13 From: "unknown" ;tag=as5c9b3308 To: ;tag=as17021b52 Call-ID: 5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK72f888e2;received=64.19.145.13 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as5c9b3308 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as17021b52 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060 (Checking To) --From tag as5c9b3308 --To-tag as17021b52 [May 23 13:12:17] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042625 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Destroying SIP dialog 5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '5be4408e0ae3b0b60964b09062f92df4@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> INVITE sip:Anonymous@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=C5E6782D-50C4FB20 To: "Anonymous";tag=as552f30c6 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 1 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK1f55e02bF3C74C0E Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Supported: 100rel Supported: replaces Max-Forwards: 70 Allow-Events: talk,hold,conference Content-Type: application/SDP Content-Length: 177 v=0 o=- 1306170710 1306170712 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendonly m=audio 51818 RTP/AVP 0 a=sendonly a=rtpmap:0 PCMU/8000 <-------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 46]: INVITE sip:Anonymous@64.19.145.13:5060 SIP/2.0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [102]: From: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 58]: To: "Anonymous";tag=as552f30c6 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK1f55e02bF3C74C0E [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Supported: 100rel [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 13 [ 34]: Allow-Events: talk,hold,conference [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 14 [ 29]: Content-Type: application/SDP [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 15 [ 19]: Content-Length: 177 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 16 [ 0]: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306170710 1306170712 IN IP4 209.191.39.117 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 5 [ 10]: a=sendonly [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 6 [ 23]: m=audio 51818 RTP/AVP 0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 7 [ 10]: a=sendonly [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: --- (16 headers 9 lines) --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking From) --From tag C5E6782D-50C4FB20 --To-tag as552f30c6 [May 23 13:12:17] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 13:12:17] DEBUG[13067] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel" [May 23 13:12:17] DEBUG[13067] sip/reqresp_parser.c: Found SIP option: -100rel- [May 23 13:12:17] DEBUG[13067] sip/reqresp_parser.c: Matched SIP option: 100rel [May 23 13:12:17] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:17] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 13:12:17] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 1306170710 1306170712 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [May 23 13:12:17] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 13:12:17] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP a=sendonly... OK. [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:17] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd39c [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:17] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd39c [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 13:12:17] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6646b68' [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51818 [May 23 13:12:17] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd39c to 0xb6646d14 [May 23 13:12:17] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 13:12:17] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 13:12:17] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6646b68' [May 23 13:12:17] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:17] DEBUG[13067] chan_sip.c: Got a SIP re-invite for call 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:17] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:17] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 13:12:17] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 13:12:17] DEBUG[13067] chan_sip.c: SIP/322-eng-00000016: This call is UP.... [May 23 13:12:17] DEBUG[13109] manager.c: Examining event: Event: Hold Privilege: call,all Status: On Channel: SIP/322-eng-00000016 Uniqueid: 1306170731.22 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK1f55e02bF3C74C0E;received=209.191.39.117 From: "Poly_test ENG";tag=C5E6782D-50C4FB20 To: "Anonymous";tag=as552f30c6 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Setting framing from config on incoming call [May 23 13:12:17] DEBUG[13067] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 13:12:17] DEBUG[13067] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Audio is at 5060 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:17] DEBUG[13067] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:17] DEBUG[13067] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- Reliably Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK1f55e02bF3C74C0E;received=209.191.39.117 From: "Poly_test ENG";tag=C5E6782D-50C4FB20 To: "Anonymous";tag=as552f30c6 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 189 v=0 o=root 41473503 41473505 IN IP4 64.19.145.7 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.7 t=0 0 m=audio 12008 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=recvonly <------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042628 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13070] app_queue.c: Extension '322@local-extensions-eng' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [May 23 13:12:17] DEBUG[19459] chan_sip.c: Sending reinvite on SIP '4eff848341deec190001f2470396b9ea@64.19.145.7' - It's audio soon redirected to IP 64.19.145.13:5060 [May 23 13:12:17] DEBUG[19459] chan_sip.c: Strict routing enforced for session 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:17] VERBOSE[19459] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:17] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 322 Context: local-extensions-eng Hint: SIP/322-eng Status: 16 [May 23 13:12:17] DEBUG[19459] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:17] DEBUG[19459] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:17] VERBOSE[19459] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 13:12:17] DEBUG[19459] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 13:12:17] DEBUG[19459] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 13:12:17] VERBOSE[19459] chan_sip.c: Audio is at 5060 [May 23 13:12:17] VERBOSE[19459] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:17] DEBUG[19459] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:17] DEBUG[19459] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 13:12:17] DEBUG[19459] chan_sip.c: Initializing already initialized SIP dialog 4eff848341deec190001f2470396b9ea@64.19.145.7 (presumably reinvite) [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 0 [ 40]: INVITE sip:Anonymous@64.19.145.7 SIP/2.0 [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229bb10a;rport [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 3 [ 50]: From: ;tag=as60db63bb [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 4 [ 64]: To: "Anonymous" ;tag=as6423a45f [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 5 [ 43]: Contact: [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 6 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 10 [ 19]: Supported: replaces [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 23 13:12:17] DEBUG[19459] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 13:12:17] VERBOSE[19459] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: INVITE sip:Anonymous@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229bb10a;rport Max-Forwards: 70 From: ;tag=as60db63bb To: "Anonymous" ;tag=as6423a45f Contact: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 193 v=0 o=root 272259394 272259396 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.13 t=0 0 m=audio 12964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 13:12:17] DEBUG[19459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042629 [May 23 13:12:17] DEBUG[19459] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:17] DEBUG[19459] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:17] VERBOSE[19459] res_musiconhold.c: -- Started music on hold, class 'default-eng', on SIP/mg2-00000015 [May 23 13:12:17] DEBUG[19459] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [May 23 13:12:17] DEBUG[19459] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:17] DEBUG[13109] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/mg2-00000015 UniqueID: 1306170730.21 Class: default-eng [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229bb10a;received=64.19.145.13;rport=5060 From: ;tag=as60db63bb To: "Anonymous" ;tag=as6423a45f Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <-------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229bb10a;received=64.19.145.13;rport=5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as60db63bb [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 64]: To: "Anonymous" ;tag=as6423a45f [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 36]: Contact: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking To) --From tag as60db63bb --To-tag as6423a45f [May 23 13:12:17] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1042629 - INVITE (got response) [May 23 13:12:17] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4eff848341deec190001f2470396b9ea@64.19.145.7' Request 103: Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: SIP response 100 to RE-invite on outgoing call 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/mg2-00000015~ Value: SIP 100 Trying Uniqueid: 1306170730.21 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229bb10a;received=64.19.145.13;rport=5060 From: ;tag=as60db63bb To: "Anonymous" ;tag=as6423a45f Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 223 v=0 o=root 1302972891 1302972894 IN IP4 64.19.145.7 s=Asterisk PBX SVN-branch-1.6.1-r230383M c=IN IP4 64.19.145.7 t=0 0 m=audio 12008 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229bb10a;received=64.19.145.13;rport=5060 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as60db63bb [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 3 [ 64]: To: "Anonymous" ;tag=as6423a45f [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 9 [ 36]: Contact: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Content-Length: 223 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Header 12 [ 0]: [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=root 1302972891 1302972894 IN IP4 64.19.145.7 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 2 [ 40]: s=Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 3 [ 20]: c=IN IP4 64.19.145.7 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 12008 RTP/AVP 0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 7 [ 25]: a=silenceSupp:off - - - - [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 8 [ 10]: a=ptime:20 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=sendrecv [May 23 13:12:17] VERBOSE[13067] chan_sip.c: --- (12 headers 10 lines) --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking To) --From tag as60db63bb --To-tag as6423a45f [May 23 13:12:17] DEBUG[13067] chan_sip.c: Acked pending invite 103 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '4eff848341deec190001f2470396b9ea@64.19.145.7' of Request 103: Match Found [May 23 13:12:17] DEBUG[13067] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP o=root 1302972891 1302972894 IN IP4 64.19.145.7... UNSUPPORTED. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Asterisk PBX SVN-branch-1.6.1-r230383M... UNSUPPORTED. [May 23 13:12:17] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:17] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 64.19.145.7... OK. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:17] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 13:12:17] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 13:12:17] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb71129c8' [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 64.19.145.7:12008 [May 23 13:12:17] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xb7112b74 [May 23 13:12:17] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 13:12:17] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 13:12:17] DEBUG[13067] chan_sip.c: Updating call counter for incoming call [May 23 13:12:17] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:17] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:17] DEBUG[13067] chan_sip.c: Strict routing enforced for session 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:17] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 13:12:17] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 13:12:17] VERBOSE[13067] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 13:12:17] VERBOSE[13067] chan_sip.c: Transmitting (no NAT) to 64.19.145.7:5060: ACK sip:Anonymous@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK660181db;rport Max-Forwards: 70 From: ;tag=as60db63bb To: "Anonymous" ;tag=as6423a45f Contact: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 13:12:17] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:Ano' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ~HASH~SIP_CAUSE~SIP/mg2-00000015~ Value: SIP 200 OK Uniqueid: 1306170730.21 [May 23 13:12:17] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 8 Paused: 0 [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:17] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 13:12:17] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_musiconhold.c: SIP/mg2-00000015 Opened file 0 '/var/lib/asterisk/moh/eng/default/macroform-cold_day' [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xb71129c8' [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:18] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 68f0e8d821466efc07037e277ac9d97e@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:18] DEBUG[13067] acl.c: For destination '64.19.145.11', our source address is '64.19.145.13'. [May 23 13:12:18] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.11 SIP/2.0 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK78e8784f [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as11f94e39 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:18 GMT [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:18] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.11:5060: OPTIONS sip:64.19.145.11 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK78e8784f Max-Forwards: 70 From: "unknown" ;tag=as11f94e39 To: Contact: Call-ID: 19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:18] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042631 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.11:5060 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:18] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.11:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK78e8784f;received=64.19.145.13 From: "unknown" ;tag=as11f94e39 To: ;tag=as7ee247a7 Call-ID: 19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK78e8784f;received=64.19.145.13 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as11f94e39 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as7ee247a7 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:18] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:18] DEBUG[13067] chan_sip.c: = Looking for Call ID: 19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060 (Checking To) --From tag as11f94e39 --To-tag as7ee247a7 [May 23 13:12:18] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042631 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Stopping retransmission on '19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: INVITE [May 23 13:12:18] DEBUG[13067] chan_sip.c: Destroying SIP dialog 19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060 [May 23 13:12:18] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '19b7ef5802386a885eb0b4c51a5db487@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> ACK sip:Anonymous@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=C5E6782D-50C4FB20 To: "Anonymous";tag=as552f30c6 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 1 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK646ea10a9F21F865 Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 0 [ 43]: ACK sip:Anonymous@64.19.145.13:5060 SIP/2.0 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 1 [102]: From: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 2 [ 58]: To: "Anonymous";tag=as552f30c6 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK646ea10a9F21F865 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 13:12:18] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 13:12:18] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking From) --From tag C5E6782D-50C4FB20 --To-tag as552f30c6 [May 23 13:12:18] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 13:12:18] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042628 [May 23 13:12:18] DEBUG[13067] chan_sip.c: Stopping retransmission on '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' of Response 1: Match Found [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '008a59cc1271d0c95a52b34215500a1d@209.191.44.130' [May 23 13:12:18] DEBUG[13067] chan_sip.c: Destroying SIP dialog 008a59cc1271d0c95a52b34215500a1d@209.191.44.130 [May 23 13:12:18] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '008a59cc1271d0c95a52b34215500a1d@209.191.44.130' Method: OPTIONS [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:18] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 304b5fe25cfd06872baa885029979e9a@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:19] DEBUG[13067] acl.c: For destination '209.191.13.243', our source address is '64.19.145.13'. [May 23 13:12:19] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 0 [ 46]: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK3f742c31;rport [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as4cc71cb7 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 4 [ 36]: To: [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:19 GMT [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:19] VERBOSE[13067] chan_sip.c: Reliably Transmitting (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK3f742c31;rport Max-Forwards: 70 From: "unknown" ;tag=as4cc71cb7 To: Contact: Call-ID: 08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:19] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042634 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113876 NOTIFY [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:19] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:19] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:19] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:19] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:19] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:19] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:19] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:19] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:19] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113876 NOTIFY [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:20] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:20] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 113876, ours 113876) [May 23 13:12:20] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042634:OPTIONS (Method 3) (No timer T1) [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Retransmitting #1 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK3f742c31;rport Max-Forwards: 70 From: "unknown" ;tag=as4cc71cb7 To: Contact: Call-ID: 08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:20] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> INVITE sip:312@64.19.145.13;user=phone SIP/2.0 From: "Poly_test ENG";tag=B96595C-F28D6247 To: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 1 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK92560118CDA2B9E3 Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Supported: 100rel Supported: replaces Max-Forwards: 70 Allow-Events: talk,hold,conference Content-Type: application/SDP Content-Length: 252 v=0 o=- 1306170717 1306170717 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendrecv m=audio 51820 RTP/AVP 0 110 127 a=rtpmap:0 PCMU/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=30 a=rtpmap:127 telephone-event/8000 <-------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 0 [ 46]: INVITE sip:312@64.19.145.13;user=phone SIP/2.0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "Poly_test ENG";tag=B96595C-F28D6247 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 2 [ 37]: To: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK92560118CDA2B9E3 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Supported: 100rel [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 13 [ 34]: Allow-Events: talk,hold,conference [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 14 [ 29]: Content-Type: application/SDP [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 15 [ 19]: Content-Length: 252 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 16 [ 0]: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306170717 1306170717 IN IP4 209.191.39.117 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 5 [ 10]: a=sendrecv [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 6 [ 31]: m=audio 51820 RTP/AVP 0 110 127 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 8 [ 22]: a=rtpmap:110 iLBC/8000 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 9 [ 18]: a=fmtp:110 mode=30 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 10 [ 33]: a=rtpmap:127 telephone-event/8000 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: --- (16 headers 11 lines) --- [May 23 13:12:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 (Checking From) --From tag B96595C-F28D6247 --To-tag [May 23 13:12:20] DEBUG[13067] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 13:12:20] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 - INVITE (No RTP) [May 23 13:12:20] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 13:12:20] DEBUG[13067] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel" [May 23 13:12:20] DEBUG[13067] sip/reqresp_parser.c: Found SIP option: -100rel- [May 23 13:12:20] DEBUG[13067] sip/reqresp_parser.c: Matched SIP option: 100rel [May 23 13:12:20] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:20] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 13:12:20] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Using INVITE request as basis request - a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Found peer '322-eng' for '322-eng' from 209.191.39.117:5060 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- Reliably Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK92560118CDA2B9E3;received=209.191.39.117 From: "Poly_test ENG";tag=B96595C-F28D6247 To: ;tag=as3ae0f6cf Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="418a2ab7" Content-Length: 0 <------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042638 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' in 32000 ms (Method: INVITE) [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> ACK sip:312@64.19.145.13;user=phone SIP/2.0 From: "Poly_test ENG";tag=B96595C-F28D6247 To: ;tag=as3ae0f6cf Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 1 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK92560118CDA2B9E3 Content-Length: 0 <-------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 0 [ 43]: ACK sip:312@64.19.145.13;user=phone SIP/2.0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "Poly_test ENG";tag=B96595C-F28D6247 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 2 [ 52]: To: ;tag=as3ae0f6cf [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK92560118CDA2B9E3 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: --- (7 headers 0 lines) --- [May 23 13:12:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 (Checking From) --From tag B96595C-F28D6247 --To-tag as3ae0f6cf [May 23 13:12:20] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042638 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Stopping retransmission on 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' of Response 1: Match Found [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> INVITE sip:312@64.19.145.13;user=phone SIP/2.0 From: "Poly_test ENG";tag=B96595C-F28D6247 To: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 2 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfae9d6ffF4B3FF82 Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Supported: 100rel Supported: replaces Max-Forwards: 70 Allow-Events: talk,hold,conference Authorization: Digest username="322-eng",realm="asterisk",nonce="418a2ab7",uri="sip:312@64.19.145.13;user=phone",response="d8ba6ae2424ea26c35322aefe70b7219",algorithm=MD5 Content-Type: application/SDP Content-Length: 252 v=0 o=- 1306170717 1306170717 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendrecv m=audio 51820 RTP/AVP 0 110 127 a=rtpmap:0 PCMU/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=30 a=rtpmap:127 telephone-event/8000 <-------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 0 [ 46]: INVITE sip:312@64.19.145.13;user=phone SIP/2.0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "Poly_test ENG";tag=B96595C-F28D6247 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 2 [ 37]: To: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 4 [ 14]: CSeq: 2 INVITE [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfae9d6ffF4B3FF82 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Supported: 100rel [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 13 [ 34]: Allow-Events: talk,hold,conference [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 14 [170]: Authorization: Digest username="322-eng",realm="asterisk",nonce="418a2ab7",uri="sip:312@64.19.145.13;user=phone",response="d8ba6ae2424ea26c35322aefe70b7219",algorithm=MD5 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 15 [ 29]: Content-Type: application/SDP [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 16 [ 19]: Content-Length: 252 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 17 [ 0]: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306170717 1306170717 IN IP4 209.191.39.117 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 5 [ 10]: a=sendrecv [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 6 [ 31]: m=audio 51820 RTP/AVP 0 110 127 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 8 [ 22]: a=rtpmap:110 iLBC/8000 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 9 [ 18]: a=fmtp:110 mode=30 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Body 10 [ 33]: a=rtpmap:127 telephone-event/8000 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: --- (17 headers 11 lines) --- [May 23 13:12:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 (Checking From) --From tag B96595C-F28D6247 --To-tag [May 23 13:12:20] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:20] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:20] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:20] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:20] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 13:12:20] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:20] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 13:12:20] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Using INVITE request as basis request - a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Found peer '322-eng' for '322-eng' from 209.191.39.117:5060 [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb7331dd8' [May 23 13:12:20] DEBUG[13067] res_rtp_asterisk.c: Allocated port 18842 for RTP instance '0xb7331dd8' [May 23 13:12:20] DEBUG[13067] rtp_engine.c: RTP instance '0xb7331dd8' is setup and ready to go [May 23 13:12:20] DEBUG[13067] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb7331dd8' [May 23 13:12:20] VERBOSE[13067] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Setting NAT on RTP to Off [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 1306170717 1306170717 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [May 23 13:12:20] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 13:12:20] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd39c [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Found RTP audio format 110 [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Setting payload 110 based on m type on 0xb7cfd39c [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Found RTP audio format 127 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Found audio description format iLBC for ID 110 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 iLBC/8000... OK. [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=fmtp:110 mode=30... UNSUPPORTED. [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Found audio description format telephone-event for ID 127 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:127 telephone-event/8000... OK. [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd39c [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Incorporating payload 110 on 0xb7cfd39c [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Incorporating payload 127 on 0xb7cfd39c [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x404 (ulaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x404 (ulaw|ilbc) [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [May 23 13:12:20] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb7331dd8' [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51820 [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd39c to 0xb7331f84 [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Copying payload 110 from 0xb7cfd39c to 0xb7331f84 [May 23 13:12:20] DEBUG[13067] rtp_engine.c: Copying payload 127 from 0xb7cfd39c to 0xb7331f84 [May 23 13:12:20] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x404 (ulaw|ilbc) [May 23 13:12:20] DEBUG[13067] chan_sip.c: Checking SIP call limits for device 322-eng [May 23 13:12:20] DEBUG[13067] chan_sip.c: Updating call counter for incoming call [May 23 13:12:20] DEBUG[13067] chan_sip.c: Call from peer '322-eng' is 2 out of 2147483647 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: Looking for 312 in from-inside-eng (domain 64.19.145.13) [May 23 13:12:20] DEBUG[13067] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 13:12:20] DEBUG[13067] chan_sip.c: *** Joint capabilities are 0x404 (ulaw|ilbc) [May 23 13:12:20] DEBUG[13067] chan_sip.c: *** Our capabilities are 0x404 (ulaw|ilbc) [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/322-eng-00000018 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 322 CallerIDName: Poly_test ENG AccountCode: eng Exten: 312 Context: from-inside-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13067] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 13:12:20] DEBUG[13067] chan_sip.c: This channel will not be able to handle video. [May 23 13:12:20] DEBUG[13067] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 13:12:20] DEBUG[13067] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 13:12:20] DEBUG[13067] chan_sip.c: build_route: Contact hop: [May 23 13:12:20] VERBOSE[13067] chan_sip.c: list_route: hop: [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: SIPURI Value: sip:322-eng@209.191.39.117:5060 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13067] chan_sip.c: SIP/322-eng-00000018: New call is still down.... Trying... [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: SIPDOMAIN Value: 64.19.145.13 Uniqueid: 1306170740.24 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfae9d6ffF4B3FF82;received=209.191.39.117 From: "Poly_test ENG";tag=B96595C-F28D6247 To: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: SIPCALLID Value: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/322-eng-00000018 Uniqueid: 1306170740.24 Channeltype: SIP SIPcallid: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 SIPfullcontact: sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/322-eng-00000018 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 322 CallerIDName: Poly_test ENG Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Macro' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [312@from-inside-eng:1] Macro("SIP/322-eng-00000018", "tl-set-variables2,from-inside-redir-eng,eng") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: from-inside-eng Extension: 312 Priority: 1 Application: Macro AppData: tl-set-variables2,from-inside-redir-eng,eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_EXTEN Value: 312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_CONTEXT Value: from-inside-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_PRIORITY Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ARG1 Value: from-inside-redir-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG2' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-variables2:1] Set("SIP/322-eng-00000018", "__tenant=eng") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ARG2 Value: eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-variables2 Extension: s Priority: 1 Application: Set AppData: __tenant=eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __tenant Value: eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG2' is 'eng' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-variables2:2] Set("SIP/322-eng-00000018", "CDR(userfield)=eng") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-variables2 Extension: s Priority: 2 Application: Set AppData: CDR(userfield)=eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-variables2:3] Set("SIP/322-eng-00000018", "__FROM_INSIDE=1") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-variables2 Extension: s Priority: 3 Application: Set AppData: __FROM_INSIDE=1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __FROM_INSIDE Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TL_DASH' is '-' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG2' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is 'default-eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-variables2:4] Set("SIP/322-eng-00000018", "__MOH=default-eng") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DB_RESULT Value: default-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-variables2 Extension: s Priority: 4 Application: Set AppData: __MOH=default-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __MOH Value: default-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'MOH' is 'default-eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-variables2:5] GotoIf("SIP/322-eng-00000018", "1 ?setmoh") in new stack [May 23 13:12:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:20] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:20] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 13:12:20] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-variables2 Extension: s Priority: 5 Application: GotoIf AppData: 1 ?setmoh Uniqueid: 1306170740.24 [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-set-variables2,s,7) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'MOH' is 'default-eng' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-variables2:7] Set("SIP/322-eng-00000018", "CHANNEL(musicclass)=default-eng") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-variables2 Extension: s Priority: 7 Application: Set AppData: CHANNEL(musicclass)=default-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG2' is 'eng' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-variables2:8] Set("SIP/322-eng-00000018", "CHANNEL(parkinglot)=parkinglot_eng") in new stack [May 23 13:12:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-variables2 Extension: s Priority: 8 Application: Set AppData: CHANNEL(parkinglot)=parkinglot_eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG1' is 'from-inside-redir-eng' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'MACRO_EXTEN' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Goto' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-variables2:9] Goto("SIP/322-eng-00000018", "from-inside-redir-eng,312,1") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-variables2 Extension: s Priority: 9 Application: Goto AppData: from-inside-redir-eng,312,1 Uniqueid: 1306170740.24 [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (from-inside-redir-eng,312,1) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Goto [May 23 13:12:20] VERBOSE[19460] app_macro.c: == Channel 'SIP/322-eng-00000018' jumping out of macro 'tl-set-variables2' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 0 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Macro' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [312@from-inside-redir-eng:1] Macro("SIP/322-eng-00000018", "tl-userexten,SIP/312-eng,312@default-eng,") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: from-inside-redir-eng Extension: 312 Priority: 1 Application: Macro AppData: tl-userexten,SIP/312-eng,312@default-eng, Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_EXTEN Value: 312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_CONTEXT Value: from-inside-redir-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_PRIORITY Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ARG1 Value: SIP/312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ARG2 Value: 312@default-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ARG3 Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'MACRO_EXTEN' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten:1] Set("SIP/322-eng-00000018", "__DIALED_NUMBER=312") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten Extension: s Priority: 1 Application: Set AppData: __DIALED_NUMBER=312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __DIALED_NUMBER Value: 312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'MACRO_EXTEN' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TL_DASH' is '-' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten:2] Set("SIP/322-eng-00000018", "__PICKUPMARK=312-eng") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten Extension: s Priority: 2 Application: Set AppData: __PICKUPMARK=312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __PICKUPMARK Value: 312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '322' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '3' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'TENANT/eng/usedistinctring' in family 'TL' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DB_RESULT Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'ExecIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten:3] ExecIf("SIP/322-eng-00000018", "0?SIPAddHeader(Alert-Info: <>)") in new stack [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten Extension: s Priority: 3 Application: ExecIf AppData: 0?SIPAddHeader(Alert-Info: <>) Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: ExecIf [May 23 13:12:20] DEBUG[19460] pbx.c: Evaluating 'LEN(${CALLERID(num)})' (from 'LEN(${CALLERID(num)})} < 7' len 21) [May 23 13:12:20] DEBUG[19460] pbx.c: Evaluating 'CALLERID(num)' (from 'CALLERID(num)})' len 13) [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '322' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '3' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Evaluating 'DB_EXISTS(TL/TENANT/${tenant}/usedistinctring)' (from 'DB_EXISTS(TL/TENANT/${tenant}/usedistinctring)}' len 46) [May 23 13:12:20] DEBUG[19460] pbx.c: Evaluating 'tenant' (from 'tenant}/usedistinctring)' len 6) [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'TENANT/eng/usedistinctring' in family 'TL' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Evaluating 'DB(TL/TENANT/${tenant}/intalertinfo)' (from 'DB(TL/TENANT/${tenant}/intalertinfo)}>)' len 36) [May 23 13:12:20] DEBUG[19460] pbx.c: Evaluating 'tenant' (from 'tenant}/intalertinfo)' len 6) [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'RINGGROUP_TIMEOUT' is NULL [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'BLINDTRANSFER' is NULL [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten:4] GotoIf("SIP/322-eng-00000018", "0?doingringgroup") in new stack [May 23 13:12:20] DEBUG[19460] pbx.c: Not taking any branch [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG1' is 'SIP/312-eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG2' is '312@default-eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG3' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Macro' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten:5] Macro("SIP/322-eng-00000018", "tl-userexten-base,SIP/312-eng,312@default-eng,") in new stack [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '322' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'DIALED_NUMBER' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:1] GotoIf("SIP/322-eng-00000018", "1?set_options") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,8) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:8] Set("SIP/322-eng-00000018", "OPTIONS=rtT") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG1' is 'SIP/312-eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:9] Set("SIP/322-eng-00000018", "__PHONE=SIP/312-eng") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ARG2' is '312@default-eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:10] Set("SIP/322-eng-00000018", "__VM_MBOX=312@default-eng") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TL_DASH' is '-' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'DIALED_NUMBER' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:11] Set("SIP/322-eng-00000018", "THISEXT=TL/eng-312") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TL_DASH' is '-' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'DIALED_NUMBER' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:12] Set("SIP/322-eng-00000018", "_CLIMYID=eng-312") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'DIALED_NUMBER' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TL_DASH' is '-' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:13] Set("SIP/322-eng-00000018", "THISCHAN=TL/312-eng") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DB_RESULT Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten Extension: s Priority: 4 Application: GotoIf AppData: 0?doingringgroup Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten Extension: s Priority: 5 Application: Macro AppData: tl-userexten-base,SIP/312-eng,312@default-eng, Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_EXTEN Value: s Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_CONTEXT Value: macro-tl-userexten Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_PRIORITY Value: 5 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ARG1 Value: SIP/312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ARG2 Value: 312@default-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ARG3 Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 1 Application: GotoIf AppData: 1?set_options Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 8 Application: Set AppData: OPTIONS=rtT Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: OPTIONS Value: rtT Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 9 Application: Set AppData: __PHONE=SIP/312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __PHONE Value: SIP/312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 10 Application: Set AppData: __VM_MBOX=312@default-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __VM_MBOX Value: 312@default-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 11 Application: Set AppData: THISEXT=TL/eng-312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: THISEXT Value: TL/eng-312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 12 Application: Set AppData: _CLIMYID=eng-312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: _CLIMYID Value: eng-312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 13 Application: Set AppData: THISCHAN=TL/312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: THISCHAN Value: TL/312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ORIG_EXTEN' is NULL [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:14] GotoIf("SIP/322-eng-00000018", "0?beenhere") in new stack [May 23 13:12:20] DEBUG[19460] pbx.c: Not taking any branch [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'DIALED_NUMBER' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:15] Set("SIP/322-eng-00000018", "_ORIG_EXTEN=312") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TL_DASH' is '-' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'DIALED_NUMBER' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:16] Set("SIP/322-eng-00000018", "_ORIG_EXTEN_USER=TL/eng-312") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Macro' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:17] Macro("SIP/322-eng-00000018", "tl-notify") in new stack [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/ADDRESS' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/ADDRESS not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-notify:1] Set("SIP/322-eng-00000018", "ADDRESS=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'DIALED_NUMBER' is '312' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '322' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'UserEvent' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-notify:2] UserEvent("SIP/322-eng-00000018", "TlNotify,dialed: 312|callerID: 322|tenant: eng") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: UserEvent [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'NoOp' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-notify:3] NoOp("SIP/322-eng-00000018", "TL/eng-312") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: NoOp [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ADDRESS' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-notify:4] GotoIf("SIP/322-eng-00000018", "1?s-exit,1") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-notify,s-exit,1) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'MacroExit' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s-exit@macro-tl-notify:1] MacroExit("SIP/322-eng-00000018", "") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Macro [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Goto' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:18] Goto("SIP/322-eng-00000018", "checkformat") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,20) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Goto [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'RECORDING_FORMAT' is 'ulaw' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:20] GotoIf("SIP/322-eng-00000018", "1?cont1") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,22) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:22] Set("SIP/322-eng-00000018", "RECORD_CALLEE=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Macro' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:23] Macro("SIP/322-eng-00000018", "tl-set-myvariables") in new stack [May 23 13:12:20] DEBUG[19460] pbx.c: Evaluating 'CHANNEL' (from 'CHANNEL}' len 7) [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'CHANNEL' is 'SIP/322-eng-00000018' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '322-eng-00000018' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-myvariables:1] Set("SIP/322-eng-00000018", "MY_CHAN=322-eng-00000018") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'CHANNEL' is 'SIP/322-eng-00000018' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'NoOp' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-myvariables:2] NoOp("SIP/322-eng-00000018", "THECHANNEL=SIP/322-eng-00000018") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: NoOp [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'CHANNEL' is 'SIP/322-eng-00000018' [May 23 13:12:20] DEBUG[19460] func_strings.c: FUNCTION REGEX (^Zap/|^DAHDI/)(SIP/322-eng-00000018) [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-myvariables:3] Set("SIP/322-eng-00000018", "zap=0") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'zap' is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-myvariables:4] GotoIf("SIP/322-eng-00000018", "1?usechannel") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-set-myvariables,s,9) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'CHANNEL' is 'SIP/322-eng-00000018' [May 23 13:12:20] DEBUG[19460] func_strings.c: FUNCTION REGEX (^Local/)(SIP/322-eng-00000018) [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-myvariables:9] Set("SIP/322-eng-00000018", "local=0") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'local' is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-myvariables:10] GotoIf("SIP/322-eng-00000018", "1?useit") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-set-myvariables,s,12) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Evaluating 'MY_CHAN' (from 'MY_CHAN}' len 7) [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'MY_CHAN' is '322-eng-00000018' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '322' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-myvariables:12] Set("SIP/322-eng-00000018", "__MYEXTENSION=322") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TL_DASH' is '-' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'MYEXTENSION' is '322' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-set-myvariables:13] Set("SIP/322-eng-00000018", "__MYID=eng-322") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Macro [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'MYID' is 'eng-322' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:24] Set("SIP/322-eng-00000018", "RECORD_CALLER=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:25] Set("SIP/322-eng-00000018", "VM=1") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/VMT0' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/VMT0 not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:26] Set("SIP/322-eng-00000018", "VMT0=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/CFNAEXT' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/CFNAEXT not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:27] Set("SIP/322-eng-00000018", "CFNAEXT=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/CFNAAN' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/CFNAAN not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:28] Set("SIP/322-eng-00000018", "CFNAAN=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'RECORD_CALLEE' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'RECORD_CALLER' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:29] GotoIf("SIP/322-eng-00000018", "1?done_checkrecord") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,47) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'NoOp' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:47] NoOp("SIP/322-eng-00000018", "") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: NoOp [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'RECORD_CALLEE' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'NoOp' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:48] NoOp("SIP/322-eng-00000018", "RECORD_CALLEE=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: NoOp [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'RECORD_CALLER' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'NoOp' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:49] NoOp("SIP/322-eng-00000018", "RECORD_CALLER=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: NoOp [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'OPTIONS' is 'rtT' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'NoOp' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:50] NoOp("SIP/322-eng-00000018", "OPTIONS=rtT") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: NoOp [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TOUCH_MONITOR' is NULL [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'NoOp' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:51] NoOp("SIP/322-eng-00000018", "TOUCH_MONITOR=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: NoOp [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'VM' is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:52] GotoIf("SIP/322-eng-00000018", "0?next1") in new stack [May 23 13:12:20] DEBUG[19460] pbx.c: Not taking any branch [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'VMT0' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:53] Set("SIP/322-eng-00000018", "TIMEOUT=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TIMEOUT' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:54] GotoIf("SIP/322-eng-00000018", "0?next1") in new stack [May 23 13:12:20] DEBUG[19460] pbx.c: Not taking any branch [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:55] Set("SIP/322-eng-00000018", "TIMEOUT=20") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'tenant' is 'eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:56] Set("SIP/322-eng-00000018", "CDR(userfield)=eng") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/SCREEN' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/SCREEN not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:57] Set("SIP/322-eng-00000018", "SCREEN=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'ORIG_EXTEN_USER' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/CFCONFIRM' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/CFCONFIRM not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:58] Set("SIP/322-eng-00000018", "CONFIRM=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'SCREEN' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:59] GotoIf("SIP/322-eng-00000018", "1?getblock") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,64) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/BLOCK' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/BLOCK not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:64] Set("SIP/322-eng-00000018", "BLOCK=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'BLOCK' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:65] GotoIf("SIP/322-eng-00000018", "1?getrecord") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,69) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:69] Set("SIP/322-eng-00000018", "RECORD=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '322' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'NoOp' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:70] NoOp("SIP/322-eng-00000018", "calleridnum=322 ") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: NoOp [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'BLOCK' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:71] GotoIf("SIP/322-eng-00000018", "1?screening") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,103) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'SCREEN' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:103] GotoIf("SIP/322-eng-00000018", "1?recording") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,129) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'RECORD' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:129] GotoIf("SIP/322-eng-00000018", "1?forwarding") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,131) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/CFA' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/CFA not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:131] Set("SIP/322-eng-00000018", "FORWARD=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'FORWARD' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:132] GotoIf("SIP/322-eng-00000018", "1?followmecheck") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,140) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'THISEXT' is 'TL/eng-312' [May 23 13:12:20] DEBUG[19460] db.c: Unable to find key 'eng-312/CFNA' in family 'TL' [May 23 13:12:20] DEBUG[19460] func_db.c: DB: TL/eng-312/CFNA not found in database. [May 23 13:12:20] DEBUG[19460] pbx.c: Function result is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:140] Set("SIP/322-eng-00000018", "FORWARD=") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Set' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:141] Set("SIP/322-eng-00000018", "__FOLLOWME=0") in new stack [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: Set [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'FORWARD' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'FORWARD' is '' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:142] GotoIf("SIP/322-eng-00000018", "1?checkchannel") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,154) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'PHONE' is 'SIP/312-eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'ChanIsAvail' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:154] ChanIsAvail("SIP/322-eng-00000018", "SIP/312-eng") in new stack [May 23 13:12:20] DEBUG[19460] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: Allocating new SIP dialog for 72554ce343b9556164ebe9342c4a3f9f@127.0.0.1:0 - INVITE (No RTP) [May 23 13:12:20] DEBUG[19460] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa0c8ee8' [May 23 13:12:20] DEBUG[19460] res_rtp_asterisk.c: Allocated port 11830 for RTP instance '0xa0c8ee8' [May 23 13:12:20] DEBUG[19460] rtp_engine.c: RTP instance '0xa0c8ee8' is setup and ready to go [May 23 13:12:20] DEBUG[19460] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa0c8ee8' [May 23 13:12:20] VERBOSE[19460] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Setting NAT on RTP to On [May 23 13:12:20] DEBUG[19460] chan_sip.c: OBPROXY: Not applying OBproxy to this call [May 23 13:12:20] DEBUG[19460] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 13:12:20] DEBUG[19460] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: This channel will not be able to handle video. [May 23 13:12:20] DEBUG[19460] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 13:12:20] DEBUG[19460] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 13:12:20] DEBUG[19460] channel.c: Hanging up channel 'SIP/312-eng-00000019' [May 23 13:12:20] DEBUG[19460] chan_sip.c: Hangup call SIP/312-eng-00000019, SIP callid 6f4c9d455bbb278d3d8622a635ae3420@64.19.145.13:5060 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Hanging up channel in state Down (not UP) [May 23 13:12:20] DEBUG[19460] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa0c8ee8' [May 23 13:12:20] VERBOSE[19460] chan_sip.c: Scheduling destruction of SIP dialog '6f4c9d455bbb278d3d8622a635ae3420@64.19.145.13:5060' in 32000 ms (Method: INVITE) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: ChanIsAvail [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'AVAILCHAN' is 'SIP/312-eng-00000019' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:155] GotoIf("SIP/322-eng-00000018", "1?chanavail") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,157) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'FOLLOWME' is '0' [May 23 13:12:20] DEBUG[19460] pbx.c: Expression result is '1' [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'GotoIf' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:157] GotoIf("SIP/322-eng-00000018", "1?dial") in new stack [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Goto (macro-tl-userexten-base,s,163) [May 23 13:12:20] DEBUG[19460] app_macro.c: Executed application: GotoIf [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'PHONE' is 'SIP/312-eng' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'TIMEOUT' is '20' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'OPTIONS' is 'rtT' [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'SCREENOPTIONS' is NULL [May 23 13:12:20] DEBUG[19460] pbx.c: Result of 'RECORDOPTIONS' is NULL [May 23 13:12:20] DEBUG[19460] pbx.c: Launching 'Dial' [May 23 13:12:20] VERBOSE[19460] pbx.c: -- Executing [s@macro-tl-userexten-base:163] Dial("SIP/322-eng-00000018", "SIP/312-eng,20,rtT") in new stack [May 23 13:12:20] DEBUG[19460] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: Allocating new SIP dialog for 0151b27708baccfd6a69f9c818c45e5d@127.0.0.1:0 - INVITE (No RTP) [May 23 13:12:20] DEBUG[19460] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9f98b48' [May 23 13:12:20] DEBUG[19460] res_rtp_asterisk.c: Allocated port 17682 for RTP instance '0x9f98b48' [May 23 13:12:20] DEBUG[19460] rtp_engine.c: RTP instance '0x9f98b48' is setup and ready to go [May 23 13:12:20] DEBUG[19460] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9f98b48' [May 23 13:12:20] VERBOSE[19460] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Setting NAT on RTP to On [May 23 13:12:20] DEBUG[19460] chan_sip.c: OBPROXY: Not applying OBproxy to this call [May 23 13:12:20] DEBUG[19460] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 13:12:20] DEBUG[19460] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: This channel will not be able to handle video. [May 23 13:12:20] DEBUG[19460] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 13:12:20] DEBUG[19460] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 13:12:20] DEBUG[19460] rtp_engine.c: Seeded SDP of 'SIP/312-eng-0000001a' with that of 'SIP/322-eng-00000018' [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable DIALEDTIME. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable ANSWEREDTIME. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable DIALEDPEERNAME. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable DIALEDPEERNUMBER. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable DIALSTATUS. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable MACRO_DEPTH. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable AVAILCAUSECODE. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable AVAILSTATUS. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable AVAILORIGCHAN. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable AVAILCHAN. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable FOLLOWME. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable FORWARD. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable RECORD. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable DB_RESULT. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable BLOCK. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable CONFIRM. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable SCREEN. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable TIMEOUT. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable CFNAAN. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable CFNAEXT. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable VMT0. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable VM. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable RECORD_CALLER. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable MACRO_PRIORITY. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable MACRO_CONTEXT. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable MACRO_EXTEN. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable MYID. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable MYEXTENSION. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable local. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable zap. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable MY_CHAN. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable RECORD_CALLEE. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable ADDRESS. [May 23 13:12:20] DEBUG[19460] channel.c: Copying soft-transferable variable ORIG_EXTEN_USER. [May 23 13:12:20] DEBUG[19460] channel.c: Copying soft-transferable variable ORIG_EXTEN. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable THISCHAN. [May 23 13:12:20] DEBUG[19460] channel.c: Copying soft-transferable variable CLIMYID. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable THISEXT. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable VM_MBOX. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable PHONE. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable OPTIONS. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable ARG3. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable ARG2. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable ARG1. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable PICKUPMARK. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable DIALED_NUMBER. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable MOH. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable FROM_INSIDE. [May 23 13:12:20] DEBUG[19460] channel.c: Copying hard-transferable variable tenant. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable SIPCALLID. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable SIPDOMAIN. [May 23 13:12:20] DEBUG[19460] channel.c: Not copying variable SIPURI. [May 23 13:12:20] DEBUG[19460] chan_sip.c: Outgoing Call for 312-eng [May 23 13:12:20] DEBUG[19460] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:20] DEBUG[19460] chan_sip.c: Call to peer '312-eng' is 1 out of 2147483647 [May 23 13:12:20] DEBUG[19460] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [May 23 13:12:20] DEBUG[19460] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 13:12:20] VERBOSE[19460] chan_sip.c: Audio is at 5060 [May 23 13:12:20] VERBOSE[19460] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:20] DEBUG[19460] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:20] DEBUG[19460] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 13:12:20] DEBUG[19460] chan_sip.c: Initializing initreq for method INVITE - callid 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 0 [ 72]: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5cd64896;rport [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 3 [ 59]: From: "Poly_test ENG" ;tag=as6dcacad8 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 4 [ 63]: To: [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 5 [ 36]: Contact: [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 6 [ 59]: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:20 GMT [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 12 [ 91]: Remote-Party-ID: "Poly_test ENG" ;party=calling;privacy=off;screen=no [May 23 13:12:20] DEBUG[19460] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [May 23 13:12:20] VERBOSE[19460] chan_sip.c: Reliably Transmitting (NAT) to 209.191.39.117:5060: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5cd64896;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as6dcacad8 To: Contact: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "Poly_test ENG" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 195 v=0 o=root 2012993120 2012993120 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.13 t=0 0 m=audio 17682 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 13:12:20] DEBUG[19460] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042642 [May 23 13:12:20] DEBUG[19460] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:20] VERBOSE[19460] app_dial.c: -- Called SIP/312-eng [May 23 13:12:20] VERBOSE[19460] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfae9d6ffF4B3FF82;received=209.191.39.117 From: "Poly_test ENG";tag=B96595C-F28D6247 To: ;tag=as2c5ec579 Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [May 23 13:12:20] DEBUG[19460] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 14 Application: GotoIf AppData: 0?beenhere Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 15 Application: Set AppData: _ORIG_EXTEN=312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: _ORIG_EXTEN Value: 312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 16 Application: Set AppData: _ORIG_EXTEN_USER=TL/eng-312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: _ORIG_EXTEN_USER Value: TL/eng-312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 17 Application: Macro AppData: tl-notify Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_EXTEN Value: s Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_CONTEXT Value: macro-tl-userexten-base Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_PRIORITY Value: 17 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-notify Extension: s Priority: 1 Application: Set AppData: ADDRESS= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ADDRESS Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-notify Extension: s Priority: 2 Application: UserEvent AppData: TlNotify,dialed: 312|callerID: 322|tenant: eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: UserEvent Privilege: user,all UserEvent: TlNotify dialed: 312|callerID: 322|tenant: eng [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-notify Extension: s Priority: 3 Application: NoOp AppData: TL/eng-312 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-notify Extension: s Priority: 4 Application: GotoIf AppData: 1?s-exit,1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-notify Extension: s-exit Priority: 1 Application: MacroExit AppData: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_EXTEN Value: s Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_CONTEXT Value: macro-tl-userexten Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_PRIORITY Value: 5 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 18 Application: Goto AppData: checkformat Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 20 Application: GotoIf AppData: 1?cont1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DB_RESULT Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 22 Application: Set AppData: RECORD_CALLEE= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: RECORD_CALLEE Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 23 Application: Macro AppData: tl-set-myvariables Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_EXTEN Value: s Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_CONTEXT Value: macro-tl-userexten-base Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_PRIORITY Value: 23 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-myvariables Extension: s Priority: 1 Application: Set AppData: MY_CHAN=322-eng-00000018 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MY_CHAN Value: 322-eng-00000018 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-myvariables Extension: s Priority: 2 Application: NoOp AppData: THECHANNEL=SIP/322-eng-00000018 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-myvariables Extension: s Priority: 3 Application: Set AppData: zap=0 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: zap Value: 0 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-myvariables Extension: s Priority: 4 Application: GotoIf AppData: 1?usechannel Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-myvariables Extension: s Priority: 9 Application: Set AppData: local=0 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: local Value: 0 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-myvariables Extension: s Priority: 10 Application: GotoIf AppData: 1?useit Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-myvariables Extension: s Priority: 12 Application: Set AppData: __MYEXTENSION=322 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __MYEXTENSION Value: 322 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 3 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-set-myvariables Extension: s Priority: 13 Application: Set AppData: __MYID=eng-322 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __MYID Value: eng-322 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_EXTEN Value: s Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_CONTEXT Value: macro-tl-userexten Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_PRIORITY Value: 5 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DB_RESULT Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 24 Application: Set AppData: RECORD_CALLER= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: RECORD_CALLER Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DB_RESULT Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 25 Application: Set AppData: VM=1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: VM Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 26 Application: Set AppData: VMT0= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: VMT0 Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 27 Application: Set AppData: CFNAEXT= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: CFNAEXT Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 28 Application: Set AppData: CFNAAN= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: CFNAAN Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 29 Application: GotoIf AppData: 1?done_checkrecord Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 47 Application: NoOp AppData: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 48 Application: NoOp AppData: RECORD_CALLEE= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 49 Application: NoOp AppData: RECORD_CALLER= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 50 Application: NoOp AppData: OPTIONS=rtT Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 51 Application: NoOp AppData: TOUCH_MONITOR= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 52 Application: GotoIf AppData: 0?next1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 53 Application: Set AppData: TIMEOUT= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: TIMEOUT Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 54 Application: GotoIf AppData: 0?next1 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 55 Application: Set AppData: TIMEOUT=20 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: TIMEOUT Value: 20 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 56 Application: Set AppData: CDR(userfield)=eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 57 Application: Set AppData: SCREEN= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: SCREEN Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 58 Application: Set AppData: CONFIRM= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: CONFIRM Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 59 Application: GotoIf AppData: 1?getblock Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 64 Application: Set AppData: BLOCK= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: BLOCK Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 65 Application: GotoIf AppData: 1?getrecord Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DB_RESULT Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 69 Application: Set AppData: RECORD= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: RECORD Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 70 Application: NoOp AppData: calleridnum=322 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 71 Application: GotoIf AppData: 1?screening Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 103 Application: GotoIf AppData: 1?recording Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 129 Application: GotoIf AppData: 1?forwarding Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 131 Application: Set AppData: FORWARD= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: FORWARD Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 132 Application: GotoIf AppData: 1?followmecheck Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 140 Application: Set AppData: FORWARD= Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: FORWARD Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 141 Application: Set AppData: __FOLLOWME=0 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: __FOLLOWME Value: 0 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 142 Application: GotoIf AppData: 1?checkchannel Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 154 Application: ChanIsAvail AppData: SIP/312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/312-eng-00000019 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 312 CallerIDName: SPA303 Cisco AccountCode: eng Exten: Context: from-inside-eng Uniqueid: 1306170740.25 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000019 Variable: SIPCALLID Value: 6f4c9d455bbb278d3d8622a635ae3420@64.19.145.13:5060 Uniqueid: 1306170740.25 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-00000019 Uniqueid: 1306170740.25 Channeltype: SIP SIPcallid: 6f4c9d455bbb278d3d8622a635ae3420@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-00000019 Channeltype: SIP SIPcallid: 6f4c9d455bbb278d3d8622a635ae3420@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 Peername: 312-eng [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/312-eng-00000019 Uniqueid: 1306170740.25 CallerIDNum: 312 CallerIDName: SPA303 Cisco Cause: 0 Cause-txt: Unknown [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: AVAILCHAN Value: SIP/312-eng-00000019 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: AVAILORIGCHAN Value: SIP/312-eng Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: AVAILSTATUS Value: 135365008 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: AVAILCAUSECODE Value: 0 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 155 Application: GotoIf AppData: 1?chanavail Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 157 Application: GotoIf AppData: 1?dial Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: MACRO_DEPTH Value: 2 Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000018 Context: macro-tl-userexten-base Extension: s Priority: 163 Application: Dial AppData: SIP/312-eng,20,rtT Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALSTATUS Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALEDPEERNUMBER Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALEDPEERNAME Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ANSWEREDTIME Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALEDTIME Value: Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/312-eng-0000001a ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 312 CallerIDName: SPA303 Cisco AccountCode: eng Exten: Context: from-inside-eng Uniqueid: 1306170740.26 [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: SIPCALLID Value: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 Uniqueid: 1306170740.26 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-0000001a Uniqueid: 1306170740.26 Channeltype: SIP SIPcallid: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-0000001a Channeltype: SIP SIPcallid: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 Peername: 312-eng [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: DIALEDPEERNUMBER Value: 312-eng Uniqueid: 1306170740.26 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/322-eng-00000018 Destination: SIP/312-eng-0000001a CallerIDNum: 322 CallerIDName: Poly_test ENG UniqueID: 1306170740.24 DestUniqueID: 1306170740.26 Dialstring: 312-eng [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 8 Paused: 0 [May 23 13:12:20] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 8 Paused: 0 [May 23 13:12:20] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 13:12:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:20] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:20] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 6 (Ringing) [May 23 13:12:20] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '6' [May 23 13:12:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:20] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:20] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 6 (Ringing) [May 23 13:12:20] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '6' [May 23 13:12:20] DEBUG[13070] app_queue.c: Extension '312@local-extensions-eng' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 312 Context: local-extensions-eng Hint: SIP/312-eng Status: 8 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 100 Trying From: "Poly_test ENG";tag=as6dcacad8 To: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK5cd64896 Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as6dcacad8 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 2 [ 63]: To: [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK5cd64896 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 6 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: --- (8 headers 0 lines) --- [May 23 13:12:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 (Checking To) --From tag as6dcacad8 --To-tag [May 23 13:12:20] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1042642 - INVITE (got response) [May 23 13:12:20] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Request 102: Found [May 23 13:12:20] DEBUG[13067] chan_sip.c: SIP response 100 to standard invite [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '6' (Ringing) [May 23 13:12:20] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '6' (Ringing) [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-0000001a~ Value: SIP 100 Trying Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 6 Paused: 0 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 6 Paused: 0 [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 180 Ringing From: "Poly_test ENG";tag=as6dcacad8 To: ;tag=9b31ad4c592f3d91i0 Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK5cd64896 Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as6dcacad8 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 2 [ 86]: To: ;tag=9b31ad4c592f3d91i0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK5cd64896 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 6 [ 82]: Contact: "SPA303 Cisco" [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [May 23 13:12:20] VERBOSE[13067] chan_sip.c: --- (9 headers 0 lines) --- [May 23 13:12:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 (Checking To) --From tag as6dcacad8 --To-tag 9b31ad4c592f3d91i0 [May 23 13:12:20] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Request 102: Found [May 23 13:12:20] DEBUG[13067] chan_sip.c: SIP response 180 to standard invite [May 23 13:12:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:20] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:20] VERBOSE[19460] app_dial.c: -- SIP/312-eng-0000001a is ringing [May 23 13:12:20] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 6 (Ringing) [May 23 13:12:20] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '6' [May 23 13:12:20] DEBUG[19460] rtp_engine.c: Setting early bridge SDP of 'SIP/322-eng-00000018' with that of 'SIP/312-eng-0000001a' [May 23 13:12:20] VERBOSE[19460] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfae9d6ffF4B3FF82;received=209.191.39.117 From: "Poly_test ENG";tag=B96595C-F28D6247 To: ;tag=as2c5ec579 Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [May 23 13:12:20] DEBUG[19460] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/312-eng-0000001a ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 312 CallerIDName: SPA303 Cisco Uniqueid: 1306170740.26 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-0000001a~ Value: SIP 180 Ringing Uniqueid: 1306170740.24 [May 23 13:12:20] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 6 Paused: 0 [May 23 13:12:20] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '6' (Ringing) [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:20] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113876 NOTIFY [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:21] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:21] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:21] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:21] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:21] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 113876, ours 113876) [May 23 13:12:21] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:21] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:21] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:21] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:21] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:21] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:21] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:21] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042634:OPTIONS (Method 3) (No timer T1) [May 23 13:12:21] VERBOSE[13067] chan_sip.c: Retransmitting #2 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK3f742c31;rport Max-Forwards: 70 From: "unknown" ;tag=as4cc71cb7 To: Contact: Call-ID: 08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:21] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:21] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:21] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:21] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042634:OPTIONS (Method 3) (No timer T1) [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Retransmitting #3 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK3f742c31;rport Max-Forwards: 70 From: "unknown" ;tag=as4cc71cb7 To: Contact: Call-ID: 08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:22] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Poly_test ENG";tag=as6dcacad8 To: ;tag=9b31ad4c592f3d91i0 Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK5cd64896 Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Supported: replaces Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Content-Type: application/SDP Content-Length: 214 v=0 o=- 42383342 42383342 IN IP4 209.191.39.117 s=- c=IN IP4 209.191.39.117 t=0 0 m=audio 51822 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as6dcacad8 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 2 [ 86]: To: ;tag=9b31ad4c592f3d91i0 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK5cd64896 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 6 [ 82]: Contact: "SPA303 Cisco" [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Supported: replaces [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 9 [ 61]: Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 10 [ 29]: Content-Type: application/SDP [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Content-Length: 214 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 12 [ 0]: [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 1 [ 43]: o=- 42383342 42383342 IN IP4 209.191.39.117 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 2 [ 3]: s=- [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 5 [ 27]: m=audio 51822 RTP/AVP 0 101 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=ptime:20 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Body 10 [ 10]: a=sendrecv [May 23 13:12:22] VERBOSE[13067] chan_sip.c: --- (12 headers 11 lines) --- [May 23 13:12:22] DEBUG[13067] chan_sip.c: = Looking for Call ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 (Checking To) --From tag as6dcacad8 --To-tag 9b31ad4c592f3d91i0 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Acked pending invite 102 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Stopping retransmission on '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:22] DEBUG[13067] chan_sip.c: SIP response 200 to standard invite [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 42383342 42383342 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [May 23 13:12:22] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 13:12:22] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:22] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Found RTP audio format 101 [May 23 13:12:22] DEBUG[13067] rtp_engine.c: Setting payload 101 based on m type on 0xb7cfd50c [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Found audio description format telephone-event for ID 101 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 13:12:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 13:12:22] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 13:12:22] DEBUG[13067] rtp_engine.c: Incorporating payload 101 on 0xb7cfd50c [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [May 23 13:12:22] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9f98b48' [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51822 [May 23 13:12:22] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0x9f98cf4 [May 23 13:12:22] DEBUG[13067] rtp_engine.c: Copying payload 101 from 0xb7cfd50c to 0x9f98cf4 [May 23 13:12:22] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 13:12:22] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 13:12:22] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:22] DEBUG[13067] chan_sip.c: build_route: Contact hop: "SPA303 Cisco" [May 23 13:12:22] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:22] VERBOSE[13067] chan_sip.c: list_route: hop: [May 23 13:12:22] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:22] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 2 (In use) [May 23 13:12:22] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '2' [May 23 13:12:22] DEBUG[13067] chan_sip.c: Strict routing enforced for session 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:22] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:22] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-0000001a Channeltype: SIP Uniqueid: 1306170740.26 SIPcallid: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 Peername: 312-eng [May 23 13:12:22] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:22] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:22] VERBOSE[13067] chan_sip.c: Transmitting (NAT) to 209.191.39.117:5060: ACK sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4ad057c4;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as6dcacad8 To: ;tag=9b31ad4c592f3d91i0 Contact: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 13:12:22] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:312' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-0000001a~ Value: SIP 200 OK Uniqueid: 1306170740.24 [May 23 13:12:22] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:22] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:22] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 2 (In use) [May 23 13:12:22] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '2' [May 23 13:12:22] VERBOSE[19460] app_dial.c: -- SIP/312-eng-0000001a answered SIP/322-eng-00000018 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/312-eng-0000001a ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 312 CallerIDName: SPA303 Cisco Uniqueid: 1306170740.26 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 2 Paused: 0 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 312 Context: local-extensions-eng Hint: SIP/312-eng Status: 1 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1306170740.24 [May 23 13:12:22] DEBUG[19460] chan_sip.c: SIP answering channel: SIP/322-eng-00000018 [May 23 13:12:22] DEBUG[19460] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:22] DEBUG[19460] chan_sip.c: Setting framing from config on incoming call [May 23 13:12:22] DEBUG[19460] chan_sip.c: ** Our capability: 0x404 (ulaw|ilbc) Video flag: True Text flag: True [May 23 13:12:22] DEBUG[19460] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 13:12:22] VERBOSE[19460] chan_sip.c: Audio is at 5060 [May 23 13:12:22] VERBOSE[19460] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:22] VERBOSE[19460] chan_sip.c: Adding codec 0x400 (ilbc) to SDP [May 23 13:12:22] DEBUG[19460] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:22] DEBUG[19460] chan_sip.c: Done building SDP. Settling with this capability: 0x404 (ulaw|ilbc) [May 23 13:12:22] VERBOSE[19460] chan_sip.c: <--- Reliably Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfae9d6ffF4B3FF82;received=209.191.39.117 From: "Poly_test ENG";tag=B96595C-F28D6247 To: ;tag=as2c5ec579 Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 2146169482 2146169482 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.13 t=0 0 m=audio 18842 RTP/AVP 0 110 a=rtpmap:0 PCMU/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=30 a=ptime:20 a=sendrecv <------------> [May 23 13:12:22] DEBUG[19460] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042647 [May 23 13:12:22] DEBUG[19460] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:22] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:22] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:22] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 13:12:22] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:22] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '2' (In use) [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALEDPEERNAME Value: SIP/312-eng-0000001a Uniqueid: 1306170740.24 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALEDPEERNUMBER Value: 312-eng Uniqueid: 1306170740.24 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: BRIDGEPEER Value: SIP/312-eng-0000001a Uniqueid: 1306170740.24 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: BRIDGEPEER Value: SIP/322-eng-00000018 Uniqueid: 1306170740.26 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/322-eng-00000018 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 322 CallerIDName: Poly_test ENG Uniqueid: 1306170740.24 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/312-eng-0000001a Uniqueid: 1306170740.26 AccountCode: eng OldAccountCode: eng [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:22] DEBUG[19460] features.c: bridge answer set, chan answer set [May 23 13:12:22] DEBUG[19460] features.c: Removing dialed interfaces datastore on SIP/312-eng-0000001a since we're bridging [May 23 13:12:22] DEBUG[19460] channel.c: setting peeraccount to eng for SIP/322-eng-00000018 from data on channel SIP/312-eng-0000001a [May 23 13:12:22] DEBUG[19460] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:22] DEBUG[19460] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:22] VERBOSE[19460] rtp_engine.c: -- Locally bridging SIP/322-eng-00000018 and SIP/312-eng-0000001a [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/322-eng-00000018 Channel2: SIP/312-eng-0000001a Uniqueid1: 1306170740.24 Uniqueid2: 1306170740.26 CallerID1: 322 CallerID2: 312 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: BRIDGEPEER Value: SIP/312-eng-0000001a Uniqueid: 1306170740.24 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: BRIDGEPVTCALLID Value: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 Uniqueid: 1306170740.24 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: BRIDGEPEER Value: SIP/322-eng-00000018 Uniqueid: 1306170740.26 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: BRIDGEPVTCALLID Value: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 Uniqueid: 1306170740.26 [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 2 Paused: 0 [May 23 13:12:22] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '2' (In use) [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 8 Paused: 0 [May 23 13:12:22] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 13:12:22] DEBUG[13070] app_queue.c: Extension '312@local-extensions-eng' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> ACK sip:312@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=B96595C-F28D6247 To: ;tag=as2c5ec579 Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 2 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK33bca99bFEE8FE Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 0 [ 37]: ACK sip:312@64.19.145.13:5060 SIP/2.0 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "Poly_test ENG";tag=B96595C-F28D6247 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 2 [ 52]: To: ;tag=as2c5ec579 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 2 ACK [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 5 [ 65]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK33bca99bFEE8FE [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 13:12:22] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 13:12:22] DEBUG[13067] chan_sip.c: = Looking for Call ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 (Checking From) --From tag B96595C-F28D6247 --To-tag as2c5ec579 [May 23 13:12:22] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 13:12:22] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042647 [May 23 13:12:22] DEBUG[13067] chan_sip.c: Stopping retransmission on 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' of Response 2: Match Found [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' Method: ACK [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' Method: ACK [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:22] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' Method: ACK [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:23] DEBUG[13109] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 64.19.145.7:12009 OurSSRC: 900900610 SentNTP: 1306170743.0029671424 SentRTP: 40160 SentPackets: 251 SentOctets: 40160 ReportBlock: FractionLost: 256 CumulativeLoss: 1 IAJitter: 0.0000 TheirLastSR: 0 DLSR: 65525.9830 (sec) [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113876 NOTIFY [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:23] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:23] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:23] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:23] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:23] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 113876, ours 113876) [May 23 13:12:23] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:23] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:23] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:23] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' Method: ACK [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042634:OPTIONS (Method 3) (No timer T1) [May 23 13:12:23] VERBOSE[13067] chan_sip.c: Retransmitting #4 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK3f742c31;rport Max-Forwards: 70 From: "unknown" ;tag=as4cc71cb7 To: Contact: Call-ID: 08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:23] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:23] DEBUG[13067] chan_sip.c: Destroying SIP dialog 08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060 [May 23 13:12:23] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '08abc63220b9f3393677da2971f4efd2@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' Method: ACK [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:23] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' Method: ACK [May 23 13:12:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' Method: ACK [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 13:12:24] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> REFER sip:Anonymous@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=C5E6782D-50C4FB20 To: "Anonymous";tag=as552f30c6 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 2 REFER Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK73f675f910ED114C Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Refer-To: Referred-By: Content-Length: 0 <-------------> [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 0 [ 45]: REFER sip:Anonymous@64.19.145.13:5060 SIP/2.0 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 1 [102]: From: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 2 [ 58]: To: "Anonymous";tag=as552f30c6 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 4 [ 13]: CSeq: 2 REFER [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK73f675f910ED114C [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 10 [145]: Refer-To: [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 11 [ 39]: Referred-By: [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 13:12:24] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:24] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking From) --From tag C5E6782D-50C4FB20 --To-tag as552f30c6 [May 23 13:12:24] DEBUG[13067] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [May 23 13:12:24] VERBOSE[13067] chan_sip.c: Call 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 got a SIP call transfer from caller: (REFER)! [May 23 13:12:24] DEBUG[13067] chan_sip.c: Attended transfer: Will use Replace-Call-ID : a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 F-tag: B96595C-F28D6247 T-tag: as2c5ec579 [May 23 13:12:24] VERBOSE[13067] chan_sip.c: SIP transfer to extension 312@from-inside-eng by 322-eng@64.19.145.13 [May 23 13:12:24] DEBUG[13067] chan_sip.c: SIP attended transfer: Transferer channel SIP/322-eng-00000016, transferee channel SIP/mg2-00000015 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/mg2-00000015' [May 23 13:12:24] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK73f675f910ED114C;received=209.191.39.117 From: "Poly_test ENG";tag=C5E6782D-50C4FB20 To: "Anonymous";tag=as552f30c6 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 2 REFER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [May 23 13:12:24] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Looking for callid a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 (fromtag B96595C-F28D6247 totag as2c5ec579) [May 23 13:12:24] DEBUG[13067] chan_sip.c: Matched INCOMING call - their tag is B96595C-F28D6247 Our tag is as2c5ec579 [May 23 13:12:24] DEBUG[13067] chan_sip.c: SIP attended transfer: trying to bridge SIP/322-eng-00000018 and SIP/mg2-00000015 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Sip transfer:-------------------- [May 23 13:12:24] DEBUG[13067] chan_sip.c: -- Transferer to PBX channel: SIP/322-eng-00000016 State Up [May 23 13:12:24] DEBUG[13067] chan_sip.c: -- Transferer to PBX second channel (target): SIP/322-eng-00000018 State Up [May 23 13:12:24] DEBUG[13067] chan_sip.c: -- Bridged call to transferee: SIP/mg2-00000015 State Up [May 23 13:12:24] DEBUG[13067] chan_sip.c: -- Bridged call to transfer target: SIP/312-eng-0000001a State Up [May 23 13:12:24] DEBUG[13067] chan_sip.c: -- END Sip transfer:-------------------- [May 23 13:12:24] DEBUG[13067] chan_sip.c: SIP transfer: Four channels to handle [May 23 13:12:24] VERBOSE[13067] res_musiconhold.c: -- Stopped music on hold on SIP/mg2-00000015 [May 23 13:12:24] DEBUG[13067] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 23 13:12:24] DEBUG[13067] chan_sip.c: SIP transfer: trying to masquerade SIP/mg2-00000015 into SIP/322-eng-00000018 [May 23 13:12:24] DEBUG[13067] channel.c: Planning to masquerade channel SIP/mg2-00000015 into the structure of SIP/322-eng-00000018 [May 23 13:12:24] DEBUG[13067] channel.c: Done planning to masquerade channel SIP/mg2-00000015 into the structure of SIP/322-eng-00000018 [May 23 13:12:24] DEBUG[13067] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [May 23 13:12:24] DEBUG[13067] chan_sip.c: Strict routing enforced for session 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:24] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:24] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:24] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:24] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:24] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 209.191.39.117:5060: NOTIFY sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2c4b1882 Max-Forwards: 70 From: "Anonymous";tag=as552f30c6 To: "Poly_test ENG";tag=C5E6782D-50C4FB20 Contact: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 104 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- [May 23 13:12:24] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042650 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:24] DEBUG[13067] channel.c: Set channel SIP/312-eng-0000001a to write format gsm [May 23 13:12:24] DEBUG[13067] channel.c: Thread -1211106416 Blocking 'SIP/312-eng-0000001a', already blocked by thread -1253295216 in procedure ast_waitfor_nandfds [May 23 13:12:24] DEBUG[13067] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [May 23 13:12:24] DEBUG[13067] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [May 23 13:12:24] DEBUG[13067] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x9f98b48' [May 23 13:12:24] DEBUG[13067] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [May 23 13:12:24] VERBOSE[13067] file.c: -- Playing 'beep.gsm' (language 'en') [May 23 13:12:24] DEBUG[19459] rtp_engine.c: Oooh, something is weird, backing out [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: SIPREFERRINGCONTEXT Value: from-inside-eng Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: SIPREFERREDBYHDR Value: Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Attended Channel: SIP/322-eng-00000016 Uniqueid: 1306170731.22 SIP-Callid: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 TargetChannel: SIP/322-eng-00000018 TargetUniqueid: 1306170740.24 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/mg2-00000015 UniqueID: 1306170730.21 [May 23 13:12:24] DEBUG[13067] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 23 13:12:24] DEBUG[13067] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 23 13:12:24] DEBUG[13067] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 23 13:12:24] DEBUG[13067] channel.c: Set channel SIP/312-eng-0000001a to write format ulaw [May 23 13:12:24] DEBUG[19460] channel.c: Actually Masquerading SIP/mg2-00000015(6) into the structure of SIP/322-eng-00000018(6) [May 23 13:12:24] DEBUG[19460] chan_sip.c: SIP Fixup: New owner for dialogue a4a9b786-cc5fd4c1-c80c6254@10.0.15.105: SIP/mg2-00000015 (Old parent: SIP/mg2-00000015) [May 23 13:12:24] DEBUG[19460] chan_sip.c: Hangup call SIP/mg2-00000015, SIP callid a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:24] DEBUG[19460] chan_sip.c: update_call_counter(322-eng) - decrement call limit counter on hangup [May 23 13:12:24] DEBUG[19460] chan_sip.c: Updating call counter for incoming call [May 23 13:12:24] DEBUG[19460] chan_sip.c: Call from peer '322-eng' removed from call limit 2147483647 [May 23 13:12:24] DEBUG[19460] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb7331dd8' [May 23 13:12:24] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:24] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:24] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 13:12:24] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 13:12:24] VERBOSE[19460] chan_sip.c: Scheduling destruction of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' in 32000 ms (Method: ACK) [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/mg2-00000015 CloneState: Up Original: SIP/322-eng-00000018 OriginalState: Up [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/mg2-00000015 Newname: SIP/mg2-00000015 Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/322-eng-00000018 Newname: SIP/mg2-00000015 Uniqueid: 1306170740.24 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOS Value: ssrc=726615324;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=726615324;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 8 Paused: 0 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOS Value: ssrc=1211044864;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=1211044864;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Strict routing enforced for session a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:24] VERBOSE[19460] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:24] DEBUG[19460] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:24] DEBUG[19460] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170730.21 [May 23 13:12:24] VERBOSE[19460] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] VERBOSE[19460] chan_sip.c: Reliably Transmitting (no NAT) to 209.191.39.117:5060: BYE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK6cc0e320 Max-Forwards: 70 From: ;tag=as2c5ec579 To: "Poly_test ENG";tag=B96595C-F28D6247 Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Proxy-Authorization: Digest username="322-eng", realm="asterisk", algorithm=MD5, uri="64.19.145.13", nonce="", response="eac3218b89666699bb97133fa8966982" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 23 13:12:24] DEBUG[19460] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042653 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Trying to put 'BYE sip:322' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOS Value: ssrc=726615324;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/mg2-00000015 Newname: SIP/322-eng-00000018 Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_DEPTH Value: 0 Uniqueid: 1306170740.24 [May 23 13:12:24] DEBUG[19460] channel.c: Putting channel SIP/mg2-00000015 in ulaw/ulaw formats [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/mg2-00000015 CallerIDNum: Anonymous CallerIDName: Anonymous Uniqueid: 1306170740.24 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [May 23 13:12:24] DEBUG[19460] chan_sip.c: SIP Fixup: New owner for dialogue 4eff848341deec190001f2470396b9ea@64.19.145.7: SIP/mg2-00000015 (Old parent: SIP/322-eng-00000018) [May 23 13:12:24] DEBUG[19460] channel.c: Released clone lock on 'SIP/322-eng-00000018' [May 23 13:12:24] DEBUG[19460] channel.c: Done Masquerading SIP/mg2-00000015 (6) [May 23 13:12:24] DEBUG[19460] res_rtp_asterisk.c: Changing ssrc from 1600793530 to 211803779 due to a source change [May 23 13:12:24] DEBUG[19460] res_rtp_asterisk.c: Changing ssrc from 900900610 to 408778772 due to a source change [May 23 13:12:24] DEBUG[19459] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/322-eng-00000018, c1=SIP/322-eng-00000016, flags: Yes,Yes,No,No [May 23 13:12:24] DEBUG[19459] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:24] DEBUG[19459] channel.c: Bridge stops bridging channels SIP/322-eng-00000018 and SIP/322-eng-00000016 [May 23 13:12:24] DEBUG[13067] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:24] DEBUG[19460] rtp_engine.c: rtp-engine-local-bridge: Oooh, something is weird, backing out [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/322-eng-00000018 Channel2: SIP/322-eng-00000016 Uniqueid1: 1306170730.21 Uniqueid2: 1306170731.22 CallerID1: 322 CallerID2: 322 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ANSWEREDTIME Value: 2 Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/mg2-00000015 UniqueID: 1306170740.24 [May 23 13:12:24] DEBUG[13067] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/312-eng-0000001a UniqueID: 1306170740.26 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALEDTIME Value: 4 Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(clid)' (from 'CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '"Anonymous" ' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)}","${CDR(dst)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 8) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'Anonymous' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 8) [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: BRIDGEPEER Value: SIP/312-eng-0000001a Uniqueid: 1306170740.24 [May 23 13:12:24] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Anonymous";tag=as552f30c6 To: "Poly_test ENG";tag=C5E6782D-50C4FB20 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 104 NOTIFY Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2c4b1882 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Length: 0 <-------------> [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 1 [ 60]: From: "Anonymous";tag=as552f30c6 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 104 NOTIFY [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2c4b1882 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 7 [ 17]: Event: refer;id=2 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:24] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:24] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking To) --From tag as552f30c6 --To-tag C5E6782D-50C4FB20 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: BRIDGEPVTCALLID Value: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 Uniqueid: 1306170740.24 [May 23 13:12:24] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042650 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Stopping retransmission on '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' of Request 104: Match Found [May 23 13:12:24] VERBOSE[13067] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [May 23 13:12:24] DEBUG[13067] chan_sip.c: Got 200 OK on NOTIFY for transfer [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 's' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)}","${CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 12) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'SIP/mg2-00000015' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'SIP/322-eng-00000016' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 10) [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: BRIDGEPEER Value: SIP/mg2-00000015 Uniqueid: 1306170740.26 [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '2011-05-23 13:12:10' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 11) [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: BRIDGEPVTCALLID Value: 4eff848341deec190001f2470396b9ea@64.19.145.7 Uniqueid: 1306170740.26 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000016~ Value: SIP 200 OK Uniqueid: 1306170730.21 [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '2011-05-23 13:12:14' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 8) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '2011-05-23 13:12:24' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(duration)' (from 'CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 13) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '14' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(billsec)' (from 'CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 12) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '10' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 16) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'ANSWERED' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 13) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '1306170730.21' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(SIPCALLID1)' (from 'CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '(null)' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(SIPCALLID2)' (from 'CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '(null)' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(CGPN)' (from 'CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '(null)' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(CDPN)' (from 'CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '(null)' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(CHRN)' (from 'CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '(null)' [May 23 13:12:24] DEBUG[19459] pbx.c: Evaluating 'CDR(calltype)' (from 'CDR(calltype)}" ' len 13) [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '(null)' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '2011-05-23 13:12:10' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '"Anonymous" ' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'engtest-eng' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'SIP/mg2-00000015' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'SIP/322-eng-00000016' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'Dial' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'SIP/322-eng&SIP/312-eng' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '14' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '10' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'ANSWERED' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'DOCUMENTATION' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'eng' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '1306170730.21' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is 'eng' [May 23 13:12:24] DEBUG[19459] pbx.c: Function result is '(null)' [May 23 13:12:24] DEBUG[19459] 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 ('2011-05-23 13:12:10','"Anonymous" ','engtest-eng','SIP/mg2-00000015','SIP/322-eng-00000016','Dial','SIP/322-eng&SIP/312-eng','14','10','ANSWERED','DOCUMENTATION','eng','1306170730.21','eng','') [May 23 13:12:24] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> BYE sip:Anonymous@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=C5E6782D-50C4FB20 To: "Anonymous";tag=as552f30c6 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 3 BYE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK8e3b21b75DF1E3FA Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 0 [ 43]: BYE sip:Anonymous@64.19.145.13:5060 SIP/2.0 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 1 [102]: From: "Poly_test ENG";tag=C5E6782D-50C4FB20 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 2 [ 58]: To: "Anonymous";tag=as552f30c6 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 3 BYE [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK8e3b21b75DF1E3FA [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:24] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:24] DEBUG[13067] chan_sip.c: = Looking for Call ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 (Checking From) --From tag C5E6782D-50C4FB20 --To-tag as552f30c6 [May 23 13:12:24] DEBUG[13067] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [May 23 13:12:24] DEBUG[13067] chan_sip.c: Initializing initreq for method BYE - callid 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:24] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:24] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:24] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 13:12:24] DEBUG[13067] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 [May 23 13:12:24] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6646b68' [May 23 13:12:24] VERBOSE[19460] rtp_engine.c: -- Locally bridging SIP/mg2-00000015 and SIP/312-eng-0000001a [May 23 13:12:24] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060' in 32000 ms (Method: BYE) [May 23 13:12:24] DEBUG[13067] chan_sip.c: Received bye, issuing owner hangup [May 23 13:12:24] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK8e3b21b75DF1E3FA;received=209.191.39.117 From: "Poly_test ENG";tag=C5E6782D-50C4FB20 To: "Anonymous";tag=as552f30c6 Call-ID: 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060 CSeq: 3 BYE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:24] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOS Value: ssrc=1211044864;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Strict routing enforced for session 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:24] VERBOSE[19460] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:24] DEBUG[19460] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:24] DEBUG[19460] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:24] VERBOSE[19460] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:24] DEBUG[19460] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 13:12:24] DEBUG[19460] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 13:12:24] VERBOSE[19460] chan_sip.c: Audio is at 5060 [May 23 13:12:24] VERBOSE[19460] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 13:12:24] DEBUG[19460] chan_sip.c: -- Done with adding codecs to SDP [May 23 13:12:24] DEBUG[19460] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 13:12:24] DEBUG[19460] chan_sip.c: Initializing already initialized SIP dialog 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 (presumably reinvite) [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 0 [ 72]: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK477bd026;rport [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 3 [ 59]: From: "Poly_test ENG" ;tag=as6dcacad8 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 4 [ 86]: To: ;tag=9b31ad4c592f3d91i0 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 5 [ 36]: Contact: [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 6 [ 59]: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 10 [ 19]: Supported: replaces [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 11 [ 93]: Remote-Party-ID: "Anonymous" ;party=calling;privacy=off;screen=no [May 23 13:12:24] DEBUG[19460] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 13:12:24] VERBOSE[19460] chan_sip.c: Reliably Transmitting (NAT) to 209.191.39.117:5060: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK477bd026;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as6dcacad8 To: ;tag=9b31ad4c592f3d91i0 Contact: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "Anonymous" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 195 v=0 o=root 2012993120 2012993121 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319997 c=IN IP4 64.19.145.13 t=0 0 m=audio 17682 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[19460] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042655 [May 23 13:12:24] DEBUG[19460] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000016 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170731.22 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:24] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 400 SIP Parser Error : Unexpected '\"', line 9, column 99 From: ;tag=as2c5ec579 To: "Poly_test ENG";tag=B96595C-F28D6247 Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 102 BYE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK6cc0e320 Max-Forwards: 70 User-Agent: Asterisk PBX SVN-branch-1.8-r319997 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Proxy-Authorization: Digest username="322-eng" realm="asterisk" algorithm=MD5 uri="64.19.145.13", nonce="", response="eac3218b89666699bb97133fa8966982" Content-Length: 0 <-------------> [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 0 [ 65]: SIP/2.0 400 SIP Parser Error : Unexpected '\"', line 9, column 99 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 1 [ 54]: From: ;tag=as2c5ec579 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 2 [ 66]: To: "Poly_test ENG";tag=B96595C-F28D6247 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 4 [ 13]: CSeq: 102 BYE [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK6cc0e320 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 7 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 10 [151]: Proxy-Authorization: Digest username="322-eng" realm="asterisk" algorithm=MD5 uri="64.19.145.13", nonce="", response="eac3218b89666699bb97133fa8966982" [May 23 13:12:24] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 13:12:24] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 13:12:24] DEBUG[13067] chan_sip.c: = Looking for Call ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 (Checking To) --From tag as2c5ec579 --To-tag B96595C-F28D6247 [May 23 13:12:24] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042653 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Stopping retransmission on 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' of Request 102: Match Found [May 23 13:12:24] VERBOSE[13067] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [May 23 13:12:24] VERBOSE[13067] chan_sip.c: -- Incoming call: Got SIP response 400 "SIP Parser Error : Unexpected '\"', line 9, column 99" back from 209.191.39.117:5060 [May 23 13:12:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:25] DEBUG[19459] channel.c: Hanging up channel 'SIP/322-eng-00000016' [May 23 13:12:25] DEBUG[19459] chan_sip.c: update_call_counter(322-eng) - decrement call limit counter on hangup [May 23 13:12:25] DEBUG[19459] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:25] DEBUG[19459] chan_sip.c: Call to peer '322-eng' removed from call limit 2147483647 [May 23 13:12:25] DEBUG[19459] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 3776339f22a88f8c47a4d0837cac49d7@64.19.145.13:5060. [May 23 13:12:25] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:25] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:25] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 1 (Not in use) [May 23 13:12:25] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '1' [May 23 13:12:25] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:25] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:25] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 1 (Not in use) [May 23 13:12:25] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '1' [May 23 13:12:25] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:25] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:25] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 1 (Not in use) [May 23 13:12:25] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '1' [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/322-eng-00000016 Uniqueid: 1306170731.22 CallerIDNum: 322 CallerIDName: Poly_test ENG Cause: 16 Cause-txt: Normal Clearing [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000018 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1306170730.21 [May 23 13:12:25] DEBUG[19459] app_dial.c: Exiting with DIALSTATUS=ANSWER. [May 23 13:12:25] DEBUG[19459] pbx.c: Spawn extension (engtest-eng,s,2) exited non-zero on 'SIP/322-eng-00000018' [May 23 13:12:25] VERBOSE[19459] pbx.c: == Spawn extension (engtest-eng, s, 2) exited non-zero on 'SIP/322-eng-00000018' [May 23 13:12:25] DEBUG[19459] channel.c: Soft-Hanging up channel 'SIP/322-eng-00000018' [May 23 13:12:25] DEBUG[19459] channel.c: Hanging up zombie 'SIP/322-eng-00000018' [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/322-eng-00000018 UniqueID: 1306170730.21 DialStatus: ANSWER [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/322-eng-00000018 Uniqueid: 1306170730.21 CallerIDNum: 322 CallerIDName: Poly_test ENG Cause: 16 Cause-txt: Normal Clearing [May 23 13:12:25] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 13:12:25] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 13:12:25] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 1 (Not in use) [May 23 13:12:25] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '1' [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:25] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '1' (Not in use) [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:25] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '1' (Not in use) [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 322 Context: local-extensions-eng Hint: SIP/322-eng Status: 0 [May 23 13:12:25] DEBUG[13070] app_queue.c: Extension '322@local-extensions-eng' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:25] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '1' (Not in use) [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/322-eng MemberName: SIP/322-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:25] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '1' (Not in use) [May 23 13:12:25] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Poly_test ENG";tag=as6dcacad8 To: ;tag=9b31ad4c592f3d91i0 Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK477bd026 Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Content-Type: application/SDP Content-Length: 214 v=0 o=- 42383342 42383343 IN IP4 209.191.39.117 s=- c=IN IP4 209.191.39.117 t=0 0 m=audio 51822 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as6dcacad8 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 2 [ 86]: To: ;tag=9b31ad4c592f3d91i0 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK477bd026 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 6 [ 82]: Contact: "SPA303 Cisco" [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 8 [ 29]: Content-Type: application/SDP [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Content-Length: 214 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Header 10 [ 0]: [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 1 [ 43]: o=- 42383342 42383343 IN IP4 209.191.39.117 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 2 [ 3]: s=- [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 5 [ 27]: m=audio 51822 RTP/AVP 0 101 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=ptime:20 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Body 10 [ 10]: a=sendrecv [May 23 13:12:25] VERBOSE[13067] chan_sip.c: --- (10 headers 11 lines) --- [May 23 13:12:25] DEBUG[13067] chan_sip.c: = Looking for Call ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 (Checking To) --From tag as6dcacad8 --To-tag 9b31ad4c592f3d91i0 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Acked pending invite 103 [May 23 13:12:25] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042655 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Stopping retransmission on '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' of Request 103: Match Found [May 23 13:12:25] DEBUG[13067] chan_sip.c: SIP response 200 to RE-invite on outgoing call 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 42383342 42383343 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [May 23 13:12:25] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 13:12:25] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 13:12:25] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 13:12:25] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 13:12:25] VERBOSE[13067] chan_sip.c: Found RTP audio format 101 [May 23 13:12:25] DEBUG[13067] rtp_engine.c: Setting payload 101 based on m type on 0xb7cfd50c [May 23 13:12:25] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 13:12:25] VERBOSE[13067] chan_sip.c: Found audio description format telephone-event for ID 101 [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 13:12:25] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 13:12:25] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 13:12:25] DEBUG[13067] rtp_engine.c: Incorporating payload 101 on 0xb7cfd50c [May 23 13:12:25] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 13:12:25] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [May 23 13:12:25] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9f98b48' [May 23 13:12:25] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51822 [May 23 13:12:25] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0x9f98cf4 [May 23 13:12:25] DEBUG[13067] rtp_engine.c: Copying payload 101 from 0xb7cfd50c to 0x9f98cf4 [May 23 13:12:25] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 13:12:25] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 13:12:25] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:25] DEBUG[13067] chan_sip.c: Strict routing enforced for session 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:25] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:25] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:25] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:25] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:25] VERBOSE[13067] chan_sip.c: Transmitting (NAT) to 209.191.39.117:5060: ACK sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK32a406e4;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as6dcacad8 To: ;tag=9b31ad4c592f3d91i0 Contact: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 13:12:25] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:312' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: ~HASH~SIP_CAUSE~SIP/312-eng-0000001a~ Value: SIP 200 OK Uniqueid: 1306170740.26 [May 23 13:12:25] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:25] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:25] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 2 (In use) [May 23 13:12:25] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '2' [May 23 13:12:25] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:25] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:25] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 2 Paused: 0 [May 23 13:12:25] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '2' (In use) [May 23 13:12:26] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:26] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:26] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.44.130:5060 ---> OPTIONS sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK0df6cd5a;rport Max-Forwards: 70 From: "asterisk" ;tag=as49387af2 To: Contact: Call-ID: 73ce883c2ef820b17a2fd54a62b8fc9e@209.191.44.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 17:12:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK0df6cd5a;rport [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as49387af2 [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 5 [ 38]: Contact: [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 6 [ 56]: Call-ID: 73ce883c2ef820b17a2fd54a62b8fc9e@209.191.44.130 [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:26 GMT [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 13:12:26] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 13:12:26] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:26] DEBUG[13067] chan_sip.c: = Looking for Call ID: 73ce883c2ef820b17a2fd54a62b8fc9e@209.191.44.130 (Checking From) --From tag as49387af2 --To-tag [May 23 13:12:26] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 13:12:26] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:26] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 73ce883c2ef820b17a2fd54a62b8fc9e@209.191.44.130 - OPTIONS (No RTP) [May 23 13:12:26] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 13:12:26] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 13:12:26] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.44.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK0df6cd5a;rport;received=209.191.44.130 From: "asterisk" ;tag=as49387af2 To: ;tag=as7097ca2e Call-ID: 73ce883c2ef820b17a2fd54a62b8fc9e@209.191.44.130 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> [May 23 13:12:26] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 13:12:26] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '73ce883c2ef820b17a2fd54a62b8fc9e@209.191.44.130' in 32000 ms (Method: OPTIONS) [May 23 13:12:26] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:26] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 046563ea0a1f2d9b170bbe9d66a5cbc7@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:27] DEBUG[13067] acl.c: For destination '64.19.145.18', our source address is '64.19.145.13'. [May 23 13:12:27] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 248f429e27956d644f9d42be04fda643@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.18 SIP/2.0 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1b5c23e4 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as7f6800e3 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 248f429e27956d644f9d42be04fda643@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:27 GMT [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.18:5060: OPTIONS sip:64.19.145.18 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1b5c23e4 Max-Forwards: 70 From: "unknown" ;tag=as7f6800e3 To: Contact: Call-ID: 248f429e27956d644f9d42be04fda643@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042657 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.18:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.18:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1b5c23e4;received=64.19.145.13 From: "unknown" ;tag=as7f6800e3 To: ;tag=as199a97df Call-ID: 248f429e27956d644f9d42be04fda643@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1b5c23e4;received=64.19.145.13 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as7f6800e3 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as199a97df [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 248f429e27956d644f9d42be04fda643@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: = Looking for Call ID: 248f429e27956d644f9d42be04fda643@64.19.145.13:5060 (Checking To) --From tag as7f6800e3 --To-tag as199a97df [May 23 13:12:27] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042657 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Stopping retransmission on '248f429e27956d644f9d42be04fda643@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Destroying SIP dialog 248f429e27956d644f9d42be04fda643@64.19.145.13:5060 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '248f429e27956d644f9d42be04fda643@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:27] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113876 NOTIFY [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:27] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:27] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 113876, ours 113876) [May 23 13:12:27] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:27] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:27] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 1eed927d537bca2375ed06eb0753e922@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:27] DEBUG[13067] acl.c: For destination '64.19.145.15', our source address is '64.19.145.13'. [May 23 13:12:27] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.15 SIP/2.0 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK707e7bc7 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as6b2b4692 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:27 GMT [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.15:5060: OPTIONS sip:64.19.145.15 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK707e7bc7 Max-Forwards: 70 From: "unknown" ;tag=as6b2b4692 To: Contact: Call-ID: 66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042661 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.15:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.15:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK707e7bc7;received=64.19.145.13 From: "unknown" ;tag=as6b2b4692 To: ;tag=as6dc4cb82 Call-ID: 66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK707e7bc7;received=64.19.145.13 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as6b2b4692 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as6dc4cb82 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: = Looking for Call ID: 66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060 (Checking To) --From tag as6b2b4692 --To-tag as6dc4cb82 [May 23 13:12:27] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042661 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Stopping retransmission on '66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Destroying SIP dialog 66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '66be09115277a11708d5d1ae6bcf38f1@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 69f294ab40d8c72c620b37635303100d@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:27] DEBUG[13067] acl.c: For destination '64.19.145.20', our source address is '64.19.145.13'. [May 23 13:12:27] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 03980438187a21db5ae7553e3d50620f@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.20 SIP/2.0 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK198feda5 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as31589ca4 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 03980438187a21db5ae7553e3d50620f@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:27 GMT [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK198feda5 Max-Forwards: 70 From: "unknown" ;tag=as31589ca4 To: Contact: Call-ID: 03980438187a21db5ae7553e3d50620f@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042664 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 56bfe471062662b02459d16f309ab098@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:27] DEBUG[13067] acl.c: For destination '64.19.145.7', our source address is '64.19.145.13'. [May 23 13:12:27] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 31]: OPTIONS sip:64.19.145.7 SIP/2.0 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1d5e1d94 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as0edec457 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 21]: To: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:27 GMT [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: OPTIONS sip:64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1d5e1d94 Max-Forwards: 70 From: "unknown" ;tag=as0edec457 To: Contact: Call-ID: 029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042666 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1d5e1d94;received=64.19.145.13 From: "unknown" ;tag=as0edec457 To: ;tag=as57b8c1d2 Call-ID: 029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1d5e1d94;received=64.19.145.13 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as0edec457 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 36]: To: ;tag=as57b8c1d2 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: = Looking for Call ID: 029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060 (Checking To) --From tag as0edec457 --To-tag as57b8c1d2 [May 23 13:12:27] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042666 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Stopping retransmission on '029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Destroying SIP dialog 029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '029d334a10fa69f64b7ab3f4347b6368@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 1c346fd46a15bf364ae0550b54f6dd6a@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:27] DEBUG[13067] acl.c: For destination '64.19.145.12', our source address is '64.19.145.13'. [May 23 13:12:27] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.12 SIP/2.0 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK373ddbbc [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as0c9d1162 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:27 GMT [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.12:5060: OPTIONS sip:64.19.145.12 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK373ddbbc Max-Forwards: 70 From: "unknown" ;tag=as0c9d1162 To: Contact: Call-ID: 7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042669 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.12:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.12:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK373ddbbc;received=64.19.145.13 From: "unknown" ;tag=as0c9d1162 To: ;tag=as4e716dfc Call-ID: 7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK373ddbbc;received=64.19.145.13 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as0c9d1162 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as4e716dfc [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:27] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:27] DEBUG[13067] chan_sip.c: = Looking for Call ID: 7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060 (Checking To) --From tag as0c9d1162 --To-tag as4e716dfc [May 23 13:12:27] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042669 [May 23 13:12:27] DEBUG[13067] chan_sip.c: Stopping retransmission on '7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:27] DEBUG[13067] chan_sip.c: Destroying SIP dialog 7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060 [May 23 13:12:27] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '7494e2921c14aaaa3a37b1d36ec7354c@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:28] DEBUG[13109] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 64.19.145.7:12009 OurSSRC: 408778772 SentNTP: 1306170748.0032858112 SentRTP: 52160 SentPackets: 326 SentOctets: 52160 ReportBlock: FractionLost: 0 CumulativeLoss: 1 IAJitter: 0.0000 TheirLastSR: 0 DLSR: 65530.9840 (sec) [May 23 13:12:28] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 1fe89f5942748a4f2041853b558a1281@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:28] DEBUG[13067] acl.c: For destination '64.19.145.11', our source address is '64.19.145.13'. [May 23 13:12:28] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.11 SIP/2.0 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4bae9479 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as550aaf99 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:28 GMT [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:28] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.11:5060: OPTIONS sip:64.19.145.11 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4bae9479 Max-Forwards: 70 From: "unknown" ;tag=as550aaf99 To: Contact: Call-ID: 1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:28] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042672 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.11:5060 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:28] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.11:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4bae9479;received=64.19.145.13 From: "unknown" ;tag=as550aaf99 To: ;tag=as452c6697 Call-ID: 1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4bae9479;received=64.19.145.13 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as550aaf99 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as452c6697 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:28] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:28] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:28] DEBUG[13067] chan_sip.c: = Looking for Call ID: 1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060 (Checking To) --From tag as550aaf99 --To-tag as452c6697 [May 23 13:12:28] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042672 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Stopping retransmission on '1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:28] DEBUG[13067] chan_sip.c: Destroying SIP dialog 1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060 [May 23 13:12:28] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '1b8b5ad37707f581795cf43b2018d4ca@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:28] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '16a883360a2e6cf847fb042438148024@209.191.44.130' [May 23 13:12:28] DEBUG[13067] chan_sip.c: Destroying SIP dialog 16a883360a2e6cf847fb042438148024@209.191.44.130 [May 23 13:12:28] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '16a883360a2e6cf847fb042438148024@209.191.44.130' Method: OPTIONS [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:28] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042664:OPTIONS (Method 3) (No timer T1) [May 23 13:12:28] VERBOSE[13067] chan_sip.c: Retransmitting #1 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK198feda5 Max-Forwards: 70 From: "unknown" ;tag=as31589ca4 To: Contact: Call-ID: 03980438187a21db5ae7553e3d50620f@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:28] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:29] DEBUG[19460] res_rtp_asterisk.c: Got RTCP report of 64 bytes [May 23 13:12:29] DEBUG[13109] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 64.19.145.7:12009 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 2278883578 FractionLost: 250 PacketsLost: 54663 HighestSequence: 12501 SequenceNumberCycles: 0 IAJitter: 5 LastSR: 5116.3489660928 DLSR: 1.1110(sec) [May 23 13:12:29] DEBUG[13109] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 209.191.39.117:51823 OurSSRC: 211803779 SentNTP: 1306170749.2147368960 SentRTP: 3520 SentPackets: 22 SentOctets: 3520 ReportBlock: FractionLost: 256 CumulativeLoss: 1 IAJitter: 0.0000 TheirLastSR: 0 DLSR: 65532.5000 (sec) [May 23 13:12:29] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:29] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:29] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042664:OPTIONS (Method 3) (No timer T1) [May 23 13:12:29] VERBOSE[13067] chan_sip.c: Retransmitting #2 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK198feda5 Max-Forwards: 70 From: "unknown" ;tag=as31589ca4 To: Contact: Call-ID: 03980438187a21db5ae7553e3d50620f@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:29] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:29] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:29] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:30] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042664:OPTIONS (Method 3) (No timer T1) [May 23 13:12:30] VERBOSE[13067] chan_sip.c: Retransmitting #3 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK198feda5 Max-Forwards: 70 From: "unknown" ;tag=as31589ca4 To: Contact: Call-ID: 03980438187a21db5ae7553e3d50620f@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:30] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:30] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:30] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:31] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113876 NOTIFY [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:31] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:31] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:31] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:31] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 113876, ours 113876) [May 23 13:12:31] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:31] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:31] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:31] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:31] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:31] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:31] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> BYE sip:312@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=B96595C-F28D6247 To: ;tag=as2c5ec579 Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 3 BYE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK4b18c26fF1BBCE72 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Authorization: Digest username="322-eng",realm="asterisk",nonce="418a2ab7",uri="sip:312@64.19.145.13;user=phone",response="9c928576050fb483456f4d1b41e52382",algorithm=MD5 Content-Length: 0 <-------------> [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 0 [ 37]: BYE sip:312@64.19.145.13:5060 SIP/2.0 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "Poly_test ENG";tag=B96595C-F28D6247 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 2 [ 52]: To: ;tag=as2c5ec579 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 3 BYE [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK4b18c26fF1BBCE72 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 10 [170]: Authorization: Digest username="322-eng",realm="asterisk",nonce="418a2ab7",uri="sip:312@64.19.145.13;user=phone",response="9c928576050fb483456f4d1b41e52382",algorithm=MD5 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 13:12:31] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 13:12:31] DEBUG[13067] chan_sip.c: = Looking for Call ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 (Checking From) --From tag B96595C-F28D6247 --To-tag as2c5ec579 [May 23 13:12:31] DEBUG[13067] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [May 23 13:12:31] DEBUG[13067] chan_sip.c: Initializing initreq for method BYE - callid a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:31] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:31] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:31] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 13:12:31] DEBUG[13067] chan_sip.c: Setting SIP_ALREADYGONE on dialog a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 [May 23 13:12:31] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb7331dd8' [May 23 13:12:31] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'a4a9b786-cc5fd4c1-c80c6254@10.0.15.105' in 32000 ms (Method: BYE) [May 23 13:12:31] DEBUG[13067] chan_sip.c: Received bye, no owner, selfdestruct soon. [May 23 13:12:31] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK4b18c26fF1BBCE72;received=209.191.39.117 From: "Poly_test ENG";tag=B96595C-F28D6247 To: ;tag=as2c5ec579 Call-ID: a4a9b786-cc5fd4c1-c80c6254@10.0.15.105 CSeq: 3 BYE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:31] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:31] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:31] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042664:OPTIONS (Method 3) (No timer T1) [May 23 13:12:31] VERBOSE[13067] chan_sip.c: Retransmitting #4 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK198feda5 Max-Forwards: 70 From: "unknown" ;tag=as31589ca4 To: Contact: Call-ID: 03980438187a21db5ae7553e3d50620f@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:31] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Destroying SIP dialog 03980438187a21db5ae7553e3d50620f@64.19.145.13:5060 [May 23 13:12:31] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '03980438187a21db5ae7553e3d50620f@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:31] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:31] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:31] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> NOTIFY sip:64.19.145.13 SIP/2.0 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 28175 NOTIFY Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-684c3dd7 Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Contact: "SPA303 Cisco" Event: keep-alive Content-Length: 0 <-------------> [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 2 [ 22]: To: [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 3 [ 38]: Call-ID: 5d07fe66-394bec48@10.0.15.101 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 4 [ 18]: CSeq: 28175 NOTIFY [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 5 [ 60]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-684c3dd7 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 7 [ 30]: User-Agent: Cisco/SPA303-7.4.6 [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 8 [ 82]: Contact: "SPA303 Cisco" [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Event: keep-alive [May 23 13:12:31] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:31] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:31] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5d07fe66-394bec48@10.0.15.101 (Checking From) --From tag c7d0e91e95d40f0o0 --To-tag [May 23 13:12:31] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:31] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:31] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-684c3dd7;received=209.191.39.117 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: ;tag=as1046500e Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 28175 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:31] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:31] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '5d07fe66-394bec48@10.0.15.101' in 32000 ms (Method: NOTIFY) [May 23 13:12:31] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:31] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:32] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:32] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:33] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:33] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:33] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 3751c5464163a389187a127507eea1ab@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:33] DEBUG[13067] acl.c: For destination '209.191.13.243', our source address is '64.19.145.13'. [May 23 13:12:33] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 0 [ 46]: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1e6f3e4e;rport [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as4a77c9dd [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 4 [ 36]: To: [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:33 GMT [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:33] VERBOSE[13067] chan_sip.c: Reliably Transmitting (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1e6f3e4e;rport Max-Forwards: 70 From: "unknown" ;tag=as4a77c9dd To: Contact: Call-ID: 53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:33] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042679 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:33] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:33] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 3134229c05958bc91161c9c17dee7003@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:33] DEBUG[13067] acl.c: For destination '64.19.145.4', our source address is '64.19.145.13'. [May 23 13:12:33] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 0 [ 31]: OPTIONS sip:64.19.145.4 SIP/2.0 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4b931263 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as4e85c964 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 4 [ 21]: To: [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:33 GMT [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:33] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.4:5060: OPTIONS sip:64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4b931263 Max-Forwards: 70 From: "unknown" ;tag=as4e85c964 To: Contact: Call-ID: 6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:33] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042681 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.4:5060 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:33] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:33] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.4:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4b931263;received=64.19.145.13 From: "unknown" ;tag=as4e85c964 To: ;tag=as7f86f672 Call-ID: 6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4b931263;received=64.19.145.13 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as4e85c964 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 3 [ 36]: To: ;tag=as7f86f672 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 6 [ 24]: User-Agent: Asterisk PBX [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 7 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Supported: replaces [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:33] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:33] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:33] DEBUG[13067] chan_sip.c: = Looking for Call ID: 6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060 (Checking To) --From tag as4e85c964 --To-tag as7f86f672 [May 23 13:12:33] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042681 [May 23 13:12:33] DEBUG[13067] chan_sip.c: Stopping retransmission on '6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:33] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:33] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:33] DEBUG[13067] chan_sip.c: Destroying SIP dialog 6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060 [May 23 13:12:33] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '6e3595f854ff27b11056abaf4340087f@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:34] DEBUG[19460] res_rtp_asterisk.c: Got RTCP report of 64 bytes [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 64.19.145.7:12009 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 2278883328 FractionLost: 0 PacketsLost: 54663 HighestSequence: 12751 SequenceNumberCycles: 0 IAJitter: 18 LastSR: 5116.3489660928 DLSR: 6.1110(sec) RTT: 1(sec) [May 23 13:12:34] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042679:OPTIONS (Method 3) (No timer T1) [May 23 13:12:34] VERBOSE[13067] chan_sip.c: Retransmitting #1 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1e6f3e4e;rport Max-Forwards: 70 From: "unknown" ;tag=as4a77c9dd To: Contact: Call-ID: 53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:34] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:34] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:34] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: ACK [May 23 13:12:34] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:34] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> BYE sip:7327049020@64.19.145.13:5060 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK5a9278bb;rport Max-Forwards: 70 From: "Anonymous" ;tag=as6423a45f To: ;tag=as60db63bb Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Authorization: Digest username="mg2", realm="asterisk", algorithm=MD5, uri="sip:7327049020@64.19.145.13:5060", nonce="63632ae0", response="79e1ffb3be0f512309a32a87d9e56924" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 0 [ 44]: BYE sip:7327049020@64.19.145.13:5060 SIP/2.0 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK5a9278bb;rport [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as6423a45f [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 4 [ 48]: To: ;tag=as60db63bb [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 5 [ 53]: Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 6 [ 13]: CSeq: 104 BYE [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 7 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 8 [172]: Authorization: Digest username="mg2", realm="asterisk", algorithm=MD5, uri="sip:7327049020@64.19.145.13:5060", nonce="63632ae0", response="79e1ffb3be0f512309a32a87d9e56924" [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 9 [ 39]: X-Asterisk-HangupCause: Normal Clearing [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 10 [ 30]: X-Asterisk-HangupCauseCode: 16 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 13:12:34] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 13:12:34] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 (Checking From) --From tag as6423a45f --To-tag as60db63bb [May 23 13:12:34] DEBUG[13067] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [May 23 13:12:34] DEBUG[13067] chan_sip.c: Initializing initreq for method BYE - callid 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:34] DEBUG[13067] netsock2.c: Splitting '64.19.145.7:5060' gives... [May 23 13:12:34] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '5060'. [May 23 13:12:34] VERBOSE[13067] chan_sip.c: Sending to 64.19.145.7:5060 (no NAT) [May 23 13:12:34] DEBUG[13067] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOS Value: ssrc=408778772;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=326;rlp=54663;rtt=0.001000 Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSBRIDGED Value: ssrc=408778772;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=326;rlp=54663;rtt=0.001000 Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOS Value: ssrc=211803779;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=22;rlp=0;rtt=0.000000 Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=211803779;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=22;rlp=0;rtt=0.000000 Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb71129c8' [May 23 13:12:34] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' in 6400 ms (Method: BYE) [May 23 13:12:34] DEBUG[13067] chan_sip.c: Received bye, issuing owner hangup [May 23 13:12:34] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 64.19.145.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK5a9278bb;received=64.19.145.7;rport=5060 From: "Anonymous" ;tag=as6423a45f To: ;tag=as60db63bb Call-ID: 4eff848341deec190001f2470396b9ea@64.19.145.7 CSeq: 104 BYE Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:34] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:34] DEBUG[19460] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a hangup [May 23 13:12:34] DEBUG[19460] channel.c: Returning from native bridge, channels: SIP/mg2-00000015, SIP/312-eng-0000001a [May 23 13:12:34] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: BYE [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/mg2-00000015 Channel2: SIP/312-eng-0000001a Uniqueid1: 1306170740.24 Uniqueid2: 1306170740.26 CallerID1: Anonymous CallerID2: 312 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ANSWEREDTIME Value: 20 Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALEDTIME Value: 24 Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[19460] pbx.c: Launching 'Hangup' [May 23 13:12:34] VERBOSE[19460] pbx.c: -- Executing [h@from-inside-redir-eng:1] Hangup("SIP/mg2-00000015", "") in new stack [May 23 13:12:34] DEBUG[19460] features.c: Spawn extension (from-inside-redir-eng,h,1) exited non-zero on 'SIP/mg2-00000015' [May 23 13:12:34] VERBOSE[19460] features.c: == Spawn extension (from-inside-redir-eng, h, 1) exited non-zero on 'SIP/mg2-00000015' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(clid)' (from 'CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '"Poly_test ENG" <322>' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)}","${CDR(dst)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 8) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '322' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 8) [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000015 Context: from-inside-redir-eng Extension: h Priority: 1 Application: Hangup AppData: Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '312' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)}","${CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 12) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'SIP/322-eng-00000018' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'SIP/312-eng-0000001a' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 10) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '2011-05-23 13:12:20' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 11) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '2011-05-23 13:12:22' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 8) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '2011-05-23 13:12:34' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(duration)' (from 'CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 13) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '14' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(billsec)' (from 'CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 12) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '12' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 16) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'ANSWERED' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 13) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '1306170740.24' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(SIPCALLID1)' (from 'CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '(null)' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(SIPCALLID2)' (from 'CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '(null)' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(CGPN)' (from 'CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '(null)' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(CDPN)' (from 'CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '(null)' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(CHRN)' (from 'CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '(null)' [May 23 13:12:34] DEBUG[19460] pbx.c: Evaluating 'CDR(calltype)' (from 'CDR(calltype)}" ' len 13) [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '(null)' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '2011-05-23 13:12:20' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '"Poly_test ENG" <322>' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'from-inside-redir-eng' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'SIP/322-eng-00000018' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'SIP/312-eng-0000001a' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'Dial' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'SIP/312-eng,20,rtT' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '14' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '12' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'ANSWERED' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'DOCUMENTATION' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'eng' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '1306170740.24' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is 'eng' [May 23 13:12:34] DEBUG[19460] pbx.c: Function result is '(null)' [May 23 13:12:34] DEBUG[19460] 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 ('2011-05-23 13:12:20','"Poly_test ENG" <322>','from-inside-redir-eng','SIP/322-eng-00000018','SIP/312-eng-0000001a','Dial','SIP/312-eng,20,rtT','14','12','ANSWERED','DOCUMENTATION','eng','1306170740.24','eng','') [May 23 13:12:34] DEBUG[19460] channel.c: Hanging up channel 'SIP/312-eng-0000001a' [May 23 13:12:34] DEBUG[19460] chan_sip.c: Hangup call SIP/312-eng-0000001a, SIP callid 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:34] DEBUG[19460] chan_sip.c: update_call_counter(312-eng) - decrement call limit counter on hangup [May 23 13:12:34] DEBUG[19460] chan_sip.c: Updating call counter for outgoing call [May 23 13:12:34] DEBUG[19460] chan_sip.c: Call to peer '312-eng' removed from call limit 2147483647 [May 23 13:12:34] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:34] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:34] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 1 (Not in use) [May 23 13:12:34] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '1' [May 23 13:12:34] DEBUG[19460] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9f98b48' [May 23 13:12:34] VERBOSE[19460] chan_sip.c: Scheduling destruction of SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' in 32000 ms (Method: INVITE) [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOS Value: ssrc=211803779;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=22;rlp=0;rtt=0.000000 Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-0000001a Variable: RTPAUDIOQOS Value: ssrc=211803779;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=22;rlp=0;rtt=0.000000 Uniqueid: 1306170740.26 [May 23 13:12:34] DEBUG[19460] chan_sip.c: Strict routing enforced for session 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:34] VERBOSE[19460] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 13:12:34] DEBUG[19460] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:34] DEBUG[19460] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:34] VERBOSE[19460] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 13:12:34] VERBOSE[19460] chan_sip.c: Reliably Transmitting (NAT) to 209.191.39.117:5060: BYE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1ddf6448;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as6dcacad8 To: ;tag=9b31ad4c592f3d91i0 Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 312 Context: local-extensions-eng Hint: SIP/312-eng Status: 0 [May 23 13:12:34] DEBUG[19460] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042686 [May 23 13:12:34] DEBUG[19460] chan_sip.c: Trying to put 'BYE sip:312' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/312-eng-0000001a Uniqueid: 1306170740.26 CallerIDNum: 312 CallerIDName: SPA303 Cisco Cause: 16 Cause-txt: Normal Clearing [May 23 13:12:34] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '1' (Not in use) [May 23 13:12:34] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 13:12:34] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 13:12:34] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 1 (Not in use) [May 23 13:12:34] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '1' [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/mg2-00000015 UniqueID: 1306170740.24 DialStatus: ANSWER [May 23 13:12:34] DEBUG[19460] app_dial.c: Exiting with DIALSTATUS=ANSWER. [May 23 13:12:34] DEBUG[19460] app_macro.c: Spawn extension (macro-tl-userexten-base,s,163) exited non-zero on 'SIP/mg2-00000015' in macro 'tl-userexten-base' [May 23 13:12:34] VERBOSE[19460] app_macro.c: == Spawn extension (macro-tl-userexten-base, s, 163) exited non-zero on 'SIP/mg2-00000015' in macro 'tl-userexten-base' [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ARG1 Value: SIP/312-eng Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ARG2 Value: 312@default-eng Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: ARG3 Value: Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_EXTEN Value: 312 Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_CONTEXT Value: from-inside-redir-eng Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_PRIORITY Value: 1 Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[19460] app_macro.c: Spawn extension (macro-tl-userexten,s,5) exited non-zero on 'SIP/mg2-00000015' in macro 'tl-userexten' [May 23 13:12:34] VERBOSE[19460] app_macro.c: == Spawn extension (macro-tl-userexten, s, 5) exited non-zero on 'SIP/mg2-00000015' in macro 'tl-userexten' [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000015 Variable: MACRO_DEPTH Value: 0 Uniqueid: 1306170740.24 [May 23 13:12:34] DEBUG[19460] pbx.c: Spawn extension (from-inside-redir-eng,312,1) exited non-zero on 'SIP/mg2-00000015' [May 23 13:12:34] VERBOSE[19460] pbx.c: == Spawn extension (from-inside-redir-eng, 312, 1) exited non-zero on 'SIP/mg2-00000015' [May 23 13:12:34] DEBUG[19460] channel.c: Soft-Hanging up channel 'SIP/mg2-00000015' [May 23 13:12:34] DEBUG[19460] channel.c: Hanging up channel 'SIP/mg2-00000015' [May 23 13:12:34] DEBUG[19460] chan_sip.c: Hangup call SIP/mg2-00000015, SIP callid 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:34] DEBUG[19460] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb71129c8' [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/mg2-00000015 Uniqueid: 1306170740.24 CallerIDNum: Anonymous CallerIDName: Anonymous Cause: 16 Cause-txt: Normal Clearing [May 23 13:12:34] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - mg2 [May 23 13:12:34] DEBUG[13069] chan_sip.c: Checking device state for peer mg2 [May 23 13:12:34] DEBUG[13069] devicestate.c: Changing state for SIP/mg2 - state 1 (Not in use) [May 23 13:12:34] DEBUG[13069] devicestate.c: device 'SIP/mg2' state '1' [May 23 13:12:34] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/312-eng MemberName: SIP/312-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:34] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '1' (Not in use) [May 23 13:12:34] DEBUG[13070] app_queue.c: Extension '312@local-extensions-eng' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 23 13:12:34] DEBUG[13094] app_queue.c: Device 'SIP/mg2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 23 13:12:34] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Poly_test ENG";tag=as6dcacad8 To: ;tag=9b31ad4c592f3d91i0 Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 CSeq: 104 BYE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK1ddf6448 Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as6dcacad8 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 2 [ 86]: To: ;tag=9b31ad4c592f3d91i0 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 4 [ 13]: CSeq: 104 BYE [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK1ddf6448 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 6 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [May 23 13:12:34] VERBOSE[13067] chan_sip.c: --- (8 headers 0 lines) --- [May 23 13:12:34] DEBUG[13067] chan_sip.c: = Looking for Call ID: 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 (Checking To) --From tag as6dcacad8 --To-tag 9b31ad4c592f3d91i0 [May 23 13:12:34] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042686 [May 23 13:12:34] DEBUG[13067] chan_sip.c: Stopping retransmission on '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' of Request 104: Match Found [May 23 13:12:34] DEBUG[13067] chan_sip.c: Destroying SIP dialog 6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060 [May 23 13:12:34] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '6adf518c35e7be7d17ab1ce774ec309c@64.19.145.13:5060' Method: INVITE [May 23 13:12:34] DEBUG[13067] rtp_engine.c: Destroyed RTP instance '0x9f98b48' [May 23 13:12:35] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> NOTIFY sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 113876 NOTIFY [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:35] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:35] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:35] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:35] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 113876, ours 113876) [May 23 13:12:35] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 13:12:35] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.13.243:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-250938db;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 113876 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:35] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:12:35] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 13:12:35] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042679:OPTIONS (Method 3) (No timer T1) [May 23 13:12:35] VERBOSE[13067] chan_sip.c: Retransmitting #2 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1e6f3e4e;rport Max-Forwards: 70 From: "unknown" ;tag=as4a77c9dd To: Contact: Call-ID: 53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:35] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:35] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> REGISTER sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-c158d466 From: ;tag=5e35c995200173e1o3 To: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94306 REGISTER Max-Forwards: 70 Authorization: Digest username="175-eng",realm="asterisk",nonce="2d5fbf24",uri="sip:64.19.145.13",algorithm=MD5,response="6b3ef8867b6e7df7dae858c862a04f42" Contact: ;expires=3600 User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 0 [ 33]: REGISTER sip:64.19.145.13 SIP/2.0 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-c158d466 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 3 [ 30]: To: [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 5 [ 20]: CSeq: 94306 REGISTER [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 7 [155]: Authorization: Digest username="175-eng",realm="asterisk",nonce="2d5fbf24",uri="sip:64.19.145.13",algorithm=MD5,response="6b3ef8867b6e7df7dae858c862a04f42" [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 8 [ 55]: Contact: ;expires=3600 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 11 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 12 [ 19]: Supported: replaces [May 23 13:12:35] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:35] DEBUG[13067] chan_sip.c: = Looking for Call ID: b7b02bc6-e8d28b72@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:35] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:35] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:35] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:35] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:35] DEBUG[13067] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [May 23 13:12:35] DEBUG[13067] chan_sip.c: Initializing initreq for method REGISTER - callid b7b02bc6-e8d28b72@192.168.15.187 [May 23 13:12:35] DEBUG[13067] netsock2.c: Splitting '192.168.15.187:5063' gives... [May 23 13:12:35] DEBUG[13067] netsock2.c: ...host '192.168.15.187' and port '5063'. [May 23 13:12:35] VERBOSE[13067] chan_sip.c: Sending to 209.191.13.243:17616 (NAT) [May 23 13:12:35] NOTICE[13067] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=5e35c995200173e1o3' [May 23 13:12:35] VERBOSE[13067] chan_sip.c: <--- Transmitting (NAT) to 209.191.13.243:17616 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-c158d466;received=209.191.13.243;rport=17616 From: ;tag=5e35c995200173e1o3 To: ;tag=as1f6b40e0 Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94306 REGISTER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79d71926", stale=true Content-Length: 0 <------------> [May 23 13:12:35] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 209.191.13.243:17616 [May 23 13:12:35] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'b7b02bc6-e8d28b72@192.168.15.187' in 32000 ms (Method: REGISTER) [May 23 13:12:35] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.13.243:17616 ---> REGISTER sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-1abea94f From: ;tag=5e35c995200173e1o3 To: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94307 REGISTER Max-Forwards: 70 Authorization: Digest username="175-eng",realm="asterisk",nonce="79d71926",uri="sip:64.19.145.13",algorithm=MD5,response="40e4c8e0d34902ecc8ba7f57a3b3baa1" Contact: ;expires=3600 User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 0 [ 33]: REGISTER sip:64.19.145.13 SIP/2.0 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-1abea94f [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 3 [ 30]: To: [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 5 [ 20]: CSeq: 94307 REGISTER [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 7 [155]: Authorization: Digest username="175-eng",realm="asterisk",nonce="79d71926",uri="sip:64.19.145.13",algorithm=MD5,response="40e4c8e0d34902ecc8ba7f57a3b3baa1" [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 8 [ 55]: Contact: ;expires=3600 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 11 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [May 23 13:12:35] DEBUG[13067] chan_sip.c: Header 12 [ 19]: Supported: replaces [May 23 13:12:35] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:35] DEBUG[13067] chan_sip.c: = Looking for Call ID: b7b02bc6-e8d28b72@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:35] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:35] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:35] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 13:12:35] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 13:12:35] DEBUG[13067] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [May 23 13:12:35] DEBUG[13067] chan_sip.c: Initializing initreq for method REGISTER - callid b7b02bc6-e8d28b72@192.168.15.187 [May 23 13:12:35] DEBUG[13067] netsock2.c: Splitting '192.168.15.187:5063' gives... [May 23 13:12:35] DEBUG[13067] netsock2.c: ...host '192.168.15.187' and port '5063'. [May 23 13:12:35] VERBOSE[13067] chan_sip.c: Sending to 209.191.13.243:17616 (NAT) [May 23 13:12:35] DEBUG[13067] chan_sip.c: Store REGISTER's src-IP:port for call routing. [May 23 13:12:35] DEBUG[13109] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/175-eng PeerStatus: Registered Address: 209.191.13.243:17616 [May 23 13:12:35] VERBOSE[13067] chan_sip.c: <--- Transmitting (NAT) to 209.191.13.243:17616 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-1abea94f;received=209.191.13.243;rport=17616 From: ;tag=5e35c995200173e1o3 To: ;tag=as1f6b40e0 Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94307 REGISTER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Mon, 23 May 2011 17:12:35 GMT Content-Length: 0 <------------> [May 23 13:12:35] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:17616 [May 23 13:12:35] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'b7b02bc6-e8d28b72@192.168.15.187' in 32000 ms (Method: REGISTER) [May 23 13:12:35] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 175-eng [May 23 13:12:35] DEBUG[13069] chan_sip.c: Checking device state for peer 175-eng [May 23 13:12:35] DEBUG[13069] devicestate.c: Changing state for SIP/175-eng - state 1 (Not in use) [May 23 13:12:35] DEBUG[13069] devicestate.c: device 'SIP/175-eng' state '1' [May 23 13:12:35] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: test-eng Location: SIP/175-eng MemberName: SIP/175-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:35] DEBUG[13109] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: supporthotline-eng Location: SIP/175-eng MemberName: SIP/175-eng Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 [May 23 13:12:35] DEBUG[13094] app_queue.c: Device 'SIP/175-eng' changed to state '1' (Not in use) [May 23 13:12:36] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> REGISTER sip:fsdev.monmouth.com:5060 SIP/2.0 From: ;tag=266a948-0-13c4-1ad5cc-1c9c782-1ad5cc To: Call-ID: 269f250-0-13c4-62-72b2d122-62 CSeq: 15049 REGISTER Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1ad5cc-68d3253d-4b444437 Max-Forwards: 70 Supported: 100rel,replaces Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER User-Agent: ADTRAN_Total_Access_908_2nd_Gen/A4.02.00.E Contact: Expires: 3600 Content-Length: 0 <-------------> [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 0 [ 44]: REGISTER sip:fsdev.monmouth.com:5060 SIP/2.0 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 1 [102]: From: ;tag=266a948-0-13c4-1ad5cc-1c9c782-1ad5cc [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 2 [ 59]: To: [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 3 [ 38]: Call-ID: 269f250-0-13c4-62-72b2d122-62 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 4 [ 20]: CSeq: 15049 REGISTER [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 5 [ 76]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1ad5cc-68d3253d-4b444437 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Supported: 100rel,replaces [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 8 [ 78]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 9 [ 54]: User-Agent: ADTRAN_Total_Access_908_2nd_Gen/A4.02.00.E [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 10 [ 60]: Contact: [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 11 [ 13]: Expires: 3600 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 13:12:36] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:36] DEBUG[13067] chan_sip.c: = Looking for Call ID: 269f250-0-13c4-62-72b2d122-62 (Checking From) --From tag 266a948-0-13c4-1ad5cc-1c9c782-1ad5cc --To-tag [May 23 13:12:36] DEBUG[13067] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 13:12:36] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 269f250-0-13c4-62-72b2d122-62 - REGISTER (No RTP) [May 23 13:12:36] DEBUG[13067] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [May 23 13:12:36] DEBUG[13067] chan_sip.c: Initializing initreq for method REGISTER - callid 269f250-0-13c4-62-72b2d122-62 [May 23 13:12:36] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:36] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:36] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 13:12:36] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1ad5cc-68d3253d-4b444437;received=209.191.39.117 From: ;tag=266a948-0-13c4-1ad5cc-1c9c782-1ad5cc To: Call-ID: 269f250-0-13c4-62-72b2d122-62 CSeq: 15049 REGISTER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:36] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:36] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1ad5cc-68d3253d-4b444437;received=209.191.39.117 From: ;tag=266a948-0-13c4-1ad5cc-1c9c782-1ad5cc To: ;tag=as4ede2651 Call-ID: 269f250-0-13c4-62-72b2d122-62 CSeq: 15049 REGISTER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3602de9f" Content-Length: 0 <------------> [May 23 13:12:36] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:36] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '269f250-0-13c4-62-72b2d122-62' in 32000 ms (Method: REGISTER) [May 23 13:12:36] NOTICE[13067] chan_sip.c: Registration from '' failed for '209.191.39.117:5060' - No matching peer found [May 23 13:12:36] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '269f250-0-13c4-62-72b2d122-62' in 32000 ms (Method: REGISTER) [May 23 13:12:36] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> REGISTER sip:fsdev.monmouth.com:5060 SIP/2.0 From: ;tag=266a948-0-13c4-1ad5cc-1c9c782-1ad5cc To: Call-ID: 269f250-0-13c4-62-72b2d122-62 CSeq: 15050 REGISTER Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1ad5cc-68d32553-7c376f87 Max-Forwards: 70 Supported: 100rel,replaces Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER User-Agent: ADTRAN_Total_Access_908_2nd_Gen/A4.02.00.E Contact: Expires: 3600 Authorization: Digest username="M7327049020",realm="asterisk",nonce="3602de9f",uri="sip:fsdev.monmouth.com:5060",response="d9fa7976bd5d46e82aac4974d935b6c1",algorithm=MD5 Content-Length: 0 <-------------> [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 0 [ 44]: REGISTER sip:fsdev.monmouth.com:5060 SIP/2.0 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 1 [102]: From: ;tag=266a948-0-13c4-1ad5cc-1c9c782-1ad5cc [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 2 [ 59]: To: [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 3 [ 38]: Call-ID: 269f250-0-13c4-62-72b2d122-62 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 4 [ 20]: CSeq: 15050 REGISTER [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 5 [ 76]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1ad5cc-68d32553-7c376f87 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Supported: 100rel,replaces [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 8 [ 78]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 9 [ 54]: User-Agent: ADTRAN_Total_Access_908_2nd_Gen/A4.02.00.E [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 10 [ 60]: Contact: [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 11 [ 13]: Expires: 3600 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 12 [170]: Authorization: Digest username="M7327049020",realm="asterisk",nonce="3602de9f",uri="sip:fsdev.monmouth.com:5060",response="d9fa7976bd5d46e82aac4974d935b6c1",algorithm=MD5 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [May 23 13:12:36] VERBOSE[13067] chan_sip.c: --- (14 headers 0 lines) --- [May 23 13:12:36] DEBUG[13067] chan_sip.c: = Looking for Call ID: 269f250-0-13c4-62-72b2d122-62 (Checking From) --From tag 266a948-0-13c4-1ad5cc-1c9c782-1ad5cc --To-tag [May 23 13:12:36] DEBUG[13067] netsock2.c: Splitting 'fsdev.monmouth.com:5060' gives... [May 23 13:12:36] DEBUG[13067] netsock2.c: ...host 'fsdev.monmouth.com' and port '5060'. [May 23 13:12:36] DEBUG[13067] netsock2.c: Splitting 'fsdev.monmouth.com:5060' gives... [May 23 13:12:36] DEBUG[13067] netsock2.c: ...host 'fsdev.monmouth.com' and port '5060'. [May 23 13:12:36] DEBUG[13067] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [May 23 13:12:36] DEBUG[13067] chan_sip.c: Initializing initreq for method REGISTER - callid 269f250-0-13c4-62-72b2d122-62 [May 23 13:12:36] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 13:12:36] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 13:12:36] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 13:12:36] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1ad5cc-68d32553-7c376f87;received=209.191.39.117 From: ;tag=266a948-0-13c4-1ad5cc-1c9c782-1ad5cc To: Call-ID: 269f250-0-13c4-62-72b2d122-62 CSeq: 15050 REGISTER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:36] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:36] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 403 Forbidden (Bad auth) Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1ad5cc-68d32553-7c376f87;received=209.191.39.117 From: ;tag=266a948-0-13c4-1ad5cc-1c9c782-1ad5cc To: ;tag=as4ede2651 Call-ID: 269f250-0-13c4-62-72b2d122-62 CSeq: 15050 REGISTER Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [May 23 13:12:36] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 403' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:36] NOTICE[13067] chan_sip.c: Registration from '' failed for '209.191.39.117:5060' - No matching peer found [May 23 13:12:36] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '269f250-0-13c4-62-72b2d122-62' in 32000 ms (Method: REGISTER) [May 23 13:12:36] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042679:OPTIONS (Method 3) (No timer T1) [May 23 13:12:36] VERBOSE[13067] chan_sip.c: Retransmitting #3 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1e6f3e4e;rport Max-Forwards: 70 From: "unknown" ;tag=as4a77c9dd To: Contact: Call-ID: 53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:36] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:36] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.44.130:5060 ---> OPTIONS sip:64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK66d775f8;rport Max-Forwards: 70 From: "asterisk" ;tag=as431109ff To: Contact: Call-ID: 21295cd456271d4c0c7b66d077d4cb38@209.191.44.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 17:12:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK66d775f8;rport [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as431109ff [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 5 [ 38]: Contact: [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 6 [ 56]: Call-ID: 21295cd456271d4c0c7b66d077d4cb38@209.191.44.130 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:36 GMT [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 13:12:36] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 13:12:36] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 13:12:36] DEBUG[13067] chan_sip.c: = Looking for Call ID: 21295cd456271d4c0c7b66d077d4cb38@209.191.44.130 (Checking From) --From tag as431109ff --To-tag [May 23 13:12:36] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 13:12:36] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:36] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 21295cd456271d4c0c7b66d077d4cb38@209.191.44.130 - OPTIONS (No RTP) [May 23 13:12:36] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 13:12:36] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 13:12:36] VERBOSE[13067] chan_sip.c: <--- Transmitting (no NAT) to 209.191.44.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK66d775f8;rport;received=209.191.44.130 From: "asterisk" ;tag=as431109ff To: ;tag=as053e33c4 Call-ID: 21295cd456271d4c0c7b66d077d4cb38@209.191.44.130 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.8-r319997 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> [May 23 13:12:36] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 13:12:36] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '21295cd456271d4c0c7b66d077d4cb38@209.191.44.130' in 32000 ms (Method: OPTIONS) [May 23 13:12:37] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 0535146e601a24264178a47a0b38ac72@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:37] DEBUG[13067] acl.c: For destination '64.19.145.18', our source address is '64.19.145.13'. [May 23 13:12:37] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.18 SIP/2.0 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK27103082 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as15766684 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:37 GMT [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.18:5060: OPTIONS sip:64.19.145.18 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK27103082 Max-Forwards: 70 From: "unknown" ;tag=as15766684 To: Contact: Call-ID: 0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042695 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.18:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.18:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK27103082;received=64.19.145.13 From: "unknown" ;tag=as15766684 To: ;tag=as358523f9 Call-ID: 0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK27103082;received=64.19.145.13 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as15766684 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as358523f9 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060 (Checking To) --From tag as15766684 --To-tag as358523f9 [May 23 13:12:37] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042695 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Stopping retransmission on '0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:37] DEBUG[13067] chan_sip.c: Destroying SIP dialog 0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '0d4b8c113ecc1d532631b9641c0ef984@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042679:OPTIONS (Method 3) (No timer T1) [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Retransmitting #4 (NAT) to 209.191.13.243:26300: OPTIONS sip:rjiang@192.168.15.176:5063 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1e6f3e4e;rport Max-Forwards: 70 From: "unknown" ;tag=as4a77c9dd To: Contact: Call-ID: 53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Destroying SIP dialog 53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '53ed4a863b3d0cf16f3f901b5413282d@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 39f158c770a037a2231dbae214c0f927@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:37] DEBUG[13067] acl.c: For destination '64.19.145.15', our source address is '64.19.145.13'. [May 23 13:12:37] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.15 SIP/2.0 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK493ffb1f [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as5ff673ee [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:37 GMT [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.15:5060: OPTIONS sip:64.19.145.15 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK493ffb1f Max-Forwards: 70 From: "unknown" ;tag=as5ff673ee To: Contact: Call-ID: 2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042699 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.15:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.15:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK493ffb1f;received=64.19.145.13 From: "unknown" ;tag=as5ff673ee To: ;tag=as74c821ae Call-ID: 2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK493ffb1f;received=64.19.145.13 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as5ff673ee [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as74c821ae [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: = Looking for Call ID: 2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060 (Checking To) --From tag as5ff673ee --To-tag as74c821ae [May 23 13:12:37] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042699 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Stopping retransmission on '2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:37] DEBUG[13067] chan_sip.c: Destroying SIP dialog 2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '2812d9a857a52c76493a1166396beb2e@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 38ca28701385f7731777357f5d9a15fd@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:37] DEBUG[13067] acl.c: For destination '64.19.145.7', our source address is '64.19.145.13'. [May 23 13:12:37] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 0 [ 31]: OPTIONS sip:64.19.145.7 SIP/2.0 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK44bfd289 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as3ffe6ad7 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 4 [ 21]: To: [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:37 GMT [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: OPTIONS sip:64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK44bfd289 Max-Forwards: 70 From: "unknown" ;tag=as3ffe6ad7 To: Contact: Call-ID: 4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042702 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.7:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK44bfd289;received=64.19.145.13 From: "unknown" ;tag=as3ffe6ad7 To: ;tag=as61904c15 Call-ID: 4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK44bfd289;received=64.19.145.13 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as3ffe6ad7 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 3 [ 36]: To: ;tag=as61904c15 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: = Looking for Call ID: 4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060 (Checking To) --From tag as3ffe6ad7 --To-tag as61904c15 [May 23 13:12:37] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042702 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Stopping retransmission on '4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:37] DEBUG[13067] chan_sip.c: Destroying SIP dialog 4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '4a957c6d74e9a3352d739c192593b81c@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 7b69b1fb18be099e661aabfc614ae586@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:37] DEBUG[13067] acl.c: For destination '64.19.145.12', our source address is '64.19.145.13'. [May 23 13:12:37] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.12 SIP/2.0 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK70bcd34b [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as024f5123 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:37 GMT [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.12:5060: OPTIONS sip:64.19.145.12 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK70bcd34b Max-Forwards: 70 From: "unknown" ;tag=as024f5123 To: Contact: Call-ID: 599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042705 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.12:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.12:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK70bcd34b;received=64.19.145.13 From: "unknown" ;tag=as024f5123 To: ;tag=as7689e550 Call-ID: 599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK70bcd34b;received=64.19.145.13 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as024f5123 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as7689e550 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:37] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:37] DEBUG[13067] chan_sip.c: = Looking for Call ID: 599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060 (Checking To) --From tag as024f5123 --To-tag as7689e550 [May 23 13:12:37] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042705 [May 23 13:12:37] DEBUG[13067] chan_sip.c: Stopping retransmission on '599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:37] DEBUG[13067] chan_sip.c: Destroying SIP dialog 599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060 [May 23 13:12:37] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '599ff9e64fe2336d1b2a6ebe39d72940@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:38] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 570e79d9109ab7e04e824dd92b9b2b19@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:38] DEBUG[13067] acl.c: For destination '64.19.145.11', our source address is '64.19.145.13'. [May 23 13:12:38] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.11 SIP/2.0 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5d29e181 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as0496616f [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:38 GMT [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:38] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.11:5060: OPTIONS sip:64.19.145.11 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5d29e181 Max-Forwards: 70 From: "unknown" ;tag=as0496616f To: Contact: Call-ID: 60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:38] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042708 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.11:5060 [May 23 13:12:38] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:64.19.145.11:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5d29e181;received=64.19.145.13 From: "unknown" ;tag=as0496616f To: ;tag=as205511a6 Call-ID: 60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060 CSeq: 102 OPTIONS Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5d29e181;received=64.19.145.13 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 2 [ 57]: From: "unknown" ;tag=as0496616f [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 3 [ 37]: To: ;tag=as205511a6 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 4 [ 59]: Call-ID: 60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [May 23 13:12:38] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 13:12:38] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 13:12:38] DEBUG[13067] chan_sip.c: = Looking for Call ID: 60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060 (Checking To) --From tag as0496616f --To-tag as205511a6 [May 23 13:12:38] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1042708 [May 23 13:12:38] DEBUG[13067] chan_sip.c: Stopping retransmission on '60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:38] DEBUG[13067] chan_sip.c: Destroying SIP dialog 60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060 [May 23 13:12:38] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '60c3611d46337d570e80344259aa1e7e@64.19.145.13:5060' Method: OPTIONS [May 23 13:12:38] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '1a4ec89b671a3caa051af79e59e921e6@209.191.44.130' [May 23 13:12:38] DEBUG[13067] chan_sip.c: Destroying SIP dialog 1a4ec89b671a3caa051af79e59e921e6@209.191.44.130 [May 23 13:12:38] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '1a4ec89b671a3caa051af79e59e921e6@209.191.44.130' Method: OPTIONS [May 23 13:12:40] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' [May 23 13:12:40] DEBUG[13067] chan_sip.c: Destroying SIP dialog 4eff848341deec190001f2470396b9ea@64.19.145.7 [May 23 13:12:40] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '4eff848341deec190001f2470396b9ea@64.19.145.7' Method: BYE [May 23 13:12:40] DEBUG[13067] rtp_engine.c: Destroyed RTP instance '0xb71129c8' [May 23 13:12:41] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 69e46e2d11dacb8024a36e9b03698f38@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:41] DEBUG[13067] acl.c: For destination '64.19.145.20', our source address is '64.19.145.13'. [May 23 13:12:41] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:41] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 24755a0a79484d1f13111871427a9bd5@64.19.145.13:5060 [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.20 SIP/2.0 [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK35174c07 [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 3 [ 57]: From: "unknown" ;tag=as3650b4ea [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 5 [ 40]: Contact: [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 6 [ 59]: Call-ID: 24755a0a79484d1f13111871427a9bd5@64.19.145.13:5060 [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 17:12:41 GMT [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 13:12:41] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 13:12:41] VERBOSE[13067] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK35174c07 Max-Forwards: 70 From: "unknown" ;tag=as3650b4ea To: Contact: Call-ID: 24755a0a79484d1f13111871427a9bd5@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:41] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1042711 [May 23 13:12:41] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:42] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4' [May 23 13:12:42] DEBUG[13067] chan_sip.c: Destroying SIP dialog 1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4 [May 23 13:12:42] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '1cf3f14f6b6eef50373128de63b5cd31@64.19.145.4' Method: OPTIONS [May 23 13:12:42] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042711:OPTIONS (Method 3) (No timer T1) [May 23 13:12:42] VERBOSE[13067] chan_sip.c: Retransmitting #1 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK35174c07 Max-Forwards: 70 From: "unknown" ;tag=as3650b4ea To: Contact: Call-ID: 24755a0a79484d1f13111871427a9bd5@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:42] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:43] DEBUG[13067] chan_sip.c: SIP TIMER: Not rescheduling id #1042711:OPTIONS (Method 3) (No timer T1) [May 23 13:12:43] VERBOSE[13067] chan_sip.c: Retransmitting #2 (no NAT) to 64.19.145.20:5060: OPTIONS sip:64.19.145.20 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK35174c07 Max-Forwards: 70 From: "unknown" ;tag=as3650b4ea To: Contact: Call-ID: 24755a0a79484d1f13111871427a9bd5@64.19.145.13:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 17:12:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- [May 23 13:12:43] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:44] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:45] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.20:5060 [May 23 13:12:45] DEBUG[13067] chan_sip.c: Destroying SIP dialog 24755a0a79484d1f13111871427a9bd5@64.19.145.13:5060 [May 23 13:12:46] DEBUG[13067] chan_sip.c: = Looking for Call ID: 1213f62f16db832f7755ab1167810b18@209.191.44.130 (Checking From) --From tag as7a637cd3 --To-tag [May 23 13:12:46] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 13:12:46] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:46] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 1213f62f16db832f7755ab1167810b18@209.191.44.130 - OPTIONS (No RTP) [May 23 13:12:46] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 13:12:46] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 13:12:46] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5d07fe66-394bec48@10.0.15.101 (Checking From) --From tag c7d0e91e95d40f0o0 --To-tag [May 23 13:12:46] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:46] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 055b041a66a73b4814178d5f399c8941@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:47] DEBUG[13067] acl.c: For destination '64.19.145.18', our source address is '64.19.145.13'. [May 23 13:12:47] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 38e3ca35152d477347753804078a0423@64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.18:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: = Looking for Call ID: 38e3ca35152d477347753804078a0423@64.19.145.13:5060 (Checking To) --From tag as0d2cb652 --To-tag as33daf22d [May 23 13:12:47] DEBUG[13067] chan_sip.c: Stopping retransmission on '38e3ca35152d477347753804078a0423@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:47] DEBUG[13067] chan_sip.c: Destroying SIP dialog 38e3ca35152d477347753804078a0423@64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 72cd9563496b4a020777e3fa1bc13213@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:47] DEBUG[13067] acl.c: For destination '209.191.13.243', our source address is '64.19.145.13'. [May 23 13:12:47] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 396aed6c5d0f351514f15db9451b8ee2@64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 0c6e4c806b2b83b37f5c15c2665f664b@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:47] DEBUG[13067] acl.c: For destination '64.19.145.15', our source address is '64.19.145.13'. [May 23 13:12:47] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 73334fa20820944e2c1ad0cf7f625d7f@64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.15:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: = Looking for Call ID: 73334fa20820944e2c1ad0cf7f625d7f@64.19.145.13:5060 (Checking To) --From tag as458a4c4a --To-tag as5af1b859 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Stopping retransmission on '73334fa20820944e2c1ad0cf7f625d7f@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:47] DEBUG[13067] chan_sip.c: Destroying SIP dialog 73334fa20820944e2c1ad0cf7f625d7f@64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 36df0547477fec1466cd9dbb3b919fe9@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:47] DEBUG[13067] acl.c: For destination '64.19.145.7', our source address is '64.19.145.13'. [May 23 13:12:47] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 2e2c5a6c0329f8db07b07944793601b6@64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.7:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: = Looking for Call ID: 2e2c5a6c0329f8db07b07944793601b6@64.19.145.13:5060 (Checking To) --From tag as3f0d36d0 --To-tag as2198250c [May 23 13:12:47] DEBUG[13067] chan_sip.c: Stopping retransmission on '2e2c5a6c0329f8db07b07944793601b6@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:47] DEBUG[13067] chan_sip.c: Destroying SIP dialog 2e2c5a6c0329f8db07b07944793601b6@64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 5cdd0dd22f01b15479d4253278f2984e@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:47] DEBUG[13067] acl.c: For destination '64.19.145.12', our source address is '64.19.145.13'. [May 23 13:12:47] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 6b3701775e6ed4c53c0ad70702b79a9f@64.19.145.13:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.12:5060 [May 23 13:12:47] DEBUG[13067] chan_sip.c: = Looking for Call ID: 6b3701775e6ed4c53c0ad70702b79a9f@64.19.145.13:5060 (Checking To) --From tag as3a9d07c1 --To-tag as7cbdd7be [May 23 13:12:47] DEBUG[13067] chan_sip.c: Stopping retransmission on '6b3701775e6ed4c53c0ad70702b79a9f@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:47] DEBUG[13067] chan_sip.c: Destroying SIP dialog 6b3701775e6ed4c53c0ad70702b79a9f@64.19.145.13:5060 [May 23 13:12:48] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 438d1a744ae570966080bf6f4e88901b@127.0.0.1:0 - OPTIONS (No RTP) [May 23 13:12:48] DEBUG[13067] acl.c: For destination '64.19.145.11', our source address is '64.19.145.13'. [May 23 13:12:48] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 13:12:48] DEBUG[13067] chan_sip.c: Initializing initreq for method OPTIONS - callid 55d6ffcb609c30d84b5d34af6f7e8978@64.19.145.13:5060 [May 23 13:12:48] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 64.19.145.11:5060 [May 23 13:12:48] DEBUG[13067] chan_sip.c: = Looking for Call ID: 55d6ffcb609c30d84b5d34af6f7e8978@64.19.145.13:5060 (Checking To) --From tag as68592ac8 --To-tag as52337d2f [May 23 13:12:48] DEBUG[13067] chan_sip.c: Stopping retransmission on '55d6ffcb609c30d84b5d34af6f7e8978@64.19.145.13:5060' of Request 102: Match Found [May 23 13:12:48] DEBUG[13067] chan_sip.c: Destroying SIP dialog 55d6ffcb609c30d84b5d34af6f7e8978@64.19.145.13:5060 [May 23 13:12:48] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:48] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '3cae8f2579381430056cf7be37ccb991@209.191.44.130' [May 23 13:12:48] DEBUG[13067] chan_sip.c: Destroying SIP dialog 3cae8f2579381430056cf7be37ccb991@209.191.44.130 [May 23 13:12:49] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:50] DEBUG[13067] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 209.191.13.243:26300 [May 23 13:12:50] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 13:12:50] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 13:12:50] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 13:13:36] NOTICE[13067] chan_sip.c: Registration from '' failed for '209.191.39.117:5060' - No matching peer found [May 23 13:13:36] NOTICE[13067] chan_sip.c: Registration from '' failed for '209.191.39.117:5060' - No matching peer found