[May 23 15:35:04] VERBOSE[19578] config.c: == Parsing '/etc/asterisk-2011-05-03/logger.conf': [May 23 15:35:04] DEBUG[19578] config.c: Parsing /etc/asterisk-2011-05-03/logger.conf [May 23 15:35:04] VERBOSE[19578] config.c: == Found [May 23 15:35:04] VERBOSE[19578] logger.c: Asterisk Queue Logger restarted [May 23 15:35:07] 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=z9hG4bK4a89cd31;rport Max-Forwards: 70 From: "asterisk" ;tag=as78c49654 To: Contact: Call-ID: 5dfd2c573ee05f603c81e6f2657ef48d@209.191.44.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 19:35:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK4a89cd31;rport [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as78c49654 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 5 [ 38]: Contact: [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 6 [ 56]: Call-ID: 5dfd2c573ee05f603c81e6f2657ef48d@209.191.44.130 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 19:35:07 GMT [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 15:35:07] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 15:35:07] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5dfd2c573ee05f603c81e6f2657ef48d@209.191.44.130 (Checking From) --From tag as78c49654 --To-tag [May 23 15:35:07] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 15:35:07] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 5dfd2c573ee05f603c81e6f2657ef48d@209.191.44.130 - OPTIONS (No RTP) [May 23 15:35:07] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 15:35:07] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 15:35:07] 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=z9hG4bK4a89cd31;rport;received=209.191.44.130 From: "asterisk" ;tag=as78c49654 To: ;tag=as4ebb93d7 Call-ID: 5dfd2c573ee05f603c81e6f2657ef48d@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 15:35:07] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 15:35:07] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '5dfd2c573ee05f603c81e6f2657ef48d@209.191.44.130' in 32000 ms (Method: OPTIONS) [May 23 15:35:07] 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-f4598b1c From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114428 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-f4598b1c [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 114428 NOTIFY [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:07] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:07] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:07] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:07] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:07] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 114428, ours 114428) [May 23 15:35:07] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:07] 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-f4598b1c;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114428 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 15:35:07] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 15:35:07] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 15:35:09] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '01cb5f860438b0d0285dcbf41bfed79f@209.191.44.130' [May 23 15:35:09] DEBUG[13067] chan_sip.c: Destroying SIP dialog 01cb5f860438b0d0285dcbf41bfed79f@209.191.44.130 [May 23 15:35:09] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '01cb5f860438b0d0285dcbf41bfed79f@209.191.44.130' Method: OPTIONS [May 23 15:35:09] 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: 28744 NOTIFY Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-945f4233 Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Contact: "SPA303 Cisco" Event: keep-alive Content-Length: 0 <-------------> [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 2 [ 22]: To: [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 3 [ 38]: Call-ID: 5d07fe66-394bec48@10.0.15.101 [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 4 [ 18]: CSeq: 28744 NOTIFY [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 5 [ 60]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-945f4233 [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 7 [ 30]: User-Agent: Cisco/SPA303-7.4.6 [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 8 [ 82]: Contact: "SPA303 Cisco" [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Event: keep-alive [May 23 15:35:09] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:09] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:09] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5d07fe66-394bec48@10.0.15.101 (Checking From) --From tag c7d0e91e95d40f0o0 --To-tag [May 23 15:35:09] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:09] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:09] 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-945f4233;received=209.191.39.117 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: ;tag=as1046500e Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 28744 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 15:35:09] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:09] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '5d07fe66-394bec48@10.0.15.101' in 32000 ms (Method: NOTIFY) [May 23 15:35:10] 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=z9hG4bK2d6080f6;rport From: "asterisk" ;tag=as0ab3cac6 To: Contact: Call-ID: 0708f50e36efe814200efa5a0f5cd657@64.19.145.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 23 May 2011 19:35:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <-------------> [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK2d6080f6;rport [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 2 [ 58]: From: "asterisk" ;tag=as0ab3cac6 [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 4 [ 35]: Contact: [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 5 [ 53]: Call-ID: 0708f50e36efe814200efa5a0f5cd657@64.19.145.4 [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 7 [ 24]: User-Agent: Asterisk PBX [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 19:35:10 GMT [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 10 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 15:35:10] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 15:35:10] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 15:35:10] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0708f50e36efe814200efa5a0f5cd657@64.19.145.4 (Checking From) --From tag as0ab3cac6 --To-tag [May 23 15:35:10] DEBUG[13067] acl.c: For destination '64.19.145.4', our source address is '64.19.145.13'. [May 23 15:35:10] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:10] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 0708f50e36efe814200efa5a0f5cd657@64.19.145.4 - OPTIONS (No RTP) [May 23 15:35:10] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 15:35:10] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 15:35:10] 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=z9hG4bK2d6080f6;rport;received=64.19.145.4 From: "asterisk" ;tag=as0ab3cac6 To: ;tag=as2c7b9208 Call-ID: 0708f50e36efe814200efa5a0f5cd657@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 15:35:10] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 64.19.145.4:5060 [May 23 15:35:10] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '0708f50e36efe814200efa5a0f5cd657@64.19.145.4' in 32000 ms (Method: OPTIONS) [May 23 15:35:11] 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-f4598b1c From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114428 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-f4598b1c [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 114428 NOTIFY [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:11] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:11] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:11] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:11] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:11] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 114428, ours 114428) [May 23 15:35:11] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:11] 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-f4598b1c;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114428 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 15:35:11] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 15:35:11] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 15:35:12] 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-14843ec3 From: ;tag=5e35c995200173e1o3 To: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94858 REGISTER Max-Forwards: 70 Authorization: Digest username="175-eng",realm="asterisk",nonce="6718e71a",uri="sip:64.19.145.13",algorithm=MD5,response="915ba4c566bf3bf460d59fa936f1c8d6" 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 15:35:12] DEBUG[13067] chan_sip.c: Header 0 [ 33]: REGISTER sip:64.19.145.13 SIP/2.0 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-14843ec3 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 3 [ 30]: To: [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 5 [ 20]: CSeq: 94858 REGISTER [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 7 [155]: Authorization: Digest username="175-eng",realm="asterisk",nonce="6718e71a",uri="sip:64.19.145.13",algorithm=MD5,response="915ba4c566bf3bf460d59fa936f1c8d6" [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 8 [ 55]: Contact: ;expires=3600 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 11 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 12 [ 19]: Supported: replaces [May 23 15:35:12] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 15:35:12] DEBUG[13067] chan_sip.c: = Looking for Call ID: b7b02bc6-e8d28b72@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:12] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 15:35:12] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 15:35:12] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 15:35:12] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 15:35:12] DEBUG[13067] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [May 23 15:35:12] DEBUG[13067] chan_sip.c: Initializing initreq for method REGISTER - callid b7b02bc6-e8d28b72@192.168.15.187 [May 23 15:35:12] DEBUG[13067] netsock2.c: Splitting '192.168.15.187:5063' gives... [May 23 15:35:12] DEBUG[13067] netsock2.c: ...host '192.168.15.187' and port '5063'. [May 23 15:35:12] VERBOSE[13067] chan_sip.c: Sending to 209.191.13.243:17616 (NAT) [May 23 15:35:12] NOTICE[13067] chan_sip.c: Correct auth, but based on stale nonce received from ';tag=5e35c995200173e1o3' [May 23 15:35:12] 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-14843ec3;received=209.191.13.243;rport=17616 From: ;tag=5e35c995200173e1o3 To: ;tag=as1f6b40e0 Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94858 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="5604722c", stale=true Content-Length: 0 <------------> [May 23 15:35:12] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 209.191.13.243:17616 [May 23 15:35:12] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'b7b02bc6-e8d28b72@192.168.15.187' in 32000 ms (Method: REGISTER) [May 23 15:35:12] 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-1a660674 From: ;tag=5e35c995200173e1o3 To: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94859 REGISTER Max-Forwards: 70 Authorization: Digest username="175-eng",realm="asterisk",nonce="5604722c",uri="sip:64.19.145.13",algorithm=MD5,response="f76ae6a77f3ff02e5c492969008f97c6" 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 15:35:12] DEBUG[13067] chan_sip.c: Header 0 [ 33]: REGISTER sip:64.19.145.13 SIP/2.0 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-1a660674 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 3 [ 30]: To: [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: b7b02bc6-e8d28b72@192.168.15.187 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 5 [ 20]: CSeq: 94859 REGISTER [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 7 [155]: Authorization: Digest username="175-eng",realm="asterisk",nonce="5604722c",uri="sip:64.19.145.13",algorithm=MD5,response="f76ae6a77f3ff02e5c492969008f97c6" [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 8 [ 55]: Contact: ;expires=3600 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 11 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [May 23 15:35:12] DEBUG[13067] chan_sip.c: Header 12 [ 19]: Supported: replaces [May 23 15:35:12] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 15:35:12] DEBUG[13067] chan_sip.c: = Looking for Call ID: b7b02bc6-e8d28b72@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:12] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 15:35:12] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 15:35:12] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 15:35:12] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 15:35:12] DEBUG[13067] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [May 23 15:35:12] DEBUG[13067] chan_sip.c: Initializing initreq for method REGISTER - callid b7b02bc6-e8d28b72@192.168.15.187 [May 23 15:35:12] DEBUG[13067] netsock2.c: Splitting '192.168.15.187:5063' gives... [May 23 15:35:12] DEBUG[13067] netsock2.c: ...host '192.168.15.187' and port '5063'. [May 23 15:35:12] VERBOSE[13067] chan_sip.c: Sending to 209.191.13.243:17616 (NAT) [May 23 15:35:12] DEBUG[13067] chan_sip.c: Store REGISTER's src-IP:port for call routing. [May 23 15:35:12] 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 15:35:12] 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-1a660674;received=209.191.13.243;rport=17616 From: ;tag=5e35c995200173e1o3 To: ;tag=as1f6b40e0 Call-ID: b7b02bc6-e8d28b72@192.168.15.187 CSeq: 94859 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 19:35:12 GMT Content-Length: 0 <------------> [May 23 15:35:12] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:17616 [May 23 15:35:12] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'b7b02bc6-e8d28b72@192.168.15.187' in 32000 ms (Method: REGISTER) [May 23 15:35:12] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 175-eng [May 23 15:35:12] DEBUG[13069] chan_sip.c: Checking device state for peer 175-eng [May 23 15:35:12] DEBUG[13069] devicestate.c: Changing state for SIP/175-eng - state 1 (Not in use) [May 23 15:35:12] DEBUG[13069] devicestate.c: device 'SIP/175-eng' state '1' [May 23 15:35:12] 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 15:35:12] 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 15:35:12] DEBUG[13094] app_queue.c: Device 'SIP/175-eng' changed to state '1' (Not in use) [May 23 15:35:12] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '269f250-0-13c4-62-72b2d122-62' [May 23 15:35:12] DEBUG[13067] chan_sip.c: Destroying SIP dialog 269f250-0-13c4-62-72b2d122-62 [May 23 15:35:12] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '269f250-0-13c4-62-72b2d122-62' Method: REGISTER [May 23 15:35:15] 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=z9hG4bK20b8841f;rport Max-Forwards: 70 From: "7327049020" ;tag=as22de05f6 To: Contact: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 19:35:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 221 v=0 o=root 628746205 628746205 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 10046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 0 [ 42]: INVITE sip:7327049020@64.19.145.13 SIP/2.0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK20b8841f;rport [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 3 [ 62]: From: "7327049020" ;tag=as22de05f6 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 4 [ 33]: To: [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 5 [ 37]: Contact: [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 6 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 19:35:15 GMT [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 13 [ 19]: Content-Length: 221 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 14 [ 0]: [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 1 [ 45]: o=root 628746205 628746205 IN IP4 64.19.145.7 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 2 [ 40]: s=Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 3 [ 20]: c=IN IP4 64.19.145.7 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 10046 RTP/AVP 0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 7 [ 25]: a=silenceSupp:off - - - - [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 8 [ 10]: a=ptime:20 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=sendrecv [May 23 15:35:15] VERBOSE[13067] chan_sip.c: --- (14 headers 10 lines) --- [May 23 15:35:15] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking From) --From tag as22de05f6 --To-tag [May 23 15:35:15] DEBUG[13067] acl.c: For destination '64.19.145.7', our source address is '64.19.145.13'. [May 23 15:35:15] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 0089679f1d3712a573e92dbe03d33782@64.19.145.7 - INVITE (No RTP) [May 23 15:35:15] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 15:35:15] DEBUG[13067] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer" [May 23 15:35:15] DEBUG[13067] sip/reqresp_parser.c: Found SIP option: -replaces- [May 23 15:35:15] DEBUG[13067] sip/reqresp_parser.c: Matched SIP option: replaces [May 23 15:35:15] DEBUG[13067] sip/reqresp_parser.c: Found SIP option: -timer- [May 23 15:35:15] DEBUG[13067] sip/reqresp_parser.c: Matched SIP option: timer [May 23 15:35:15] DEBUG[13067] netsock2.c: Splitting '64.19.145.7:5060' gives... [May 23 15:35:15] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '5060'. [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Sending to 64.19.145.7:5060 (no NAT) [May 23 15:35:15] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Using INVITE request as basis request - 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Found peer 'mg2' for '7327049020' from 64.19.145.7:5060 [May 23 15:35:15] 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=z9hG4bK20b8841f;received=64.19.145.7;rport=5060 From: "7327049020" ;tag=as22de05f6 To: ;tag=as3a7067f5 Call-ID: 0089679f1d3712a573e92dbe03d33782@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="7fba3d1a" Content-Length: 0 <------------> [May 23 15:35:15] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065028 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' in 32000 ms (Method: INVITE) [May 23 15:35:15] 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=z9hG4bK20b8841f;rport Max-Forwards: 70 From: "7327049020" ;tag=as22de05f6 To: ;tag=as3a7067f5 Contact: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Content-Length: 0 <-------------> [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 0 [ 39]: ACK sip:7327049020@64.19.145.13 SIP/2.0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK20b8841f;rport [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 3 [ 62]: From: "7327049020" ;tag=as22de05f6 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 4 [ 48]: To: ;tag=as3a7067f5 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 5 [ 37]: Contact: [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 6 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [May 23 15:35:15] VERBOSE[13067] chan_sip.c: --- (10 headers 0 lines) --- [May 23 15:35:15] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking From) --From tag as22de05f6 --To-tag as3a7067f5 [May 23 15:35:15] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 15:35:15] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065028 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Stopping retransmission on '0089679f1d3712a573e92dbe03d33782@64.19.145.7' of Response 102: Match Found [May 23 15:35:15] 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=z9hG4bK1bed86bf;rport Max-Forwards: 70 From: "7327049020" ;tag=as22de05f6 To: Contact: Call-ID: 0089679f1d3712a573e92dbe03d33782@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="7fba3d1a", response="120300c362d1c3a031fb3a114deee3d6" Date: Mon, 23 May 2011 19:35:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 221 v=0 o=root 628746205 628746206 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 10046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 0 [ 42]: INVITE sip:7327049020@64.19.145.13 SIP/2.0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK1bed86bf;rport [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 3 [ 62]: From: "7327049020" ;tag=as22de05f6 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 4 [ 33]: To: [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 5 [ 37]: Contact: [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 6 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 9 [167]: Authorization: Digest username="mg2", realm="asterisk", algorithm=MD5, uri="sip:7327049020@64.19.145.13", nonce="7fba3d1a", response="120300c362d1c3a031fb3a114deee3d6" [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 10 [ 35]: Date: Mon, 23 May 2011 19:35:15 GMT [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 11 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 14 [ 19]: Content-Length: 221 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 15 [ 0]: [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 1 [ 45]: o=root 628746205 628746206 IN IP4 64.19.145.7 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 2 [ 40]: s=Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 3 [ 20]: c=IN IP4 64.19.145.7 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 10046 RTP/AVP 0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 7 [ 25]: a=silenceSupp:off - - - - [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 8 [ 10]: a=ptime:20 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=sendrecv [May 23 15:35:15] VERBOSE[13067] chan_sip.c: --- (15 headers 10 lines) --- [May 23 15:35:15] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking From) --From tag as22de05f6 --To-tag [May 23 15:35:15] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 15:35:15] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 15:35:15] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 15:35:15] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 15:35:15] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 15:35:15] DEBUG[13067] netsock2.c: Splitting '64.19.145.7:5060' gives... [May 23 15:35:15] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '5060'. [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Sending to 64.19.145.7:5060 (no NAT) [May 23 15:35:15] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Using INVITE request as basis request - 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Found peer 'mg2' for '7327049020' from 64.19.145.7:5060 [May 23 15:35:15] DEBUG[13067] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb69c1068' [May 23 15:35:15] DEBUG[13067] res_rtp_asterisk.c: Allocated port 17626 for RTP instance '0xb69c1068' [May 23 15:35:15] DEBUG[13067] rtp_engine.c: RTP instance '0xb69c1068' is setup and ready to go [May 23 15:35:15] DEBUG[13067] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb69c1068' [May 23 15:35:15] VERBOSE[13067] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Setting NAT on RTP to Off [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing session-level SDP o=root 628746205 628746206 IN IP4 64.19.145.7... UNSUPPORTED. [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Asterisk PBX SVN-branch-1.6.1-r230383M... UNSUPPORTED. [May 23 15:35:15] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:15] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 64.19.145.7... OK. [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:15] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd39c [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 15:35:15] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 15:35:15] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd39c [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 15:35:15] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb69c1068' [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 64.19.145.7:10046 [May 23 15:35:15] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd39c to 0xb69c1214 [May 23 15:35:15] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 15:35:15] DEBUG[13067] chan_sip.c: Checking SIP call limits for device fsdev-mg2 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Updating call counter for incoming call [May 23 15:35:15] VERBOSE[13067] chan_sip.c: Looking for 7327049020 in mtt-from-outside (domain 64.19.145.13) [May 23 15:35:15] DEBUG[13067] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 15:35:15] DEBUG[13067] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 15:35:15] DEBUG[13067] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [May 23 15:35:15] DEBUG[13067] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 15:35:15] DEBUG[13067] chan_sip.c: This channel will not be able to handle video. [May 23 15:35:15] DEBUG[13067] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 15:35:15] DEBUG[13067] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 15:35:15] DEBUG[13067] chan_sip.c: build_route: Contact hop: [May 23 15:35:15] VERBOSE[13067] chan_sip.c: list_route: hop: [May 23 15:35:15] DEBUG[13067] chan_sip.c: SIP/mg2-00000026: New call is still down.... Trying... [May 23 15:35:15] 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=z9hG4bK1bed86bf;received=64.19.145.7;rport=5060 From: "7327049020" ;tag=as22de05f6 To: Call-ID: 0089679f1d3712a573e92dbe03d33782@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 15:35:15] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:15] DEBUG[19579] pbx.c: Launching 'Dial' [May 23 15:35:15] VERBOSE[19579] pbx.c: -- Executing [7327049020@mtt-from-outside:1] Dial("SIP/mg2-00000026", "SIP/322-eng") in new stack [May 23 15:35:15] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - mg2 [May 23 15:35:15] DEBUG[13069] chan_sip.c: Checking device state for peer mg2 [May 23 15:35:15] DEBUG[13069] devicestate.c: Changing state for SIP/mg2 - state 1 (Not in use) [May 23 15:35:15] DEBUG[13069] devicestate.c: device 'SIP/mg2' state '1' [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/mg2-00000026 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 7327049020 CallerIDName: 7327049020 AccountCode: Exten: 7327049020 Context: mtt-from-outside Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: SIPURI Value: sip:7327049020@64.19.145.7 Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: SIPDOMAIN Value: 64.19.145.13 Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: SIPCALLID Value: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/mg2-00000026 Uniqueid: 1306179315.38 Channeltype: SIP SIPcallid: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 SIPfullcontact: [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/mg2-00000026 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 7327049020 CallerIDName: 7327049020 Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/mg2-00000026 Context: mtt-from-outside Extension: 7327049020 Priority: 1 Application: Dial AppData: SIP/322-eng Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALSTATUS Value: Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALEDPEERNUMBER Value: Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALEDPEERNAME Value: Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ANSWEREDTIME Value: Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALEDTIME Value: Uniqueid: 1306179315.38 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 23 15:35:15] DEBUG[19579] chan_sip.c: Allocating new SIP dialog for 03b223143a106cf17a468844704f6015@127.0.0.1:0 - INVITE (No RTP) [May 23 15:35:15] DEBUG[19579] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xaab42f8' [May 23 15:35:15] DEBUG[19579] res_rtp_asterisk.c: Allocated port 15892 for RTP instance '0xaab42f8' [May 23 15:35:15] DEBUG[19579] rtp_engine.c: RTP instance '0xaab42f8' is setup and ready to go [May 23 15:35:15] DEBUG[19579] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xaab42f8' [May 23 15:35:15] VERBOSE[19579] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Setting NAT on RTP to Off [May 23 15:35:15] DEBUG[19579] chan_sip.c: OBPROXY: Not applying OBproxy to this call [May 23 15:35:15] DEBUG[19579] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 15:35:15] DEBUG[19579] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/322-eng-00000027 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 322 CallerIDName: Poly_test ENG AccountCode: eng Exten: Context: test Uniqueid: 1306179315.39 [May 23 15:35:15] DEBUG[19579] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 15:35:15] DEBUG[19579] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 15:35:15] DEBUG[19579] chan_sip.c: *** Our capabilities are 0x404 (ulaw|ilbc) [May 23 15:35:15] DEBUG[19579] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 15:35:15] DEBUG[19579] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 23 15:35:15] DEBUG[19579] chan_sip.c: This channel will not be able to handle video. [May 23 15:35:15] DEBUG[19579] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 15:35:15] DEBUG[19579] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: SIPCALLID Value: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 Uniqueid: 1306179315.39 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/322-eng-00000027 Uniqueid: 1306179315.39 Channeltype: SIP SIPcallid: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 SIPfullcontact: sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/322-eng-00000027 Channeltype: SIP SIPcallid: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 SIPfullcontact: sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 Peername: 322-eng [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: DIALEDPEERNUMBER Value: 322-eng Uniqueid: 1306179315.39 [May 23 15:35:15] DEBUG[19579] rtp_engine.c: Seeded SDP of 'SIP/322-eng-00000027' with that of 'SIP/mg2-00000026' [May 23 15:35:15] DEBUG[19579] channel.c: Not copying variable DIALEDTIME. [May 23 15:35:15] DEBUG[19579] channel.c: Not copying variable ANSWEREDTIME. [May 23 15:35:15] DEBUG[19579] channel.c: Not copying variable DIALEDPEERNAME. [May 23 15:35:15] DEBUG[19579] channel.c: Not copying variable DIALEDPEERNUMBER. [May 23 15:35:15] DEBUG[19579] channel.c: Not copying variable DIALSTATUS. [May 23 15:35:15] DEBUG[19579] channel.c: Not copying variable SIPCALLID. [May 23 15:35:15] DEBUG[19579] channel.c: Not copying variable SIPDOMAIN. [May 23 15:35:15] DEBUG[19579] channel.c: Not copying variable SIPURI. [May 23 15:35:15] DEBUG[19579] chan_sip.c: Outgoing Call for 322-eng [May 23 15:35:15] DEBUG[19579] chan_sip.c: Updating call counter for outgoing call [May 23 15:35:15] DEBUG[19579] chan_sip.c: Call to peer '322-eng' is 1 out of 2147483647 [May 23 15:35:15] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:15] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:15] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 6 (Ringing) [May 23 15:35:15] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '6' [May 23 15:35:15] DEBUG[19579] chan_sip.c: ** Our capability: 0x404 (ulaw|ilbc) Video flag: False Text flag: False [May 23 15:35:15] DEBUG[19579] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 15:35:15] VERBOSE[19579] chan_sip.c: Audio is at 5060 [May 23 15:35:15] VERBOSE[19579] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:15] VERBOSE[19579] chan_sip.c: Adding codec 0x400 (ilbc) to SDP [May 23 15:35:15] DEBUG[19579] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:15] DEBUG[19579] chan_sip.c: Done building SDP. Settling with this capability: 0x404 (ulaw|ilbc) [May 23 15:35:15] DEBUG[19579] chan_sip.c: Initializing initreq for method INVITE - callid 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 0 [ 72]: INVITE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2696073e [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 3 [ 63]: From: "7327049020" ;tag=as7a9f2f18 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 4 [ 63]: To: [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 5 [ 43]: Contact: [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 6 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 19:35:15 GMT [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 15:35:15] DEBUG[19579] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 15:35:15] VERBOSE[19579] 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=z9hG4bK2696073e Max-Forwards: 70 From: "7327049020" ;tag=as7a9f2f18 To: Contact: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 19:35:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 1146564904 1146564904 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 15892 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 15:35:15] DEBUG[19579] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065031 [May 23 15:35:15] DEBUG[19579] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:15] VERBOSE[19579] app_dial.c: -- Called SIP/322-eng [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/mg2-00000026 Destination: SIP/322-eng-00000027 CallerIDNum: 7327049020 CallerIDName: 7327049020 UniqueID: 1306179315.38 DestUniqueID: 1306179315.39 Dialstring: 322-eng [May 23 15:35:15] 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 15:35:15] 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 15:35:15] 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 15:35:15] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '6' (Ringing) [May 23 15:35:15] 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 15:35:15] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 100 Trying From: "7327049020";tag=as7a9f2f18 To: "Poly_test ENG";tag=82A90870-A5BD6FFB Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2696073e Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Length: 0 <-------------> [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 1 [ 62]: From: "7327049020";tag=as7a9f2f18 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2696073e [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 15:35:15] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [May 23 15:35:15] VERBOSE[13067] chan_sip.c: --- (10 headers 0 lines) --- [May 23 15:35:15] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking To) --From tag as7a9f2f18 --To-tag 82A90870-A5BD6FFB [May 23 15:35:15] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1065031 - INVITE (got response) [May 23 15:35:15] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Request 102: Found [May 23 15:35:15] DEBUG[13067] chan_sip.c: SIP response 100 to standard invite [May 23 15:35:15] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000027~ Value: SIP 100 Trying Uniqueid: 1306179315.38 [May 23 15:35:16] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 180 Ringing From: "7327049020";tag=as7a9f2f18 To: "Poly_test ENG";tag=82A90870-A5BD6FFB Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2696073e Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Allow-Events: talk,hold,conference Content-Length: 0 <-------------> [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 1 [ 62]: From: "7327049020";tag=as7a9f2f18 [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2696073e [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 9 [ 34]: Allow-Events: talk,hold,conference [May 23 15:35:16] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:16] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:16] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking To) --From tag as7a9f2f18 --To-tag 82A90870-A5BD6FFB [May 23 15:35:16] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Request 102: Found [May 23 15:35:16] DEBUG[13067] chan_sip.c: SIP response 180 to standard invite [May 23 15:35:16] VERBOSE[19579] app_dial.c: -- SIP/322-eng-00000027 is ringing [May 23 15:35:16] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:16] DEBUG[19579] rtp_engine.c: Setting early bridge SDP of 'SIP/mg2-00000026' with that of 'SIP/322-eng-00000027' [May 23 15:35:16] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:16] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 6 (Ringing) [May 23 15:35:16] VERBOSE[19579] 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=z9hG4bK1bed86bf;received=64.19.145.7;rport=5060 From: "7327049020" ;tag=as22de05f6 To: ;tag=as3a9ecd75 Call-ID: 0089679f1d3712a573e92dbe03d33782@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 15:35:16] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/322-eng-00000027 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 322 CallerIDName: Poly_test ENG Uniqueid: 1306179315.39 [May 23 15:35:16] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '6' [May 23 15:35:16] DEBUG[19579] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:16] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000027~ Value: SIP 180 Ringing Uniqueid: 1306179315.38 [May 23 15:35:16] 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 15:35:16] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '6' (Ringing) [May 23 15:35:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "7327049020";tag=as7a9f2f18 To: "Poly_test ENG";tag=82A90870-A5BD6FFB Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2696073e 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=- 1306179293 1306179293 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 m=audio 51836 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 1 [ 62]: From: "7327049020";tag=as7a9f2f18 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2696073e [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Supported: 100rel [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Supported: replaces [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 10 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Accept-Language: en [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 12 [ 29]: Content-Type: application/SDP [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 13 [ 19]: Content-Length: 165 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 14 [ 0]: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306179293 1306179293 IN IP4 209.191.39.117 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 51836 RTP/AVP 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 6 [ 10]: a=sendrecv [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: --- (14 headers 8 lines) --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking To) --From tag as7a9f2f18 --To-tag 82A90870-A5BD6FFB [May 23 15:35:17] DEBUG[13067] chan_sip.c: Acked pending invite 102 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' of Request 102: Match Found [May 23 15:35:17] DEBUG[13067] chan_sip.c: SIP response 200 to standard invite [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 1306179293 1306179293 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 15:35:17] 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 15:35:17] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 15:35:17] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaab42f8' [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51836 [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xaab44a4 [May 23 15:35:17] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 15:35:17] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 15:35:17] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 15:35:17] DEBUG[13067] chan_sip.c: build_route: Contact hop: [May 23 15:35:17] VERBOSE[13067] chan_sip.c: list_route: hop: [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Strict routing enforced for session 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:17] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:17] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:17] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:17] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 2 (In use) [May 23 15:35:17] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 15:35:17] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '2' [May 23 15:35:17] 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=z9hG4bK666552b8 Max-Forwards: 70 From: "7327049020" ;tag=as7a9f2f18 To: ;tag=82A90870-A5BD6FFB Contact: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:322' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:17] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:17] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:17] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 2 (In use) [May 23 15:35:17] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '2' [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/322-eng-00000027 Channeltype: SIP Uniqueid: 1306179315.39 SIPcallid: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 SIPfullcontact: sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 Peername: 322-eng [May 23 15:35:17] 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: 2 Paused: 0 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000027~ Value: SIP 200 OK Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/322-eng-00000027 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 322 CallerIDName: Poly_test ENG Uniqueid: 1306179315.39 [May 23 15:35:17] 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 15:35:17] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '2' (In use) [May 23 15:35:17] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '2' (In use) [May 23 15:35:17] VERBOSE[19579] app_dial.c: -- SIP/322-eng-00000027 answered SIP/mg2-00000026 [May 23 15:35:17] DEBUG[19579] rtp_engine.c: Setting early bridge SDP of 'SIP/mg2-00000026' with that of 'SIP/322-eng-00000027' [May 23 15:35:17] 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: 2 Paused: 0 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1306179315.38 [May 23 15:35:17] 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 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALEDPEERNAME Value: SIP/322-eng-00000027 Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALEDPEERNUMBER Value: 322-eng Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - mg2 [May 23 15:35:17] DEBUG[13069] chan_sip.c: Checking device state for peer mg2 [May 23 15:35:17] DEBUG[13069] devicestate.c: Changing state for SIP/mg2 - state 1 (Not in use) [May 23 15:35:17] DEBUG[13069] devicestate.c: device 'SIP/mg2' state '1' [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: BRIDGEPEER Value: SIP/322-eng-00000027 Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: BRIDGEPEER Value: SIP/mg2-00000026 Uniqueid: 1306179315.39 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/mg2-00000026 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 7327049020 CallerIDName: 7327049020 Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[19579] chan_sip.c: SIP answering channel: SIP/mg2-00000026 [May 23 15:35:17] DEBUG[19579] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:17] DEBUG[19579] chan_sip.c: Setting framing from config on incoming call [May 23 15:35:17] DEBUG[19579] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 15:35:17] DEBUG[19579] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 15:35:17] VERBOSE[19579] chan_sip.c: Audio is at 5060 [May 23 15:35:17] VERBOSE[19579] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:17] DEBUG[19579] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:17] DEBUG[19579] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 15:35:17] VERBOSE[19579] 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=z9hG4bK1bed86bf;received=64.19.145.7;rport=5060 From: "7327049020" ;tag=as22de05f6 To: ;tag=as3a9ecd75 Call-ID: 0089679f1d3712a573e92dbe03d33782@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 371132669 371132669 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 17626 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv <------------> [May 23 15:35:17] DEBUG[19579] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065034 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:17] 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 15:35:17] DEBUG[19579] features.c: bridge answer set, chan answer set [May 23 15:35:17] DEBUG[19579] features.c: Removing dialed interfaces datastore on SIP/322-eng-00000027 since we're bridging [May 23 15:35:17] 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=z9hG4bK280073f1;rport Max-Forwards: 70 From: "7327049020" ;tag=as22de05f6 To: ;tag=as3a9ecd75 Contact: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Content-Length: 0 <-------------> [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 0 [ 44]: ACK sip:7327049020@64.19.145.13:5060 SIP/2.0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.19.145.7:5060;branch=z9hG4bK280073f1;rport [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 3 [ 62]: From: "7327049020" ;tag=as22de05f6 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 4 [ 48]: To: ;tag=as3a9ecd75 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 5 [ 37]: Contact: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 6 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: --- (10 headers 0 lines) --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking From) --From tag as22de05f6 --To-tag as3a9ecd75 [May 23 15:35:17] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 15:35:17] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065034 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '0089679f1d3712a573e92dbe03d33782@64.19.145.7' of Response 103: Match Found [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/322-eng-00000027 Uniqueid: 1306179315.39 AccountCode: OldAccountCode: eng [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/mg2-00000026 Channel2: SIP/322-eng-00000027 Uniqueid1: 1306179315.38 Uniqueid2: 1306179315.39 CallerID1: 7327049020 CallerID2: 322 [May 23 15:35:17] DEBUG[19579] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:17] DEBUG[19579] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:17] VERBOSE[19579] rtp_engine.c: -- Remotely bridging SIP/mg2-00000026 and SIP/322-eng-00000027 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Sending reinvite on SIP '0089679f1d3712a573e92dbe03d33782@64.19.145.7' - It's audio soon redirected to IP 209.191.39.117:51836 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Strict routing enforced for session 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:17] VERBOSE[19579] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:17] DEBUG[19579] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:17] DEBUG[19579] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:17] VERBOSE[19579] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 15:35:17] DEBUG[19579] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 15:35:17] DEBUG[19579] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 15:35:17] DEBUG[19579] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [May 23 15:35:17] VERBOSE[19579] chan_sip.c: Audio is at 5060 [May 23 15:35:17] VERBOSE[19579] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:17] DEBUG[19579] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:17] DEBUG[19579] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 15:35:17] DEBUG[19579] chan_sip.c: Initializing already initialized SIP dialog 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (presumably reinvite) [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 0 [ 41]: INVITE sip:7327049020@64.19.145.7 SIP/2.0 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK673bdf1d;rport [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 3 [ 50]: From: ;tag=as3a9ecd75 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 4 [ 60]: To: "7327049020" ;tag=as22de05f6 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 5 [ 43]: Contact: [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 6 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 10 [ 19]: Supported: replaces [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 15:35:17] VERBOSE[19579] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: INVITE sip:7327049020@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK673bdf1d;rport Max-Forwards: 70 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Contact: Call-ID: 0089679f1d3712a573e92dbe03d33782@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 371132669 371132670 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 51836 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: BRIDGEPEER Value: SIP/322-eng-00000027 Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[19579] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065035 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Sending reinvite on SIP '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' - It's audio soon redirected to IP 64.19.145.7:10046 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Strict routing enforced for session 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:17] VERBOSE[19579] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:17] DEBUG[19579] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:17] DEBUG[19579] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:17] VERBOSE[19579] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 15:35:17] DEBUG[19579] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 15:35:17] DEBUG[19579] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 15:35:17] DEBUG[19579] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [May 23 15:35:17] VERBOSE[19579] chan_sip.c: Audio is at 5060 [May 23 15:35:17] VERBOSE[19579] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:17] DEBUG[19579] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:17] DEBUG[19579] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 15:35:17] DEBUG[19579] chan_sip.c: Initializing already initialized SIP dialog 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (presumably reinvite) [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 0 [ 72]: INVITE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK588a9985 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 3 [ 63]: From: "7327049020" ;tag=as7a9f2f18 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 4 [ 85]: To: ;tag=82A90870-A5BD6FFB [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 5 [ 43]: Contact: [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 6 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 10 [ 19]: Supported: replaces [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 23 15:35:17] DEBUG[19579] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 15:35:17] VERBOSE[19579] 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=z9hG4bK588a9985 Max-Forwards: 70 From: "7327049020" ;tag=as7a9f2f18 To: ;tag=82A90870-A5BD6FFB Contact: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@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: 193 v=0 o=root 1146564904 1146564905 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 10046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 15:35:17] DEBUG[19579] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065036 [May 23 15:35:17] DEBUG[19579] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:17] 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=z9hG4bK673bdf1d;received=64.19.145.13;rport=5060 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Call-ID: 0089679f1d3712a573e92dbe03d33782@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 15:35:17] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK673bdf1d;received=64.19.145.13;rport=5060 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as3a9ecd75 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 3 [ 60]: To: "7327049020" ;tag=as22de05f6 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 9 [ 37]: Contact: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking To) --From tag as3a9ecd75 --To-tag as22de05f6 [May 23 15:35:17] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1065035 - INVITE (got response) [May 23 15:35:17] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Request 102: Found [May 23 15:35:17] DEBUG[13067] chan_sip.c: SIP response 100 to RE-invite on outgoing call 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: BRIDGEPVTCALLID Value: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: BRIDGEPEER Value: SIP/mg2-00000026 Uniqueid: 1306179315.39 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: BRIDGEPVTCALLID Value: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 Uniqueid: 1306179315.39 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ~HASH~SIP_CAUSE~SIP/mg2-00000026~ Value: SIP 100 Trying Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:17] 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=z9hG4bK673bdf1d;received=64.19.145.13;rport=5060 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Call-ID: 0089679f1d3712a573e92dbe03d33782@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: 221 v=0 o=root 628746205 628746207 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 10046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK673bdf1d;received=64.19.145.13;rport=5060 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as3a9ecd75 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 3 [ 60]: To: "7327049020" ;tag=as22de05f6 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 9 [ 37]: Contact: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Content-Length: 221 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 12 [ 0]: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 1 [ 45]: o=root 628746205 628746207 IN IP4 64.19.145.7 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 2 [ 40]: s=Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 3 [ 20]: c=IN IP4 64.19.145.7 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 10046 RTP/AVP 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 7 [ 25]: a=silenceSupp:off - - - - [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 8 [ 10]: a=ptime:20 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=sendrecv [May 23 15:35:17] VERBOSE[13067] chan_sip.c: --- (12 headers 10 lines) --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking To) --From tag as3a9ecd75 --To-tag as22de05f6 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Acked pending invite 102 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '0089679f1d3712a573e92dbe03d33782@64.19.145.7' of Request 102: Match Found [May 23 15:35:17] DEBUG[13067] chan_sip.c: SIP response 200 to RE-invite on outgoing call 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP o=root 628746205 628746207 IN IP4 64.19.145.7... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Asterisk PBX SVN-branch-1.6.1-r230383M... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 64.19.145.7... OK. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 15:35:17] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb69c1068' [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 64.19.145.7:10046 [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xb69c1214 [May 23 15:35:17] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 15:35:17] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 15:35:17] DEBUG[13067] chan_sip.c: Updating call counter for incoming call [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Strict routing enforced for session 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:17] VERBOSE[13067] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Transmitting (no NAT) to 64.19.145.7:5060: ACK sip:7327049020@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK308b7c21;rport Max-Forwards: 70 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Contact: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:732' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ~HASH~SIP_CAUSE~SIP/mg2-00000026~ Value: SIP 200 OK Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:17] 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=z9hG4bK44a3e801;rport Max-Forwards: 70 From: "asterisk" ;tag=as2ee08bd8 To: Contact: Call-ID: 37442df44804a8370b396dea59726037@209.191.44.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 19:35:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK44a3e801;rport [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as2ee08bd8 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 5 [ 38]: Contact: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 6 [ 56]: Call-ID: 37442df44804a8370b396dea59726037@209.191.44.130 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 19:35:17 GMT [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 37442df44804a8370b396dea59726037@209.191.44.130 (Checking From) --From tag as2ee08bd8 --To-tag [May 23 15:35:17] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 37442df44804a8370b396dea59726037@209.191.44.130 - OPTIONS (No RTP) [May 23 15:35:17] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 15:35:17] 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=z9hG4bK44a3e801;rport;received=209.191.44.130 From: "asterisk" ;tag=as2ee08bd8 To: ;tag=as08f97ba6 Call-ID: 37442df44804a8370b396dea59726037@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 15:35:17] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '37442df44804a8370b396dea59726037@209.191.44.130' in 32000 ms (Method: OPTIONS) [May 23 15:35:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:17] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "7327049020";tag=as7a9f2f18 To: "Poly_test ENG";tag=82A90870-A5BD6FFB Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK588a9985 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=- 1306179293 1306179294 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 m=audio 51836 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 1 [ 62]: From: "7327049020";tag=as7a9f2f18 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK588a9985 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Supported: 100rel [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Supported: replaces [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 10 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Accept-Language: en [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 12 [ 29]: Content-Type: application/SDP [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 13 [ 19]: Content-Length: 165 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Header 14 [ 0]: [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306179293 1306179294 IN IP4 209.191.39.117 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 51836 RTP/AVP 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 6 [ 10]: a=sendrecv [May 23 15:35:17] DEBUG[13067] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: --- (14 headers 8 lines) --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking To) --From tag as7a9f2f18 --To-tag 82A90870-A5BD6FFB [May 23 15:35:17] DEBUG[13067] chan_sip.c: Acked pending invite 103 [May 23 15:35:17] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065036 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Stopping retransmission on '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' of Request 103: Match Found [May 23 15:35:17] DEBUG[13067] chan_sip.c: SIP response 200 to RE-invite on outgoing call 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 1306179293 1306179294 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 15:35:17] 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 15:35:17] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 15:35:17] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaab42f8' [May 23 15:35:17] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51836 [May 23 15:35:17] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xaab44a4 [May 23 15:35:17] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 15:35:17] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 15:35:17] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 15:35:17] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:17] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:17] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 2 (In use) [May 23 15:35:17] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '2' [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:17] DEBUG[13067] chan_sip.c: Strict routing enforced for session 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:17] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:17] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:17] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:17] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 15:35:17] 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=z9hG4bK705542cb Max-Forwards: 70 From: "7327049020" ;tag=as7a9f2f18 To: ;tag=82A90870-A5BD6FFB Contact: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 15:35:17] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:322' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:17] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000027~ Value: SIP 200 OK Uniqueid: 1306179315.38 [May 23 15:35:17] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:17] 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: 2 Paused: 0 [May 23 15:35:17] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '2' (In use) [May 23 15:35:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:18] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:19] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> INVITE sip:7327049020@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=82A90870-A5BD6FFB To: "7327049020";tag=as7a9f2f18 Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 1 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKe47b325eEF8CDA59 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=- 1306179293 1306179295 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendonly m=audio 51836 RTP/AVP 0 a=sendonly a=rtpmap:0 PCMU/8000 <-------------> [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 0 [ 47]: INVITE sip:7327049020@64.19.145.13:5060 SIP/2.0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 1 [102]: From: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 2 [ 60]: To: "7327049020";tag=as7a9f2f18 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKe47b325eEF8CDA59 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Supported: 100rel [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 13 [ 34]: Allow-Events: talk,hold,conference [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 14 [ 29]: Content-Type: application/SDP [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 15 [ 19]: Content-Length: 177 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 16 [ 0]: [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306179293 1306179295 IN IP4 209.191.39.117 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 5 [ 10]: a=sendonly [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 6 [ 23]: m=audio 51836 RTP/AVP 0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 7 [ 10]: a=sendonly [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:19] VERBOSE[13067] chan_sip.c: --- (16 headers 9 lines) --- [May 23 15:35:19] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking From) --From tag 82A90870-A5BD6FFB --To-tag as7a9f2f18 [May 23 15:35:19] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 15:35:19] DEBUG[13067] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel" [May 23 15:35:19] DEBUG[13067] sip/reqresp_parser.c: Found SIP option: -100rel- [May 23 15:35:19] DEBUG[13067] sip/reqresp_parser.c: Matched SIP option: 100rel [May 23 15:35:19] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:19] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 15:35:19] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 1306179293 1306179295 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [May 23 15:35:19] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 15:35:19] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP a=sendonly... OK. [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:19] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd39c [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:19] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd39c [May 23 15:35:19] 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 15:35:19] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 15:35:19] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaab42f8' [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51836 [May 23 15:35:19] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd39c to 0xaab44a4 [May 23 15:35:19] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 15:35:19] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 15:35:19] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaab42f8' [May 23 15:35:19] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:19] DEBUG[13067] chan_sip.c: Got a SIP re-invite for call 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:19] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:19] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 15:35:19] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 15:35:19] DEBUG[13109] manager.c: Examining event: Event: Hold Privilege: call,all Status: On Channel: SIP/322-eng-00000027 Uniqueid: 1306179315.39 [May 23 15:35:19] DEBUG[13067] chan_sip.c: SIP/322-eng-00000027: This call is UP.... [May 23 15:35:19] 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=z9hG4bKe47b325eEF8CDA59;received=209.191.39.117 From: "Poly_test ENG";tag=82A90870-A5BD6FFB To: "7327049020";tag=as7a9f2f18 Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@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 15:35:19] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Setting framing from config on incoming call [May 23 15:35:19] DEBUG[13067] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 15:35:19] DEBUG[13067] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Audio is at 5060 [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:19] DEBUG[13067] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:19] DEBUG[13067] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 15:35:19] 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=z9hG4bKe47b325eEF8CDA59;received=209.191.39.117 From: "Poly_test ENG";tag=82A90870-A5BD6FFB To: "7327049020";tag=as7a9f2f18 Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@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: 193 v=0 o=root 1146564904 1146564906 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 10046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=recvonly <------------> [May 23 15:35:19] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065038 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Sending reinvite on SIP '0089679f1d3712a573e92dbe03d33782@64.19.145.7' - It's audio soon redirected to IP 64.19.145.13:5060 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Strict routing enforced for session 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:19] VERBOSE[19579] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:19] DEBUG[19579] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:19] DEBUG[19579] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:19] VERBOSE[19579] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 15:35:19] 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 15:35:19] DEBUG[19579] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 15:35:19] DEBUG[19579] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 15:35:19] VERBOSE[19579] chan_sip.c: Audio is at 5060 [May 23 15:35:19] VERBOSE[19579] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:19] DEBUG[19579] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:19] DEBUG[19579] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 15:35:19] DEBUG[19579] chan_sip.c: Initializing already initialized SIP dialog 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (presumably reinvite) [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 0 [ 41]: INVITE sip:7327049020@64.19.145.7 SIP/2.0 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK114573b8;rport [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 3 [ 50]: From: ;tag=as3a9ecd75 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 4 [ 60]: To: "7327049020" ;tag=as22de05f6 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 5 [ 43]: Contact: [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 6 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 10 [ 19]: Supported: replaces [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 23 15:35:19] DEBUG[19579] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 15:35:19] 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 15:35:19] VERBOSE[19579] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: INVITE sip:7327049020@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK114573b8;rport Max-Forwards: 70 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Contact: Call-ID: 0089679f1d3712a573e92dbe03d33782@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 371132669 371132671 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 17626 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 15:35:19] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 15:35:19] 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 15:35:19] DEBUG[19579] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065039 [May 23 15:35:19] DEBUG[19579] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:19] DEBUG[19579] res_musiconhold.c: Music on Hold class 'default' not found in memory [May 23 15:35:19] DEBUG[19579] res_musiconhold.c: Music on Hold class 'default' not found in memory [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:19] 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=z9hG4bK114573b8;received=64.19.145.13;rport=5060 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Call-ID: 0089679f1d3712a573e92dbe03d33782@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 15:35:19] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK114573b8;received=64.19.145.13;rport=5060 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as3a9ecd75 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 3 [ 60]: To: "7327049020" ;tag=as22de05f6 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 9 [ 37]: Contact: [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:19] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:19] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking To) --From tag as3a9ecd75 --To-tag as22de05f6 [May 23 15:35:19] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1065039 - INVITE (got response) [May 23 15:35:19] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Request 103: Found [May 23 15:35:19] DEBUG[13067] chan_sip.c: SIP response 100 to RE-invite on outgoing call 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:19] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ~HASH~SIP_CAUSE~SIP/mg2-00000026~ Value: SIP 100 Trying Uniqueid: 1306179315.38 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:19] 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=z9hG4bK114573b8;received=64.19.145.13;rport=5060 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Call-ID: 0089679f1d3712a573e92dbe03d33782@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: 221 v=0 o=root 628746205 628746208 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 10046 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK114573b8;received=64.19.145.13;rport=5060 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as3a9ecd75 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 3 [ 60]: To: "7327049020" ;tag=as22de05f6 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 9 [ 37]: Contact: [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Content-Length: 221 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 12 [ 0]: [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 1 [ 45]: o=root 628746205 628746208 IN IP4 64.19.145.7 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 2 [ 40]: s=Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 3 [ 20]: c=IN IP4 64.19.145.7 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 5 [ 23]: m=audio 10046 RTP/AVP 0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 7 [ 25]: a=silenceSupp:off - - - - [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 8 [ 10]: a=ptime:20 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=sendrecv [May 23 15:35:19] VERBOSE[13067] chan_sip.c: --- (12 headers 10 lines) --- [May 23 15:35:19] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking To) --From tag as3a9ecd75 --To-tag as22de05f6 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Acked pending invite 103 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Stopping retransmission on '0089679f1d3712a573e92dbe03d33782@64.19.145.7' of Request 103: Match Found [May 23 15:35:19] DEBUG[13067] chan_sip.c: SIP response 200 to RE-invite on outgoing call 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP o=root 628746205 628746208 IN IP4 64.19.145.7... UNSUPPORTED. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Asterisk PBX SVN-branch-1.6.1-r230383M... UNSUPPORTED. [May 23 15:35:19] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:19] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 64.19.145.7... OK. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:19] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 15:35:19] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [May 23 15:35:19] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb69c1068' [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 64.19.145.7:10046 [May 23 15:35:19] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xb69c1214 [May 23 15:35:19] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 15:35:19] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 15:35:19] DEBUG[13067] chan_sip.c: Updating call counter for incoming call [May 23 15:35:19] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:19] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:19] DEBUG[13067] chan_sip.c: Strict routing enforced for session 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:19] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:19] DEBUG[13067] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:19] DEBUG[13067] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:19] VERBOSE[13067] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Transmitting (no NAT) to 64.19.145.7:5060: ACK sip:7327049020@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK146bb35e;rport Max-Forwards: 70 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Contact: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 15:35:19] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:732' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ~HASH~SIP_CAUSE~SIP/mg2-00000026~ Value: SIP 200 OK Uniqueid: 1306179315.38 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: INVITE [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> ACK sip:7327049020@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=82A90870-A5BD6FFB To: "7327049020";tag=as7a9f2f18 Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 1 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKeff8db35C98E5268 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 15:35:19] DEBUG[13067] chan_sip.c: Header 0 [ 44]: ACK sip:7327049020@64.19.145.13:5060 SIP/2.0 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 1 [102]: From: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 2 [ 60]: To: "7327049020";tag=as7a9f2f18 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKeff8db35C98E5268 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 15:35:19] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 15:35:19] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking From) --From tag 82A90870-A5BD6FFB --To-tag as7a9f2f18 [May 23 15:35:19] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 15:35:19] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065038 [May 23 15:35:19] DEBUG[13067] chan_sip.c: Stopping retransmission on '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' of Response 1: Match Found [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '535c930d43fb2e284d7c1bba4a3a61f6@209.191.44.130' [May 23 15:35:19] DEBUG[13067] chan_sip.c: Destroying SIP dialog 535c930d43fb2e284d7c1bba4a3a61f6@209.191.44.130 [May 23 15:35:19] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '535c930d43fb2e284d7c1bba4a3a61f6@209.191.44.130' Method: OPTIONS [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:19] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:19] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] 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=E7EA8417-AA13A95A To: Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 CSeq: 1 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfd8eadb3720321D6 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=- 1306179296 1306179296 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendrecv m=audio 51838 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 15:35:20] DEBUG[13067] chan_sip.c: Header 0 [ 46]: INVITE sip:312@64.19.145.13;user=phone SIP/2.0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 1 [ 69]: From: "Poly_test ENG";tag=E7EA8417-AA13A95A [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 2 [ 37]: To: [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfd8eadb3720321D6 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Supported: 100rel [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 13 [ 34]: Allow-Events: talk,hold,conference [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 14 [ 29]: Content-Type: application/SDP [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 15 [ 19]: Content-Length: 252 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 16 [ 0]: [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306179296 1306179296 IN IP4 209.191.39.117 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 5 [ 10]: a=sendrecv [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 6 [ 31]: m=audio 51838 RTP/AVP 0 110 127 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 8 [ 22]: a=rtpmap:110 iLBC/8000 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 9 [ 18]: a=fmtp:110 mode=30 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 10 [ 33]: a=rtpmap:127 telephone-event/8000 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: --- (16 headers 11 lines) --- [May 23 15:35:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 (Checking From) --From tag E7EA8417-AA13A95A --To-tag [May 23 15:35:20] DEBUG[13067] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 15:35:20] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for dd352991-ef95b5a4-7585dccf@10.0.15.105 - INVITE (No RTP) [May 23 15:35:20] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 15:35:20] DEBUG[13067] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel" [May 23 15:35:20] DEBUG[13067] sip/reqresp_parser.c: Found SIP option: -100rel- [May 23 15:35:20] DEBUG[13067] sip/reqresp_parser.c: Matched SIP option: 100rel [May 23 15:35:20] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:20] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 15:35:20] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Using INVITE request as basis request - dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Found peer '322-eng' for '322-eng' from 209.191.39.117:5060 [May 23 15:35:20] 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=z9hG4bKfd8eadb3720321D6;received=209.191.39.117 From: "Poly_test ENG";tag=E7EA8417-AA13A95A To: ;tag=as54e0d5dd Call-ID: dd352991-ef95b5a4-7585dccf@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="30becc87" Content-Length: 0 <------------> [May 23 15:35:20] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065040 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'dd352991-ef95b5a4-7585dccf@10.0.15.105' in 32000 ms (Method: INVITE) [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] 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=E7EA8417-AA13A95A To: ;tag=as54e0d5dd Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 CSeq: 1 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfd8eadb3720321D6 Content-Length: 0 <-------------> [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 0 [ 43]: ACK sip:312@64.19.145.13;user=phone SIP/2.0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 1 [ 69]: From: "Poly_test ENG";tag=E7EA8417-AA13A95A [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 2 [ 52]: To: ;tag=as54e0d5dd [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKfd8eadb3720321D6 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: --- (7 headers 0 lines) --- [May 23 15:35:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 (Checking From) --From tag E7EA8417-AA13A95A --To-tag as54e0d5dd [May 23 15:35:20] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065040 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Stopping retransmission on 'dd352991-ef95b5a4-7585dccf@10.0.15.105' of Response 1: Match Found [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] 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=E7EA8417-AA13A95A To: Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 CSeq: 2 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK14d0bd219E8B56D 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="30becc87",uri="sip:312@64.19.145.13;user=phone",response="c7d273803512f8594c665c614d8822c8",algorithm=MD5 Content-Type: application/SDP Content-Length: 252 v=0 o=- 1306179296 1306179296 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendrecv m=audio 51838 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 15:35:20] DEBUG[13067] chan_sip.c: Header 0 [ 46]: INVITE sip:312@64.19.145.13;user=phone SIP/2.0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 1 [ 69]: From: "Poly_test ENG";tag=E7EA8417-AA13A95A [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 2 [ 37]: To: [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 4 [ 14]: CSeq: 2 INVITE [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 5 [ 66]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK14d0bd219E8B56D [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Supported: 100rel [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 13 [ 34]: Allow-Events: talk,hold,conference [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 14 [170]: Authorization: Digest username="322-eng",realm="asterisk",nonce="30becc87",uri="sip:312@64.19.145.13;user=phone",response="c7d273803512f8594c665c614d8822c8",algorithm=MD5 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 15 [ 29]: Content-Type: application/SDP [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 16 [ 19]: Content-Length: 252 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 17 [ 0]: [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 1 [ 47]: o=- 1306179296 1306179296 IN IP4 209.191.39.117 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 5 [ 10]: a=sendrecv [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 6 [ 31]: m=audio 51838 RTP/AVP 0 110 127 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 8 [ 22]: a=rtpmap:110 iLBC/8000 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 9 [ 18]: a=fmtp:110 mode=30 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Body 10 [ 33]: a=rtpmap:127 telephone-event/8000 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: --- (17 headers 11 lines) --- [May 23 15:35:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 (Checking From) --From tag E7EA8417-AA13A95A --To-tag [May 23 15:35:20] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 15:35:20] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 15:35:20] DEBUG[13067] netsock2.c: Splitting '64.19.145.13' gives... [May 23 15:35:20] DEBUG[13067] netsock2.c: ...host '64.19.145.13' and port '(null)'. [May 23 15:35:20] DEBUG[13067] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 15:35:20] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:20] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 15:35:20] DEBUG[13067] chan_sip.c: Initializing initreq for method INVITE - callid dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Using INVITE request as basis request - dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Found peer '322-eng' for '322-eng' from 209.191.39.117:5060 [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6604568' [May 23 15:35:20] DEBUG[13067] res_rtp_asterisk.c: Allocated port 15438 for RTP instance '0xb6604568' [May 23 15:35:20] DEBUG[13067] rtp_engine.c: RTP instance '0xb6604568' is setup and ready to go [May 23 15:35:20] DEBUG[13067] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6604568' [May 23 15:35:20] VERBOSE[13067] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Setting NAT on RTP to Off [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 1306179296 1306179296 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [May 23 15:35:20] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 15:35:20] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd39c [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Found RTP audio format 110 [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Setting payload 110 based on m type on 0xb7cfd39c [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Found RTP audio format 127 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Found audio description format iLBC for ID 110 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 iLBC/8000... OK. [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=fmtp:110 mode=30... UNSUPPORTED. [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Found audio description format telephone-event for ID 127 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:127 telephone-event/8000... OK. [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd39c [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Incorporating payload 110 on 0xb7cfd39c [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Incorporating payload 127 on 0xb7cfd39c [May 23 15:35:20] 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 15:35:20] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [May 23 15:35:20] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6604568' [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51838 [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd39c to 0xb6604714 [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Copying payload 110 from 0xb7cfd39c to 0xb6604714 [May 23 15:35:20] DEBUG[13067] rtp_engine.c: Copying payload 127 from 0xb7cfd39c to 0xb6604714 [May 23 15:35:20] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x404 (ulaw|ilbc) [May 23 15:35:20] DEBUG[13067] chan_sip.c: Checking SIP call limits for device 322-eng [May 23 15:35:20] DEBUG[13067] chan_sip.c: Updating call counter for incoming call [May 23 15:35:20] DEBUG[13067] chan_sip.c: Call from peer '322-eng' is 2 out of 2147483647 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: Looking for 312 in test (domain 64.19.145.13) [May 23 15:35:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:20] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:20] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 15:35:20] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] 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 15:35:20] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/322-eng-00000028 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 322 CallerIDName: Poly_test ENG AccountCode: eng Exten: 312 Context: test Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13067] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 15:35:20] DEBUG[13067] chan_sip.c: *** Joint capabilities are 0x404 (ulaw|ilbc) [May 23 15:35:20] DEBUG[13067] chan_sip.c: *** Our capabilities are 0x404 (ulaw|ilbc) [May 23 15:35:20] DEBUG[13067] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 15:35:20] DEBUG[13067] chan_sip.c: This channel will not be able to handle video. [May 23 15:35:20] DEBUG[13067] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 15:35:20] DEBUG[13067] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: SIPURI Value: sip:322-eng@209.191.39.117:5060 Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13067] chan_sip.c: build_route: Contact hop: [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: SIPDOMAIN Value: 64.19.145.13 Uniqueid: 1306179320.40 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: list_route: hop: [May 23 15:35:20] DEBUG[13067] chan_sip.c: SIP/322-eng-00000028: New call is still down.... Trying... [May 23 15:35:20] 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=z9hG4bK14d0bd219E8B56D;received=209.191.39.117 From: "Poly_test ENG";tag=E7EA8417-AA13A95A To: Call-ID: dd352991-ef95b5a4-7585dccf@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 15:35:20] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: SIPCALLID Value: dd352991-ef95b5a4-7585dccf@10.0.15.105 Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:20] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 15:35:20] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/322-eng-00000028 Uniqueid: 1306179320.40 Channeltype: SIP SIPcallid: dd352991-ef95b5a4-7585dccf@10.0.15.105 SIPfullcontact: sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/322-eng-00000028 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 322 CallerIDName: Poly_test ENG Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 15:35:20] 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 15:35:20] DEBUG[19580] pbx.c: Launching 'Dial' [May 23 15:35:20] VERBOSE[19580] pbx.c: -- Executing [312@test:1] Dial("SIP/322-eng-00000028", "SIP/312-eng") in new stack [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/322-eng-00000028 Context: test Extension: 312 Priority: 1 Application: Dial AppData: SIP/312-eng Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALSTATUS Value: Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALEDPEERNUMBER Value: Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALEDPEERNAME Value: Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: ANSWEREDTIME Value: Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Allocating new SIP dialog for 35e52b6843d4d58f347bbb2f5bafe990@127.0.0.1:0 - INVITE (No RTP) [May 23 15:35:20] DEBUG[19580] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa6bdca0' [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALEDTIME Value: Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[19580] res_rtp_asterisk.c: Allocated port 14526 for RTP instance '0xa6bdca0' [May 23 15:35:20] DEBUG[19580] rtp_engine.c: RTP instance '0xa6bdca0' is setup and ready to go [May 23 15:35:20] DEBUG[19580] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa6bdca0' [May 23 15:35:20] VERBOSE[19580] netsock2.c: == Using SIP RTP CoS mark 5 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Setting NAT on RTP to On [May 23 15:35:20] DEBUG[19580] chan_sip.c: OBPROXY: Not applying OBproxy to this call [May 23 15:35:20] DEBUG[19580] acl.c: For destination '209.191.39.117', our source address is '64.19.145.13'. [May 23 15:35:20] DEBUG[19580] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:20] DEBUG[19580] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/312-eng-00000029 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 312 CallerIDName: SPA303 Cisco AccountCode: eng Exten: Context: test Uniqueid: 1306179320.41 [May 23 15:35:20] DEBUG[19580] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 15:35:20] DEBUG[19580] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [May 23 15:35:20] DEBUG[19580] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 23 15:35:20] DEBUG[19580] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 23 15:35:20] DEBUG[19580] chan_sip.c: This channel will not be able to handle video. [May 23 15:35:20] DEBUG[19580] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [May 23 15:35:20] DEBUG[19580] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: SIPCALLID Value: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 Uniqueid: 1306179320.41 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-00000029 Uniqueid: 1306179320.41 Channeltype: SIP SIPcallid: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-00000029 Channeltype: SIP SIPcallid: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 Peername: 312-eng [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: DIALEDPEERNUMBER Value: 312-eng Uniqueid: 1306179320.41 [May 23 15:35:20] DEBUG[19580] rtp_engine.c: Seeded SDP of 'SIP/312-eng-00000029' with that of 'SIP/322-eng-00000028' [May 23 15:35:20] DEBUG[19580] channel.c: Not copying variable DIALEDTIME. [May 23 15:35:20] DEBUG[19580] channel.c: Not copying variable ANSWEREDTIME. [May 23 15:35:20] DEBUG[19580] channel.c: Not copying variable DIALEDPEERNAME. [May 23 15:35:20] DEBUG[19580] channel.c: Not copying variable DIALEDPEERNUMBER. [May 23 15:35:20] DEBUG[19580] channel.c: Not copying variable DIALSTATUS. [May 23 15:35:20] DEBUG[19580] channel.c: Not copying variable SIPCALLID. [May 23 15:35:20] DEBUG[19580] channel.c: Not copying variable SIPDOMAIN. [May 23 15:35:20] DEBUG[19580] channel.c: Not copying variable SIPURI. [May 23 15:35:20] DEBUG[19580] chan_sip.c: Outgoing Call for 312-eng [May 23 15:35:20] DEBUG[19580] chan_sip.c: Updating call counter for outgoing call [May 23 15:35:20] DEBUG[19580] chan_sip.c: Call to peer '312-eng' is 1 out of 2147483647 [May 23 15:35:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 15:35:20] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 15:35:20] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 6 (Ringing) [May 23 15:35:20] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '6' [May 23 15:35:20] 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 15:35:20] 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 15:35:20] DEBUG[19580] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [May 23 15:35:20] DEBUG[19580] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 15:35:20] VERBOSE[19580] chan_sip.c: Audio is at 5060 [May 23 15:35:20] VERBOSE[19580] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:20] DEBUG[19580] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:20] DEBUG[19580] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 15:35:20] DEBUG[19580] chan_sip.c: Initializing initreq for method INVITE - callid 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 0 [ 72]: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK380287de;rport [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 3 [ 59]: From: "Poly_test ENG" ;tag=as56d98aff [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 4 [ 63]: To: [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 5 [ 36]: Contact: [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 6 [ 59]: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 19:35:20 GMT [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 11 [ 19]: Supported: replaces [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 12 [ 91]: Remote-Party-ID: "Poly_test ENG" ;party=calling;privacy=off;screen=no [May 23 15:35:20] DEBUG[19580] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [May 23 15:35:20] VERBOSE[19580] 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=z9hG4bK380287de;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as56d98aff To: Contact: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Date: Mon, 23 May 2011 19:35:20 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 1545612263 1545612263 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 14526 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 15:35:20] DEBUG[19580] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065043 [May 23 15:35:20] DEBUG[19580] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/322-eng-00000028 Destination: SIP/312-eng-00000029 CallerIDNum: 322 CallerIDName: Poly_test ENG UniqueID: 1306179320.40 DestUniqueID: 1306179320.41 Dialstring: 312-eng [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:20] VERBOSE[19580] app_dial.c: -- Called SIP/312-eng [May 23 15:35:20] 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 15:35:20] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '6' (Ringing) [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 100 Trying From: "Poly_test ENG";tag=as56d98aff To: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK380287de Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as56d98aff [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 2 [ 63]: To: [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK380287de [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 6 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: --- (8 headers 0 lines) --- [May 23 15:35:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 (Checking To) --From tag as56d98aff --To-tag [May 23 15:35:20] DEBUG[13067] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1065043 - INVITE (got response) [May 23 15:35:20] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Request 102: Found [May 23 15:35:20] DEBUG[13067] chan_sip.c: SIP response 100 to standard invite [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-00000029~ Value: SIP 100 Trying Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 180 Ringing From: "Poly_test ENG";tag=as56d98aff To: ;tag=6da1ed63e3f3bbf3i0 Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK380287de Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as56d98aff [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 2 [ 86]: To: ;tag=6da1ed63e3f3bbf3i0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK380287de [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 6 [ 82]: Contact: "SPA303 Cisco" [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [May 23 15:35:20] VERBOSE[13067] chan_sip.c: --- (9 headers 0 lines) --- [May 23 15:35:20] DEBUG[13067] chan_sip.c: = Looking for Call ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 (Checking To) --From tag as56d98aff --To-tag 6da1ed63e3f3bbf3i0 [May 23 15:35:20] DEBUG[13067] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Request 102: Found [May 23 15:35:20] DEBUG[13067] chan_sip.c: SIP response 180 to standard invite [May 23 15:35:20] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 15:35:20] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/312-eng-00000029 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 312 CallerIDName: SPA303 Cisco Uniqueid: 1306179320.41 [May 23 15:35:20] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 15:35:20] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 6 (Ringing) [May 23 15:35:20] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '6' [May 23 15:35:20] VERBOSE[19580] app_dial.c: -- SIP/312-eng-00000029 is ringing [May 23 15:35:20] DEBUG[19580] rtp_engine.c: Setting early bridge SDP of 'SIP/322-eng-00000028' with that of 'SIP/312-eng-00000029' [May 23 15:35:20] VERBOSE[19580] 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=z9hG4bK14d0bd219E8B56D;received=209.191.39.117 From: "Poly_test ENG";tag=E7EA8417-AA13A95A To: ;tag=as0868ad46 Call-ID: dd352991-ef95b5a4-7585dccf@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 15:35:20] DEBUG[19580] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] 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 15:35:20] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-00000029~ Value: SIP 180 Ringing Uniqueid: 1306179320.40 [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:20] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:20] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '6' (Ringing) [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:20] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:21] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:21] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Poly_test ENG";tag=as56d98aff To: ;tag=6da1ed63e3f3bbf3i0 Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK380287de 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=- 43241339 43241339 IN IP4 209.191.39.117 s=- c=IN IP4 209.191.39.117 t=0 0 m=audio 51840 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 15:35:22] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as56d98aff [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 2 [ 86]: To: ;tag=6da1ed63e3f3bbf3i0 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK380287de [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 6 [ 82]: Contact: "SPA303 Cisco" [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Supported: replaces [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 9 [ 61]: Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 10 [ 29]: Content-Type: application/SDP [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 11 [ 19]: Content-Length: 214 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 12 [ 0]: [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 1 [ 43]: o=- 43241339 43241339 IN IP4 209.191.39.117 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 2 [ 3]: s=- [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 5 [ 27]: m=audio 51840 RTP/AVP 0 101 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=ptime:20 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Body 10 [ 10]: a=sendrecv [May 23 15:35:22] VERBOSE[13067] chan_sip.c: --- (12 headers 11 lines) --- [May 23 15:35:22] DEBUG[13067] chan_sip.c: = Looking for Call ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 (Checking To) --From tag as56d98aff --To-tag 6da1ed63e3f3bbf3i0 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Acked pending invite 102 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Stopping retransmission on '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' of Request 102: Match Found [May 23 15:35:22] DEBUG[13067] chan_sip.c: SIP response 200 to standard invite [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 43241339 43241339 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [May 23 15:35:22] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 15:35:22] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:22] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:22] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 15:35:22] VERBOSE[13067] chan_sip.c: Found RTP audio format 101 [May 23 15:35:22] DEBUG[13067] rtp_engine.c: Setting payload 101 based on m type on 0xb7cfd50c [May 23 15:35:22] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:22] VERBOSE[13067] chan_sip.c: Found audio description format telephone-event for ID 101 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 15:35:22] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 15:35:22] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 15:35:22] DEBUG[13067] rtp_engine.c: Incorporating payload 101 on 0xb7cfd50c [May 23 15:35:22] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 15:35:22] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [May 23 15:35:22] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa6bdca0' [May 23 15:35:22] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51840 [May 23 15:35:22] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xa6bde4c [May 23 15:35:22] DEBUG[13067] rtp_engine.c: Copying payload 101 from 0xb7cfd50c to 0xa6bde4c [May 23 15:35:22] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 15:35:22] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 15:35:22] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 15:35:22] DEBUG[13067] chan_sip.c: build_route: Contact hop: "SPA303 Cisco" [May 23 15:35:22] VERBOSE[13067] chan_sip.c: list_route: hop: [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[13067] chan_sip.c: Strict routing enforced for session 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:22] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 15:35:22] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:22] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 15:35:22] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 2 (In use) [May 23 15:35:22] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '2' [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/312-eng-00000029 Channeltype: SIP Uniqueid: 1306179320.41 SIPcallid: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 SIPfullcontact: sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 Peername: 312-eng [May 23 15:35:22] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:22] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:22] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 15:35:22] 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=z9hG4bK19024165;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as56d98aff To: ;tag=6da1ed63e3f3bbf3i0 Contact: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 15:35:22] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:312' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:22] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 15:35:22] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 15:35:22] 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 15:35:22] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/312-eng-00000029 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 312 CallerIDName: SPA303 Cisco Uniqueid: 1306179320.41 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-00000029~ Value: SIP 200 OK Uniqueid: 1306179320.40 [May 23 15:35:22] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 2 (In use) [May 23 15:35:22] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '2' [May 23 15:35:22] 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 15:35:22] VERBOSE[19580] app_dial.c: -- SIP/312-eng-00000029 answered SIP/322-eng-00000028 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:22] DEBUG[19580] rtp_engine.c: Setting early bridge SDP of 'SIP/322-eng-00000028' with that of 'SIP/312-eng-00000029' [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:22] DEBUG[19580] chan_sip.c: SIP answering channel: SIP/322-eng-00000028 [May 23 15:35:22] DEBUG[19580] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:22] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1306179320.40 [May 23 15:35:22] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:22] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 15:35:22] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALEDPEERNAME Value: SIP/312-eng-00000029 Uniqueid: 1306179320.40 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALEDPEERNUMBER Value: 312-eng Uniqueid: 1306179320.40 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: BRIDGEPEER Value: SIP/312-eng-00000029 Uniqueid: 1306179320.40 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: BRIDGEPEER Value: SIP/322-eng-00000028 Uniqueid: 1306179320.41 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/322-eng-00000028 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 322 CallerIDName: Poly_test ENG Uniqueid: 1306179320.40 [May 23 15:35:22] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '2' (In use) [May 23 15:35:22] DEBUG[19580] chan_sip.c: Setting framing from config on incoming call [May 23 15:35:22] DEBUG[19580] chan_sip.c: ** Our capability: 0x404 (ulaw|ilbc) Video flag: True Text flag: True [May 23 15:35:22] DEBUG[19580] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 15:35:22] VERBOSE[19580] chan_sip.c: Audio is at 5060 [May 23 15:35:22] VERBOSE[19580] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:22] VERBOSE[19580] chan_sip.c: Adding codec 0x400 (ilbc) to SDP [May 23 15:35:22] DEBUG[19580] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:22] DEBUG[19580] chan_sip.c: Done building SDP. Settling with this capability: 0x404 (ulaw|ilbc) [May 23 15:35:22] VERBOSE[19580] 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=z9hG4bK14d0bd219E8B56D;received=209.191.39.117 From: "Poly_test ENG";tag=E7EA8417-AA13A95A To: ;tag=as0868ad46 Call-ID: dd352991-ef95b5a4-7585dccf@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 1242389390 1242389390 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 15438 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 15:35:22] 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 15:35:22] DEBUG[19580] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065046 [May 23 15:35:22] DEBUG[19580] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:22] 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 15:35:22] DEBUG[13109] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/312-eng-00000029 Uniqueid: 1306179320.41 AccountCode: eng OldAccountCode: eng [May 23 15:35:22] DEBUG[19580] features.c: bridge answer set, chan answer set [May 23 15:35:22] DEBUG[19580] features.c: Removing dialed interfaces datastore on SIP/312-eng-00000029 since we're bridging [May 23 15:35:22] DEBUG[19580] channel.c: setting peeraccount to eng for SIP/322-eng-00000028 from data on channel SIP/312-eng-00000029 [May 23 15:35:22] DEBUG[19580] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:22] DEBUG[19580] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:22] VERBOSE[19580] rtp_engine.c: -- Locally bridging SIP/322-eng-00000028 and SIP/312-eng-00000029 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/322-eng-00000028 Channel2: SIP/312-eng-00000029 Uniqueid1: 1306179320.40 Uniqueid2: 1306179320.41 CallerID1: 322 CallerID2: 312 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: BRIDGEPEER Value: SIP/312-eng-00000029 Uniqueid: 1306179320.40 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: BRIDGEPVTCALLID Value: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 Uniqueid: 1306179320.40 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: BRIDGEPEER Value: SIP/322-eng-00000028 Uniqueid: 1306179320.41 [May 23 15:35:22] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: BRIDGEPVTCALLID Value: dd352991-ef95b5a4-7585dccf@10.0.15.105 Uniqueid: 1306179320.41 [May 23 15:35:22] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '2' (In use) [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'dd352991-ef95b5a4-7585dccf@10.0.15.105' Method: INVITE [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:22] 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 15:35:22] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] 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=E7EA8417-AA13A95A To: ;tag=as0868ad46 Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 CSeq: 2 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK7e2ad74e20A06EC9 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 15:35:22] DEBUG[13067] chan_sip.c: Header 0 [ 37]: ACK sip:312@64.19.145.13:5060 SIP/2.0 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 1 [ 69]: From: "Poly_test ENG";tag=E7EA8417-AA13A95A [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 2 [ 52]: To: ;tag=as0868ad46 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 2 ACK [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK7e2ad74e20A06EC9 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 7 [ 85]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 15:35:22] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 15:35:22] DEBUG[13067] chan_sip.c: = Looking for Call ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 (Checking From) --From tag E7EA8417-AA13A95A --To-tag as0868ad46 [May 23 15:35:22] DEBUG[13067] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 15:35:22] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065046 [May 23 15:35:22] DEBUG[13067] chan_sip.c: Stopping retransmission on 'dd352991-ef95b5a4-7585dccf@10.0.15.105' of Response 2: Match Found [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'dd352991-ef95b5a4-7585dccf@10.0.15.105' Method: ACK [May 23 15:35:22] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:22] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: ACK [May 23 15:35:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'dd352991-ef95b5a4-7585dccf@10.0.15.105' Method: ACK [May 23 15:35:23] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] DEBUG[19579] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [May 23 15:35:23] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> REFER sip:7327049020@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=82A90870-A5BD6FFB To: "7327049020";tag=as7a9f2f18 Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 2 REFER Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKbcce289c41BB0B87 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 15:35:23] DEBUG[13067] chan_sip.c: Header 0 [ 46]: REFER sip:7327049020@64.19.145.13:5060 SIP/2.0 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 1 [102]: From: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 2 [ 60]: To: "7327049020";tag=as7a9f2f18 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 4 [ 13]: CSeq: 2 REFER [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKbcce289c41BB0B87 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 10 [146]: Refer-To: [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 11 [ 39]: Referred-By: [May 23 15:35:23] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 15:35:23] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 15:35:23] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking From) --From tag 82A90870-A5BD6FFB --To-tag as7a9f2f18 [May 23 15:35:23] DEBUG[13067] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [May 23 15:35:23] VERBOSE[13067] chan_sip.c: Call 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 got a SIP call transfer from caller: (REFER)! [May 23 15:35:23] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: SIPREFERRINGCONTEXT Value: test Uniqueid: 1306179315.38 [May 23 15:35:23] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: SIPREFERREDBYHDR Value: Uniqueid: 1306179315.38 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Attended transfer: Will use Replace-Call-ID : dd352991-ef95b5a4-7585dccf@10.0.15.105 F-tag: E7EA8417-AA13A95A T-tag: as0868ad46 [May 23 15:35:23] VERBOSE[13067] chan_sip.c: SIP transfer to extension 312@test by 322-eng@64.19.145.13 [May 23 15:35:23] DEBUG[13067] chan_sip.c: SIP attended transfer: Transferer channel SIP/322-eng-00000027, transferee channel SIP/mg2-00000026 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/mg2-00000026' [May 23 15:35:23] 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=z9hG4bKbcce289c41BB0B87;received=209.191.39.117 From: "Poly_test ENG";tag=82A90870-A5BD6FFB To: "7327049020";tag=as7a9f2f18 Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@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 15:35:23] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Looking for callid dd352991-ef95b5a4-7585dccf@10.0.15.105 (fromtag E7EA8417-AA13A95A totag as0868ad46) [May 23 15:35:23] DEBUG[13067] chan_sip.c: Matched INCOMING call - their tag is E7EA8417-AA13A95A Our tag is as0868ad46 [May 23 15:35:23] DEBUG[13067] chan_sip.c: SIP attended transfer: trying to bridge SIP/322-eng-00000028 and SIP/mg2-00000026 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Sip transfer:-------------------- [May 23 15:35:23] DEBUG[13109] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Attended Channel: SIP/322-eng-00000027 Uniqueid: 1306179315.39 SIP-Callid: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 TargetChannel: SIP/322-eng-00000028 TargetUniqueid: 1306179320.40 [May 23 15:35:23] DEBUG[13067] chan_sip.c: -- Transferer to PBX channel: SIP/322-eng-00000027 State Up [May 23 15:35:23] DEBUG[13067] chan_sip.c: -- Transferer to PBX second channel (target): SIP/322-eng-00000028 State Up [May 23 15:35:23] DEBUG[13067] chan_sip.c: -- Bridged call to transferee: SIP/mg2-00000026 State Up [May 23 15:35:23] DEBUG[13067] chan_sip.c: -- Bridged call to transfer target: SIP/312-eng-00000029 State Up [May 23 15:35:23] DEBUG[13067] chan_sip.c: -- END Sip transfer:-------------------- [May 23 15:35:23] DEBUG[13067] chan_sip.c: SIP transfer: Four channels to handle [May 23 15:35:23] DEBUG[13067] chan_sip.c: SIP transfer: trying to masquerade SIP/mg2-00000026 into SIP/322-eng-00000028 [May 23 15:35:23] DEBUG[13067] channel.c: Planning to masquerade channel SIP/mg2-00000026 into the structure of SIP/322-eng-00000028 [May 23 15:35:23] DEBUG[13067] channel.c: Done planning to masquerade channel SIP/mg2-00000026 into the structure of SIP/322-eng-00000028 [May 23 15:35:23] DEBUG[13067] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [May 23 15:35:23] DEBUG[13067] chan_sip.c: Strict routing enforced for session 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:23] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:23] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:23] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:23] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 15:35:23] 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=z9hG4bK4c29fba8 Max-Forwards: 70 From: "7327049020";tag=as7a9f2f18 To: "Poly_test ENG";tag=82A90870-A5BD6FFB Contact: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@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 15:35:23] DEBUG[13067] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065047 [May 23 15:35:23] DEBUG[13067] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:23] DEBUG[13067] channel.c: Set channel SIP/312-eng-00000029 to write format gsm [May 23 15:35:23] DEBUG[13067] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [May 23 15:35:23] DEBUG[13067] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [May 23 15:35:23] DEBUG[13067] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xa6bdca0' [May 23 15:35:23] DEBUG[13067] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [May 23 15:35:23] VERBOSE[13067] file.c: -- Playing 'beep.gsm' (language 'en') [May 23 15:35:24] DEBUG[13067] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 23 15:35:24] DEBUG[13067] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 23 15:35:24] DEBUG[13067] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 23 15:35:24] DEBUG[13067] channel.c: Set channel SIP/312-eng-00000029 to write format ulaw [May 23 15:35:24] DEBUG[19580] channel.c: Actually Masquerading SIP/mg2-00000026(6) into the structure of SIP/322-eng-00000028(6) [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/mg2-00000026 CloneState: Up Original: SIP/322-eng-00000028 OriginalState: Up [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/mg2-00000026 Newname: SIP/mg2-00000026 Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/322-eng-00000028 Newname: SIP/mg2-00000026 Uniqueid: 1306179320.40 [May 23 15:35:24] DEBUG[19580] chan_sip.c: SIP Fixup: New owner for dialogue dd352991-ef95b5a4-7585dccf@10.0.15.105: SIP/mg2-00000026 (Old parent: SIP/mg2-00000026) [May 23 15:35:24] DEBUG[19580] chan_sip.c: Hangup call SIP/mg2-00000026, SIP callid dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:24] DEBUG[19580] chan_sip.c: update_call_counter(322-eng) - decrement call limit counter on hangup [May 23 15:35:24] DEBUG[19580] chan_sip.c: Updating call counter for incoming call [May 23 15:35:24] DEBUG[19580] chan_sip.c: Call from peer '322-eng' removed from call limit 2147483647 [May 23 15:35:24] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:24] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:24] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 8 (On Hold) [May 23 15:35:24] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '8' [May 23 15:35:24] DEBUG[19580] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6604568' [May 23 15:35:24] VERBOSE[19580] chan_sip.c: Scheduling destruction of SIP dialog 'dd352991-ef95b5a4-7585dccf@10.0.15.105' in 32000 ms (Method: ACK) [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOS Value: ssrc=2104528582;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=2104528582;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 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: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 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: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: RTPAUDIOQOS Value: ssrc=61940957;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=61940957;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 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: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179315.38 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Strict routing enforced for session dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:24] VERBOSE[19580] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 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: 1306179315.39 [May 23 15:35:24] DEBUG[19580] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:24] DEBUG[19580] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:24] VERBOSE[19580] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 15:35:24] VERBOSE[19580] 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=z9hG4bK229eab44 Max-Forwards: 70 From: ;tag=as0868ad46 To: "Poly_test ENG";tag=E7EA8417-AA13A95A Call-ID: dd352991-ef95b5a4-7585dccf@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 15:35:24] DEBUG[19580] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065050 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Trying to put 'BYE sip:322' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOS Value: ssrc=2104528582;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/mg2-00000026 Newname: SIP/322-eng-00000028 Uniqueid: 1306179315.38 [May 23 15:35:24] 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 15:35:24] DEBUG[19580] channel.c: Putting channel SIP/mg2-00000026 in ulaw/ulaw formats [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/mg2-00000026 CallerIDNum: 7327049020 CallerIDName: 7327049020 Uniqueid: 1306179320.40 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [May 23 15:35:24] DEBUG[19580] chan_sip.c: SIP Fixup: New owner for dialogue 0089679f1d3712a573e92dbe03d33782@64.19.145.7: SIP/mg2-00000026 (Old parent: SIP/322-eng-00000028) [May 23 15:35:24] DEBUG[19580] channel.c: Released clone lock on 'SIP/322-eng-00000028' [May 23 15:35:24] DEBUG[19580] channel.c: Done Masquerading SIP/mg2-00000026 (6) [May 23 15:35:24] DEBUG[19580] res_rtp_asterisk.c: Changing ssrc from 1234294886 to 2133142003 due to a source change [May 23 15:35:24] DEBUG[19580] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [May 23 15:35:24] DEBUG[13067] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:24] DEBUG[19580] rtp_engine.c: rtp-engine-local-bridge: Oooh, something is weird, backing out [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/mg2-00000026 UniqueID: 1306179320.40 [May 23 15:35:24] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '8' (On Hold) [May 23 15:35:24] DEBUG[13067] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/312-eng-00000029 UniqueID: 1306179320.41 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' Method: REFER [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:24] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "7327049020";tag=as7a9f2f18 To: "Poly_test ENG";tag=82A90870-A5BD6FFB Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 104 NOTIFY Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4c29fba8 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Length: 0 <-------------> [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 1 [ 62]: From: "7327049020";tag=as7a9f2f18 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 2 [100]: To: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 104 NOTIFY [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: BRIDGEPEER Value: SIP/312-eng-00000029 Uniqueid: 1306179320.40 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4c29fba8 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 7 [ 17]: Event: refer;id=2 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Accept-Language: en [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:24] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:24] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking To) --From tag as7a9f2f18 --To-tag 82A90870-A5BD6FFB [May 23 15:35:24] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065047 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Stopping retransmission on '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' of Request 104: Match Found [May 23 15:35:24] VERBOSE[13067] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [May 23 15:35:24] DEBUG[13067] chan_sip.c: Got 200 OK on NOTIFY for transfer [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: BRIDGEPVTCALLID Value: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 Uniqueid: 1306179320.40 [May 23 15:35:24] DEBUG[19579] rtp_engine.c: Oooh, something is weird, backing out [May 23 15:35:24] DEBUG[19579] rtp_engine.c: Channel 'SIP/322-eng-00000028' Zombie cleardown from bridge [May 23 15:35:24] DEBUG[19579] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/322-eng-00000028, c1=SIP/322-eng-00000027, flags: Yes,Yes,No,No [May 23 15:35:24] DEBUG[19579] res_rtp_asterisk.c: Setting the marker bit due to a source update [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/322-eng-00000028 Channel2: SIP/322-eng-00000027 Uniqueid1: 1306179315.38 Uniqueid2: 1306179315.39 CallerID1: 322 CallerID2: 322 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: BRIDGEPEER Value: SIP/mg2-00000026 Uniqueid: 1306179320.41 [May 23 15:35:24] DEBUG[19579] channel.c: Bridge stops bridging channels SIP/322-eng-00000028 and SIP/322-eng-00000027 [May 23 15:35:24] DEBUG[19579] 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)}","${CDR(calltype)}" ' len 9) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '"7327049020" <7327049020>' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)}","${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)}","${CDR(calltype)}" ' len 8) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '7327049020' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)}","${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 15:35:24] DEBUG[19579] pbx.c: Function result is '7327049020' [May 23 15:35:24] DEBUG[19579] 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 15:35:24] DEBUG[19579] pbx.c: Function result is 'SIP/mg2-00000026' [May 23 15:35:24] DEBUG[19579] 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)}","${CDR(calltype)}" ' len 15) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'SIP/322-eng-00000027' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 10) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '2011-05-23 15:35:15' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)}","${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 15:35:24] DEBUG[19579] pbx.c: Function result is '2011-05-23 15:35:17' [May 23 15:35:24] DEBUG[19579] 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 15:35:24] DEBUG[19579] pbx.c: Function result is '2011-05-23 15:35:24' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(duration)' (from 'CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 13) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '9' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(billsec)' (from 'CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 12) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '7' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 16) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'ANSWERED' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 13) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '1306179315.38' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(SIPCALLID1)' (from 'CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(SIPCALLID2)' (from 'CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(CGPN)' (from 'CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(CDPN)' (from 'CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(CHRN)' (from 'CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[19579] pbx.c: Evaluating 'CDR(calltype)' (from 'CDR(calltype)}" ' len 13) [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: ANSWEREDTIME Value: 2 Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALEDTIME Value: 4 Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: BRIDGEPVTCALLID Value: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 Uniqueid: 1306179320.41 [May 23 15:35:24] VERBOSE[19580] rtp_engine.c: -- Locally bridging SIP/mg2-00000026 and SIP/312-eng-00000029 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Strict routing enforced for session 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:24] VERBOSE[19580] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:24] DEBUG[19580] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:24] DEBUG[19580] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:24] VERBOSE[19580] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 15:35:24] DEBUG[19580] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 23 15:35:24] DEBUG[19580] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [May 23 15:35:24] VERBOSE[19580] chan_sip.c: Audio is at 5060 [May 23 15:35:24] VERBOSE[19580] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [May 23 15:35:24] DEBUG[19580] chan_sip.c: -- Done with adding codecs to SDP [May 23 15:35:24] DEBUG[19580] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 15:35:24] DEBUG[19580] chan_sip.c: Initializing already initialized SIP dialog 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 (presumably reinvite) [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 0 [ 72]: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK0c1c56bd;rport [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 3 [ 59]: From: "Poly_test ENG" ;tag=as56d98aff [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 4 [ 86]: To: ;tag=6da1ed63e3f3bbf3i0 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 5 [ 36]: Contact: [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 6 [ 59]: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 10 [ 19]: Supported: replaces [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 11 [ 95]: Remote-Party-ID: "7327049020" ;party=calling;privacy=off;screen=no [May 23 15:35:24] DEBUG[19580] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [May 23 15:35:24] VERBOSE[19580] 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=z9hG4bK0c1c56bd;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as56d98aff To: ;tag=6da1ed63e3f3bbf3i0 Contact: Call-ID: 70d6a57f20eb408f45963b415a51938d@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: "7327049020" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 195 v=0 o=root 1545612263 1545612264 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 14526 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- [May 23 15:35:24] DEBUG[19580] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065051 [May 23 15:35:24] DEBUG[19580] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '2011-05-23 15:35:15' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '"7327049020" <7327049020>' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'mtt-from-outside' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'SIP/mg2-00000026' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'SIP/322-eng-00000027' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'Dial' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'SIP/322-eng' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '9' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '7' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'ANSWERED' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is 'DOCUMENTATION' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '1306179315.38' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[19579] pbx.c: Function result is '(null)' [May 23 15:35:24] DEBUG[19579] 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 15:35:15','"7327049020" <7327049020>','mtt-from-outside','SIP/mg2-00000026','SIP/322-eng-00000027','Dial','SIP/322-eng','9','7','ANSWERED','DOCUMENTATION','','1306179315.38','','') [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: ~HASH~SIP_CAUSE~SIP/322-eng-00000027~ Value: SIP 200 OK Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:24] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> BYE sip:7327049020@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=82A90870-A5BD6FFB To: "7327049020";tag=as7a9f2f18 Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 CSeq: 3 BYE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKbcbf44a1D8F45A5 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 0 [ 44]: BYE sip:7327049020@64.19.145.13:5060 SIP/2.0 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 1 [102]: From: "Poly_test ENG";tag=82A90870-A5BD6FFB [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 2 [ 60]: To: "7327049020";tag=as7a9f2f18 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 3 BYE [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 5 [ 66]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKbcbf44a1D8F45A5 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:24] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:24] DEBUG[13067] chan_sip.c: = Looking for Call ID: 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 (Checking From) --From tag 82A90870-A5BD6FFB --To-tag as7a9f2f18 [May 23 15:35:24] DEBUG[13067] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [May 23 15:35:24] DEBUG[13067] chan_sip.c: Initializing initreq for method BYE - callid 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:24] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:24] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 15:35:24] DEBUG[13067] chan_sip.c: Setting SIP_ALREADYGONE on dialog 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060 [May 23 15:35:24] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaab42f8' [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060' in 32000 ms (Method: BYE) [May 23 15:35:24] DEBUG[13067] chan_sip.c: Received bye, issuing owner hangup [May 23 15:35:24] 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=z9hG4bKbcbf44a1D8F45A5;received=209.191.39.117 From: "Poly_test ENG";tag=82A90870-A5BD6FFB To: "7327049020";tag=as7a9f2f18 Call-ID: 620aeb090c9fdc9e24631ed779a2bd88@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 15:35:24] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: RTPAUDIOQOS Value: ssrc=61940957;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179315.39 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 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: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 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: 1306179315.39 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000027 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179315.39 [May 23 15:35:24] 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=as0868ad46 To: "Poly_test ENG";tag=E7EA8417-AA13A95A Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 CSeq: 102 BYE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229eab44 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 15:35:24] DEBUG[13067] chan_sip.c: Header 0 [ 65]: SIP/2.0 400 SIP Parser Error : Unexpected '\"', line 9, column 99 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 1 [ 54]: From: ;tag=as0868ad46 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 2 [ 67]: To: "Poly_test ENG";tag=E7EA8417-AA13A95A [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 4 [ 13]: CSeq: 102 BYE [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229eab44 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 7 [ 47]: User-Agent: Asterisk PBX SVN-branch-1.8-r319997 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [May 23 15:35:24] 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 15:35:24] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 15:35:24] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 15:35:24] DEBUG[13067] chan_sip.c: = Looking for Call ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 (Checking To) --From tag as0868ad46 --To-tag E7EA8417-AA13A95A [May 23 15:35:24] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065050 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Stopping retransmission on 'dd352991-ef95b5a4-7585dccf@10.0.15.105' of Request 102: Match Found [May 23 15:35:24] VERBOSE[13067] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [May 23 15:35:24] 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 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:24] DEBUG[19579] channel.c: Hanging up channel 'SIP/322-eng-00000027' [May 23 15:35:24] DEBUG[19579] chan_sip.c: update_call_counter(322-eng) - decrement call limit counter on hangup [May 23 15:35:24] DEBUG[19579] chan_sip.c: Updating call counter for outgoing call [May 23 15:35:24] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:24] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:24] DEBUG[19579] chan_sip.c: Call to peer '322-eng' removed from call limit 2147483647 [May 23 15:35:24] DEBUG[19579] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 620aeb090c9fdc9e24631ed779a2bd88@64.19.145.13:5060. [May 23 15:35:24] DEBUG[19579] app_dial.c: Exiting with DIALSTATUS=ANSWER. [May 23 15:35:24] DEBUG[19579] pbx.c: Spawn extension (mtt-from-outside,7327049020,1) exited non-zero on 'SIP/322-eng-00000028' [May 23 15:35:24] VERBOSE[19579] pbx.c: == Spawn extension (mtt-from-outside, 7327049020, 1) exited non-zero on 'SIP/322-eng-00000028' [May 23 15:35:24] DEBUG[19579] channel.c: Soft-Hanging up channel 'SIP/322-eng-00000028' [May 23 15:35:24] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 1 (Not in use) [May 23 15:35:24] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '1' [May 23 15:35:24] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:24] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:24] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 1 (Not in use) [May 23 15:35:24] DEBUG[19579] channel.c: Hanging up zombie 'SIP/322-eng-00000028' [May 23 15:35:24] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '1' [May 23 15:35:24] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:24] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:24] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 1 (Not in use) [May 23 15:35:24] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '1' [May 23 15:35:24] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '1' (Not in use) [May 23 15:35:24] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 322-eng [May 23 15:35:24] DEBUG[13069] chan_sip.c: Checking device state for peer 322-eng [May 23 15:35:24] DEBUG[13069] devicestate.c: Changing state for SIP/322-eng - state 1 (Not in use) [May 23 15:35:24] DEBUG[13069] devicestate.c: device 'SIP/322-eng' state '1' [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/322-eng-00000027 Uniqueid: 1306179315.39 CallerIDNum: 322 CallerIDName: Poly_test ENG Cause: 16 Cause-txt: Normal Clearing [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/322-eng-00000028 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1306179315.38 [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/322-eng-00000028 UniqueID: 1306179315.38 DialStatus: ANSWER [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/322-eng-00000028 Uniqueid: 1306179315.38 CallerIDNum: 322 CallerIDName: Poly_test ENG Cause: 16 Cause-txt: Normal Clearing [May 23 15:35:24] 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: 1 Paused: 0 [May 23 15:35:24] 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 15:35:24] 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: 1 Paused: 0 [May 23 15:35:24] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '1' (Not in use) [May 23 15:35:24] 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 15:35:24] 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: 1 Paused: 0 [May 23 15:35:24] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '1' (Not in use) [May 23 15:35:24] 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: 1 Paused: 0 [May 23 15:35:24] DEBUG[13094] app_queue.c: Device 'SIP/322-eng' changed to state '1' (Not in use) [May 23 15:35:24] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Poly_test ENG";tag=as56d98aff To: ;tag=6da1ed63e3f3bbf3i0 Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0c1c56bd Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Content-Type: application/SDP Content-Length: 214 v=0 o=- 43241339 43241340 IN IP4 209.191.39.117 s=- c=IN IP4 209.191.39.117 t=0 0 m=audio 51840 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 15:35:24] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 1 [ 58]: From: "Poly_test ENG";tag=as56d98aff [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 2 [ 86]: To: ;tag=6da1ed63e3f3bbf3i0 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 5 [ 68]: Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK0c1c56bd [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 6 [ 82]: Contact: "SPA303 Cisco" [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 7 [ 26]: Server: Cisco/SPA303-7.4.6 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 8 [ 29]: Content-Type: application/SDP [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 9 [ 19]: Content-Length: 214 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Header 10 [ 0]: [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 0 [ 3]: v=0 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 1 [ 43]: o=- 43241339 43241340 IN IP4 209.191.39.117 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 2 [ 3]: s=- [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 3 [ 23]: c=IN IP4 209.191.39.117 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 4 [ 5]: t=0 0 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 5 [ 27]: m=audio 51840 RTP/AVP 0 101 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 9 [ 10]: a=ptime:20 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Body 10 [ 10]: a=sendrecv [May 23 15:35:24] VERBOSE[13067] chan_sip.c: --- (10 headers 11 lines) --- [May 23 15:35:24] DEBUG[13067] chan_sip.c: = Looking for Call ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 (Checking To) --From tag as56d98aff --To-tag 6da1ed63e3f3bbf3i0 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Acked pending invite 103 [May 23 15:35:24] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065051 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Stopping retransmission on '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' of Request 103: Match Found [May 23 15:35:24] DEBUG[13067] chan_sip.c: SIP response 200 to RE-invite on outgoing call 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing session-level SDP o=- 43241339 43241340 IN IP4 209.191.39.117... UNSUPPORTED. [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [May 23 15:35:24] DEBUG[13067] netsock2.c: Splitting '209.191.39.117' gives... [May 23 15:35:24] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '(null)'. [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing session-level SDP c=IN IP4 209.191.39.117... OK. [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Found RTP audio format 0 [May 23 15:35:24] DEBUG[13067] rtp_engine.c: Setting payload 0 based on m type on 0xb7cfd50c [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Found RTP audio format 101 [May 23 15:35:24] DEBUG[13067] rtp_engine.c: Setting payload 101 based on m type on 0xb7cfd50c [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Found audio description format PCMU for ID 0 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Found audio description format telephone-event for ID 101 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [May 23 15:35:24] DEBUG[13067] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [May 23 15:35:24] DEBUG[13067] rtp_engine.c: Incorporating payload 0 on 0xb7cfd50c [May 23 15:35:24] DEBUG[13067] rtp_engine.c: Incorporating payload 101 on 0xb7cfd50c [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [May 23 15:35:24] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa6bdca0' [May 23 15:35:24] VERBOSE[13067] chan_sip.c: Peer audio RTP is at port 209.191.39.117:51840 [May 23 15:35:24] DEBUG[13067] rtp_engine.c: Copying payload 0 from 0xb7cfd50c to 0xa6bde4c [May 23 15:35:24] DEBUG[13067] rtp_engine.c: Copying payload 101 from 0xb7cfd50c to 0xa6bde4c [May 23 15:35:24] DEBUG[13067] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 15:35:24] DEBUG[13067] chan_sip.c: We have an owner, now see if we need to change this call [May 23 15:35:24] DEBUG[13067] chan_sip.c: Updating call counter for outgoing call [May 23 15:35:24] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 15:35:24] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 15:35:24] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 2 (In use) [May 23 15:35:24] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '2' [May 23 15:35:24] DEBUG[13067] chan_sip.c: Strict routing enforced for session 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:24] VERBOSE[13067] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:24] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:24] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:24] VERBOSE[13067] chan_sip.c: set_destination: set destination to 209.191.39.117:5060 [May 23 15:35:24] 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=z9hG4bK09d56b86;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as56d98aff To: ;tag=6da1ed63e3f3bbf3i0 Contact: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Content-Length: 0 --- [May 23 15:35:24] DEBUG[13067] chan_sip.c: Trying to put 'ACK sip:312' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:24] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:24] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: ~HASH~SIP_CAUSE~SIP/312-eng-00000029~ Value: SIP 200 OK Uniqueid: 1306179320.41 [May 23 15:35:24] 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 15:35:24] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '2' (In use) [May 23 15:35:25] 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: 28745 NOTIFY Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-878f530b Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Contact: "SPA303 Cisco" Event: keep-alive Content-Length: 0 <-------------> [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 1 [ 68]: From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 2 [ 22]: To: [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 3 [ 38]: Call-ID: 5d07fe66-394bec48@10.0.15.101 [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 4 [ 18]: CSeq: 28745 NOTIFY [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 5 [ 60]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-878f530b [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 7 [ 30]: User-Agent: Cisco/SPA303-7.4.6 [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 8 [ 82]: Contact: "SPA303 Cisco" [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Event: keep-alive [May 23 15:35:25] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:25] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:25] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5d07fe66-394bec48@10.0.15.101 (Checking From) --From tag c7d0e91e95d40f0o0 --To-tag [May 23 15:35:25] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:25] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:25] 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-878f530b;received=209.191.39.117 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: ;tag=as1046500e Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 28745 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 15:35:25] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:25] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '5d07fe66-394bec48@10.0.15.101' in 32000 ms (Method: NOTIFY) [May 23 15:35:25] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:25] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:26] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:26] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:27] 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-2cd72885 From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-2cd72885 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 114430 NOTIFY [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:27] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:27] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:27] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:27] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:27] 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-2cd72885;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 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 15:35:27] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 15:35:27] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 15:35:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:27] 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-2cd72885 From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-2cd72885 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 114430 NOTIFY [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:27] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:27] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:27] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:27] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 114430, ours 114430) [May 23 15:35:27] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:27] 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-2cd72885;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 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 15:35:27] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 15:35:27] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 15:35:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:27] 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=z9hG4bK20125ec0;rport Max-Forwards: 70 From: "asterisk" ;tag=as75ce61ad To: Contact: Call-ID: 1862d3d17f9e64ae1d0dff24098a6b91@209.191.44.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 19:35:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK20125ec0;rport [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as75ce61ad [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 5 [ 38]: Contact: [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 6 [ 56]: Call-ID: 1862d3d17f9e64ae1d0dff24098a6b91@209.191.44.130 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 19:35:27 GMT [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 15:35:27] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 15:35:27] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 15:35:27] DEBUG[13067] chan_sip.c: = Looking for Call ID: 1862d3d17f9e64ae1d0dff24098a6b91@209.191.44.130 (Checking From) --From tag as75ce61ad --To-tag [May 23 15:35:27] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 15:35:27] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:27] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 1862d3d17f9e64ae1d0dff24098a6b91@209.191.44.130 - OPTIONS (No RTP) [May 23 15:35:27] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 15:35:27] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 15:35:27] 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=z9hG4bK20125ec0;rport;received=209.191.44.130 From: "asterisk" ;tag=as75ce61ad To: ;tag=as3bd674d2 Call-ID: 1862d3d17f9e64ae1d0dff24098a6b91@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 15:35:27] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 15:35:27] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '1862d3d17f9e64ae1d0dff24098a6b91@209.191.44.130' in 32000 ms (Method: OPTIONS) [May 23 15:35:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:27] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:28] 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-2cd72885 From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-2cd72885 [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 114430 NOTIFY [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:28] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:28] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:28] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:28] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:28] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 114430, ours 114430) [May 23 15:35:28] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:28] 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-2cd72885;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 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 15:35:28] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 15:35:28] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 15:35:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:28] DEBUG[13109] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 209.191.39.117:51841 OurSSRC: 2133142003 SentNTP: 1306179328.2706444288 SentRTP: 3520 SentPackets: 22 SentOctets: 3520 ReportBlock: FractionLost: 256 CumulativeLoss: 1 IAJitter: 0.0000 TheirLastSR: 0 DLSR: 65526.4200 (sec) [May 23 15:35:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:28] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:29] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '067516271a8542d86a086e064b02bfc3@209.191.44.130' [May 23 15:35:29] DEBUG[13067] chan_sip.c: Destroying SIP dialog 067516271a8542d86a086e064b02bfc3@209.191.44.130 [May 23 15:35:29] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '067516271a8542d86a086e064b02bfc3@209.191.44.130' Method: OPTIONS [May 23 15:35:29] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:29] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:29] 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=E7EA8417-AA13A95A To: ;tag=as0868ad46 Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 CSeq: 3 BYE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK9b4e18582ECEEB23 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="30becc87",uri="sip:312@64.19.145.13;user=phone",response="5a38626689e365fa6014fc9bfe11fbc9",algorithm=MD5 Content-Length: 0 <-------------> [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 0 [ 37]: BYE sip:312@64.19.145.13:5060 SIP/2.0 [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 1 [ 69]: From: "Poly_test ENG";tag=E7EA8417-AA13A95A [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 2 [ 52]: To: ;tag=as0868ad46 [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 3 [ 47]: Call-ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 4 [ 11]: CSeq: 3 BYE [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 5 [ 67]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK9b4e18582ECEEB23 [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 6 [ 68]: Contact: [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 8 [ 19]: Accept-Language: en [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 10 [170]: Authorization: Digest username="322-eng",realm="asterisk",nonce="30becc87",uri="sip:312@64.19.145.13;user=phone",response="5a38626689e365fa6014fc9bfe11fbc9",algorithm=MD5 [May 23 15:35:29] DEBUG[13067] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [May 23 15:35:29] VERBOSE[13067] chan_sip.c: --- (12 headers 0 lines) --- [May 23 15:35:29] DEBUG[13067] chan_sip.c: = Looking for Call ID: dd352991-ef95b5a4-7585dccf@10.0.15.105 (Checking From) --From tag E7EA8417-AA13A95A --To-tag as0868ad46 [May 23 15:35:29] DEBUG[13067] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [May 23 15:35:29] DEBUG[13067] chan_sip.c: Initializing initreq for method BYE - callid dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:29] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:29] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:29] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (no NAT) [May 23 15:35:29] DEBUG[13067] chan_sip.c: Setting SIP_ALREADYGONE on dialog dd352991-ef95b5a4-7585dccf@10.0.15.105 [May 23 15:35:29] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6604568' [May 23 15:35:29] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog 'dd352991-ef95b5a4-7585dccf@10.0.15.105' in 32000 ms (Method: BYE) [May 23 15:35:29] DEBUG[13067] chan_sip.c: Received bye, no owner, selfdestruct soon. [May 23 15:35:29] 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=z9hG4bK9b4e18582ECEEB23;received=209.191.39.117 From: "Poly_test ENG";tag=E7EA8417-AA13A95A To: ;tag=as0868ad46 Call-ID: dd352991-ef95b5a4-7585dccf@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 15:35:29] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:29] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:29] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:30] 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-2cd72885 From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-2cd72885 [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 114430 NOTIFY [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:30] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:30] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:30] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:30] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:30] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 114430, ours 114430) [May 23 15:35:30] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:30] 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-2cd72885;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 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 15:35:30] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 15:35:30] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 15:35:30] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' Method: INVITE [May 23 15:35:30] DEBUG[13067] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:31] VERBOSE[13067] chan_sip.c: <--- SIP read from UDP:209.191.39.117:5060 ---> BYE sip:322@64.19.145.13:5060 SIP/2.0 From: ;tag=6da1ed63e3f3bbf3i0 To: "Poly_test ENG";tag=as56d98aff Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 101 BYE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-66795669 Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 0 [ 37]: BYE sip:322@64.19.145.13:5060 SIP/2.0 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 1 [ 83]: From: ;tag=6da1ed63e3f3bbf3i0 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 2 [ 56]: To: "Poly_test ENG";tag=as56d98aff [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 3 [ 59]: Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 4 [ 13]: CSeq: 101 BYE [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 5 [ 60]: Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-66795669 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 7 [ 30]: User-Agent: Cisco/SPA303-7.4.6 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [May 23 15:35:31] VERBOSE[13067] chan_sip.c: --- (9 headers 0 lines) --- [May 23 15:35:31] DEBUG[13067] chan_sip.c: = Looking for Call ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 (Checking From) --From tag 6da1ed63e3f3bbf3i0 --To-tag as56d98aff [May 23 15:35:31] DEBUG[13067] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [May 23 15:35:31] DEBUG[13067] chan_sip.c: Initializing initreq for method BYE - callid 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:31] DEBUG[13067] netsock2.c: Splitting '209.191.39.117:5060' gives... [May 23 15:35:31] DEBUG[13067] netsock2.c: ...host '209.191.39.117' and port '5060'. [May 23 15:35:31] VERBOSE[13067] chan_sip.c: Sending to 209.191.39.117:5060 (NAT) [May 23 15:35:31] DEBUG[13067] chan_sip.c: Setting SIP_ALREADYGONE on dialog 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: RTPAUDIOQOS Value: ssrc=2133142003;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=22;rlp=0;rtt=0.000000 Uniqueid: 1306179320.41 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=2133142003;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=22;rlp=0;rtt=0.000000 Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 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: 1306179320.41 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179320.40 [May 23 15:35:31] DEBUG[13067] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa6bdca0' [May 23 15:35:31] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060' in 32000 ms (Method: BYE) [May 23 15:35:31] DEBUG[13067] chan_sip.c: Received bye, issuing owner hangup [May 23 15:35:31] VERBOSE[13067] chan_sip.c: <--- Transmitting (NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-66795669;received=209.191.39.117;rport=5060 From: ;tag=6da1ed63e3f3bbf3i0 To: "Poly_test ENG";tag=as56d98aff Call-ID: 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 CSeq: 101 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 15:35:31] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060 [May 23 15:35:31] DEBUG[19580] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a hangup [May 23 15:35:31] DEBUG[19580] channel.c: Returning from native bridge, channels: SIP/mg2-00000026, SIP/312-eng-00000029 [May 23 15:35:31] DEBUG[19580] 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 15:35:31] DEBUG[19580] pbx.c: Function result is '"Poly_test ENG" <322>' [May 23 15:35:31] DEBUG[19580] 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 15:35:31] DEBUG[19580] pbx.c: Function result is '322' [May 23 15:35:31] DEBUG[19580] 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)}","${CDR(calltype)}" ' len 8) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '312' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)}","${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)}","${CDR(calltype)}" ' len 12) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'SIP/322-eng-00000028' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)}","${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 15:35:31] DEBUG[19580] pbx.c: Function result is 'SIP/312-eng-00000029' [May 23 15:35:31] DEBUG[19580] 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 15:35:31] DEBUG[19580] pbx.c: Function result is '2011-05-23 15:35:20' [May 23 15:35:31] DEBUG[19580] 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)}","${CDR(calltype)}" ' len 11) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '2011-05-23 15:35:22' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 8) [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 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: 1306179320.41 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179320.41 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOS Value: ssrc=1031959375;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=1031959375;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179320.41 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 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: 1306179320.41 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 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: 1306179320.41 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/312-eng-00000029 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179320.41 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/mg2-00000026 Channel2: SIP/312-eng-00000029 Uniqueid1: 1306179320.40 Uniqueid2: 1306179320.41 CallerID1: 7327049020 CallerID2: 312 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: ANSWEREDTIME Value: 14 Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALEDTIME Value: 16 Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '2011-05-23 15:35:31' [May 23 15:35:31] DEBUG[19580] 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 15:35:31] DEBUG[19580] pbx.c: Function result is '11' [May 23 15:35:31] DEBUG[19580] 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 15:35:31] DEBUG[19580] pbx.c: Function result is '9' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)}","${CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 16) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'ANSWERED' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)}","${CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 13) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '1306179320.40' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(SIPCALLID1)' (from 'CDR(SIPCALLID1)}","${CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '(null)' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(SIPCALLID2)' (from 'CDR(SIPCALLID2)}","${CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 15) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '(null)' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(CGPN)' (from 'CDR(CGPN)}","${CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '(null)' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(CDPN)' (from 'CDR(CDPN)}","${CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '(null)' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(CHRN)' (from 'CDR(CHRN)}","${CDR(calltype)}" ' len 9) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '(null)' [May 23 15:35:31] DEBUG[19580] pbx.c: Evaluating 'CDR(calltype)' (from 'CDR(calltype)}" ' len 13) [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '(null)' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '2011-05-23 15:35:20' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '"Poly_test ENG" <322>' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'test' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'SIP/322-eng-00000028' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'SIP/312-eng-00000029' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'Dial' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'SIP/312-eng' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '11' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '9' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'ANSWERED' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'DOCUMENTATION' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is 'eng' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '1306179320.40' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '(null)' [May 23 15:35:31] DEBUG[19580] pbx.c: Function result is '(null)' [May 23 15:35:31] DEBUG[19580] 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 15:35:20','"Poly_test ENG" <322>','test','SIP/322-eng-00000028','SIP/312-eng-00000029','Dial','SIP/312-eng','11','9','ANSWERED','DOCUMENTATION','eng','1306179320.40','','') [May 23 15:35:31] DEBUG[19580] channel.c: Hanging up channel 'SIP/312-eng-00000029' [May 23 15:35:31] DEBUG[19580] chan_sip.c: Hangup call SIP/312-eng-00000029, SIP callid 70d6a57f20eb408f45963b415a51938d@64.19.145.13:5060 [May 23 15:35:31] DEBUG[19580] chan_sip.c: update_call_counter(312-eng) - decrement call limit counter on hangup [May 23 15:35:31] DEBUG[19580] chan_sip.c: Updating call counter for outgoing call [May 23 15:35:31] DEBUG[19580] chan_sip.c: Call to peer '312-eng' removed from call limit 2147483647 [May 23 15:35:31] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 15:35:31] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 15:35:31] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 1 (Not in use) [May 23 15:35:31] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '1' [May 23 15:35:31] DEBUG[19580] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa6bdca0' [May 23 15:35:31] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - 312-eng [May 23 15:35:31] DEBUG[13069] chan_sip.c: Checking device state for peer 312-eng [May 23 15:35:31] DEBUG[13069] devicestate.c: Changing state for SIP/312-eng - state 1 (Not in use) [May 23 15:35:31] DEBUG[13069] devicestate.c: device 'SIP/312-eng' state '1' [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/312-eng-00000029 Uniqueid: 1306179320.41 CallerIDNum: 312 CallerIDName: SPA303 Cisco Cause: 16 Cause-txt: Normal Clearing [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1306179320.40 [May 23 15:35:31] 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 15:35:31] DEBUG[13109] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/mg2-00000026 UniqueID: 1306179320.40 DialStatus: ANSWER [May 23 15:35:31] DEBUG[19580] app_dial.c: Exiting with DIALSTATUS=ANSWER. [May 23 15:35:31] DEBUG[19580] pbx.c: Spawn extension (test,312,1) exited non-zero on 'SIP/mg2-00000026' [May 23 15:35:31] VERBOSE[19580] pbx.c: == Spawn extension (test, 312, 1) exited non-zero on 'SIP/mg2-00000026' [May 23 15:35:31] DEBUG[19580] channel.c: Soft-Hanging up channel 'SIP/mg2-00000026' [May 23 15:35:31] DEBUG[19580] channel.c: Hanging up channel 'SIP/mg2-00000026' [May 23 15:35:31] DEBUG[19580] chan_sip.c: Hangup call SIP/mg2-00000026, SIP callid 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:31] DEBUG[19580] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb69c1068' [May 23 15:35:31] VERBOSE[19580] chan_sip.c: Scheduling destruction of SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' in 32000 ms (Method: ACK) [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOS Value: ssrc=1031959375;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 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: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/mg2-00000026 Variable: RTPAUDIOQOS Value: ssrc=1031959375;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1306179320.40 [May 23 15:35:31] DEBUG[19580] chan_sip.c: Strict routing enforced for session 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:31] VERBOSE[19580] chan_sip.c: set_destination: Parsing for address/port to send to [May 23 15:35:31] DEBUG[19580] netsock2.c: Splitting '64.19.145.7' gives... [May 23 15:35:31] DEBUG[19580] netsock2.c: ...host '64.19.145.7' and port '(null)'. [May 23 15:35:31] VERBOSE[19580] chan_sip.c: set_destination: set destination to 64.19.145.7:5060 [May 23 15:35:31] VERBOSE[19580] chan_sip.c: Reliably Transmitting (no NAT) to 64.19.145.7:5060: BYE sip:7327049020@64.19.145.7 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK34cd60af;rport Max-Forwards: 70 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r319997 Proxy-Authorization: Digest username="fsdev-mg2", realm="asterisk", algorithm=MD5, uri="64.19.145.13", nonce="", response="3a390f9cffa78c146215835dc3013e0f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 23 15:35:31] DEBUG[19580] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1065062 [May 23 15:35:31] DEBUG[19580] chan_sip.c: Trying to put 'BYE sip:732' onto UDP socket destined for 64.19.145.7:5060 [May 23 15:35:31] DEBUG[13109] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/mg2-00000026 Uniqueid: 1306179320.40 CallerIDNum: 7327049020 CallerIDName: 7327049020 Cause: 16 Cause-txt: Normal Clearing [May 23 15:35:31] DEBUG[13069] devicestate.c: No provider found, checking channel drivers for SIP - mg2 [May 23 15:35:31] 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=z9hG4bK34cd60af;received=64.19.145.13;rport=5060 From: ;tag=as3a9ecd75 To: "7327049020" ;tag=as22de05f6 Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 CSeq: 104 BYE Server: Asterisk PBX SVN-branch-1.6.1-r230383M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK34cd60af;received=64.19.145.13;rport=5060 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 2 [ 50]: From: ;tag=as3a9ecd75 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 3 [ 60]: To: "7327049020" ;tag=as22de05f6 [May 23 15:35:31] 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 15:35:31] DEBUG[13069] chan_sip.c: Checking device state for peer mg2 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 4 [ 53]: Call-ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 6 [ 46]: Server: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [May 23 15:35:31] DEBUG[13067] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [May 23 15:35:31] VERBOSE[13067] chan_sip.c: --- (10 headers 0 lines) --- [May 23 15:35:31] DEBUG[13067] chan_sip.c: = Looking for Call ID: 0089679f1d3712a573e92dbe03d33782@64.19.145.7 (Checking To) --From tag as3a9ecd75 --To-tag as22de05f6 [May 23 15:35:31] DEBUG[13067] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1065062 [May 23 15:35:31] DEBUG[13067] chan_sip.c: Stopping retransmission on '0089679f1d3712a573e92dbe03d33782@64.19.145.7' of Request 104: Match Found [May 23 15:35:31] DEBUG[13067] chan_sip.c: Destroying SIP dialog 0089679f1d3712a573e92dbe03d33782@64.19.145.7 [May 23 15:35:31] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '0089679f1d3712a573e92dbe03d33782@64.19.145.7' Method: ACK [May 23 15:35:31] DEBUG[13067] rtp_engine.c: Destroyed RTP instance '0xb69c1068' [May 23 15:35:31] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '1' (Not in use) [May 23 15:35:31] DEBUG[13069] devicestate.c: Changing state for SIP/mg2 - state 1 (Not in use) [May 23 15:35:31] DEBUG[13069] devicestate.c: device 'SIP/mg2' state '1' [May 23 15:35:31] 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 15:35:31] DEBUG[13094] app_queue.c: Device 'SIP/312-eng' changed to state '1' (Not in use) [May 23 15:35:31] 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 15:35:31] 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 15:35:34] 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-2cd72885 From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-2cd72885 [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 114430 NOTIFY [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:34] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:34] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:34] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:34] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:34] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 114430, ours 114430) [May 23 15:35:34] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:34] 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-2cd72885;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 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 15:35:34] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 15:35:34] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 15:35:37] 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=z9hG4bK58f2e03d;rport Max-Forwards: 70 From: "asterisk" ;tag=as6334bad7 To: Contact: Call-ID: 486a26325b23a222573e7e48355131f0@209.191.44.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M Date: Mon, 23 May 2011 19:35:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 0 [ 32]: OPTIONS sip:64.19.145.13 SIP/2.0 [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 209.191.44.130:5060;branch=z9hG4bK58f2e03d;rport [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as6334bad7 [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 4 [ 22]: To: [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 5 [ 38]: Contact: [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 6 [ 56]: Call-ID: 486a26325b23a222573e7e48355131f0@209.191.44.130 [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 8 [ 50]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r230383M [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 9 [ 35]: Date: Mon, 23 May 2011 19:35:37 GMT [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 23 15:35:37] DEBUG[13067] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [May 23 15:35:37] VERBOSE[13067] chan_sip.c: --- (13 headers 0 lines) --- [May 23 15:35:37] DEBUG[13067] chan_sip.c: = Looking for Call ID: 486a26325b23a222573e7e48355131f0@209.191.44.130 (Checking From) --From tag as6334bad7 --To-tag [May 23 15:35:37] DEBUG[13067] acl.c: For destination '209.191.44.130', our source address is '64.19.145.13'. [May 23 15:35:37] DEBUG[13067] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.19.145.13:5060 [May 23 15:35:37] DEBUG[13067] chan_sip.c: Allocating new SIP dialog for 486a26325b23a222573e7e48355131f0@209.191.44.130 - OPTIONS (No RTP) [May 23 15:35:37] DEBUG[13067] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [May 23 15:35:37] VERBOSE[13067] chan_sip.c: Looking for s in from-outside (domain 64.19.145.13) [May 23 15:35:37] 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=z9hG4bK58f2e03d;rport;received=209.191.44.130 From: "asterisk" ;tag=as6334bad7 To: ;tag=as55a3d35c Call-ID: 486a26325b23a222573e7e48355131f0@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 15:35:37] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.44.130:5060 [May 23 15:35:37] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '486a26325b23a222573e7e48355131f0@209.191.44.130' in 32000 ms (Method: OPTIONS) [May 23 15:35:38] 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-2cd72885 From: ;tag=5e35c995200173e1o3 To: Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 NOTIFY Max-Forwards: 70 Contact: Event: keep-alive User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 0 [ 31]: NOTIFY sip:64.19.145.13 SIP/2.0 [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187:5063;branch=z9hG4bK-2cd72885 [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 2 [ 55]: From: ;tag=5e35c995200173e1o3 [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 3 [ 22]: To: [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 4 [ 41]: Call-ID: 171efbf5-f832e501@192.168.15.187 [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 5 [ 19]: CSeq: 114430 NOTIFY [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 7 [ 42]: Contact: [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 8 [ 17]: Event: keep-alive [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.5(a) [May 23 15:35:38] DEBUG[13067] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [May 23 15:35:38] VERBOSE[13067] chan_sip.c: --- (11 headers 0 lines) --- [May 23 15:35:38] DEBUG[13067] chan_sip.c: = Looking for Call ID: 171efbf5-f832e501@192.168.15.187 (Checking From) --From tag 5e35c995200173e1o3 --To-tag [May 23 15:35:38] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:38] DEBUG[13067] chan_sip.c: Ignoring SIP message because of retransmit (NOTIFY Seqno 114430, ours 114430) [May 23 15:35:38] DEBUG[13067] chan_sip.c: Got NOTIFY Event: keep-alive [May 23 15:35:38] 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-2cd72885;received=209.191.13.243 From: ;tag=5e35c995200173e1o3 To: ;tag=as5970bba2 Call-ID: 171efbf5-f832e501@192.168.15.187 CSeq: 114430 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 15:35:38] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.13.243:5063 [May 23 15:35:38] VERBOSE[13067] chan_sip.c: Scheduling destruction of SIP dialog '171efbf5-f832e501@192.168.15.187' in 32000 ms (Method: NOTIFY) [May 23 15:35:39] DEBUG[13067] chan_sip.c: Auto destroying SIP dialog '5dfd2c573ee05f603c81e6f2657ef48d@209.191.44.130' [May 23 15:35:39] DEBUG[13067] chan_sip.c: Destroying SIP dialog 5dfd2c573ee05f603c81e6f2657ef48d@209.191.44.130 [May 23 15:35:39] VERBOSE[13067] chan_sip.c: Really destroying SIP dialog '5dfd2c573ee05f603c81e6f2657ef48d@209.191.44.130' Method: OPTIONS [May 23 15:35:40] DEBUG[13067] chan_sip.c: = Looking for Call ID: 5d07fe66-394bec48@10.0.15.101 (Checking From) --From tag c7d0e91e95d40f0o0 --To-tag [May 23 15:35:40] DEBUG[13067] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [May 23 15:35:40] DEBUG[13067] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 209.191.39.117:5060