[Oct 20 13:58:14] VERBOSE[27662] config.c: == Parsing '/etc/asterisk/logger.conf': [Oct 20 13:58:14] DEBUG[27662] config.c: Parsing /etc/asterisk/logger.conf [Oct 20 13:58:14] VERBOSE[27662] config.c: == Found [Oct 20 13:58:14] VERBOSE[27662] logger.c: Asterisk Queue Logger restarted [Oct 20 13:58:18] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog '7153536b-13be97aa-6bf33371@192.168.1.150' [Oct 20 13:58:18] DEBUG[24099] chan_sip.c: Destroying SIP dialog 7153536b-13be97aa-6bf33371@192.168.1.150 [Oct 20 13:58:18] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '7153536b-13be97aa-6bf33371@192.168.1.150' Method: REGISTER [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog '42d918115fe771a11517c2d9555b17e7@sip01.lon01.gagenetworks.net' [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Destroying SIP dialog 42d918115fe771a11517c2d9555b17e7@sip01.lon01.gagenetworks.net [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '42d918115fe771a11517c2d9555b17e7@sip01.lon01.gagenetworks.net' Method: OPTIONS [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> INVITE sip:3002@212.62.4.230:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKae42d80eB02D4E3 From: "3001" ;tag=8707368-3FCA15D5 To: CSeq: 1 INVITE Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 297 v=0 o=- 1287579514 1287579514 IN IP4 192.168.1.231 s=Polycom IP Phone c=IN IP4 192.168.1.231 t=0 0 a=sendrecv m=audio 10008 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-------------> [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 52]: INVITE sip:3002@212.62.4.230:5060;user=phone SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKae42d80eB02D4E3 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3001" ;tag=8707368-3FCA15D5 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 38]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 26]: Supported: 100rel,replaces [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 11 [ 34]: Allow-Events: talk,hold,conference [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 14 [ 19]: Content-Length: 297 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 15 [ 0]: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 0 [ 3]: v=0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 1 [ 46]: o=- 1287579514 1287579514 IN IP4 192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 5 [ 10]: a=sendrecv [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 6 [ 34]: m=audio 10008 RTP/AVP 9 0 8 18 127 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 7 [ 20]: a=rtpmap:9 G722/8000 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 12 [ 33]: a=rtpmap:127 telephone-event/8000 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: --- (15 headers 13 lines) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: = Looking for Call ID: 257253e4-96ab661-410d306a@192.168.1.231 (Checking From) --From tag 8707368-3FCA15D5 --To-tag [Oct 20 13:58:21] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:17261 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] VERBOSE[24099] netsock.c: == Using UDPTL CoS mark 5 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting NAT on UDPTL to On [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 257253e4-96ab661-410d306a@192.168.1.231 - INVITE (No RTP) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 20 13:58:21] DEBUG[24099] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,replaces" [Oct 20 13:58:21] DEBUG[24099] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 20 13:58:21] DEBUG[24099] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 20 13:58:21] DEBUG[24099] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 20 13:58:21] DEBUG[24099] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method INVITE - callid 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Using INVITE request as basis request - 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '3001' AND host = 'dynamic' [Oct 20 13:58:21] WARNING[24099] chan_sip.c: Unknown dtmf mode '' on line 0, using rfc2833 [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '195.59.152.66' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '195.59.152.66' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 28e7bd7c278e40b94f3313d92eec9000@127.0.0.1:0 - OPTIONS (No RTP) [Oct 20 13:58:21] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method OPTIONS - callid 157c02b66251dcfa7b51022377ef1366@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 38]: OPTIONS sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK40ea853a;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as2418caea [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 157c02b66251dcfa7b51022377ef1366@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:58:21 GMT [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK40ea853a;rport Max-Forwards: 70 From: "asterisk" ;tag=as2418caea To: Contact: Call-ID: 157c02b66251dcfa7b51022377ef1366@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81014 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 195.59.152.66:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK450f669b;rport Max-Forwards: 70 Route: From: "asterisk" ;tag=as5f938435 To: ;tag=551EA012-C924FAD7 Contact: Call-ID: e928da31-250edcfa-a47cd01f@192.168.1.231 CSeq: 280 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81016 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: -REALTIME- loading peer from database to memory. Name: 3001. Peer objects: -28 [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE host = '195.59.152.66' AND port = '17261' [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE ipaddr = '195.59.152.66' AND port = '17261' [Oct 20 13:58:21] WARNING[24099] chan_sip.c: Unknown dtmf mode '' on line 0, using rfc2833 [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '195.59.152.66' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '195.59.152.66' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 20 13:58:21] NOTICE[24099] chan_sip.c: Still have a QUALIFY dialog active, deleting [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Destroying SIP dialog 157c02b66251dcfa7b51022377ef1366@212.62.4.230:5060 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '157c02b66251dcfa7b51022377ef1366@212.62.4.230:5060' Method: OPTIONS [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 3a1bd8620bf38d33343646d5419d2bb1@127.0.0.1:0 - OPTIONS (No RTP) [Oct 20 13:58:21] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method OPTIONS - callid 134be32f077ed6e807545b476a782103@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 38]: OPTIONS sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1e5a2d30;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as272ded37 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 134be32f077ed6e807545b476a782103@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:58:21 GMT [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1e5a2d30;rport Max-Forwards: 70 From: "asterisk" ;tag=as272ded37 To: Contact: Call-ID: 134be32f077ed6e807545b476a782103@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81018 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 28e28c731c247cfa14e09fe04f489b78@127.0.0.1:0 - NOTIFY (No RTP) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 20 13:58:21] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '33da456233a3bc88721a129540f85ce2@212.62.4.230:5060' in 6400 ms (Method: NOTIFY) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method NOTIFY - callid 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 37]: NOTIFY sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK14ef7a9b;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as57468f5e [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 16]: CSeq: 102 NOTIFY [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 48]: Content-Type: application/simple-message-summary [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK14ef7a9b;rport Max-Forwards: 70 From: "asterisk" ;tag=as57468f5e To: Contact: Call-ID: 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81021 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: -REALTIME- loading peer from database to memory. Name: 3001. Peer objects: -28 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting NAT on UDPTL to On [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: <--- Reliably Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKae42d80eB02D4E3;received=195.59.152.66;rport=17261 From: "3001" ;tag=8707368-3FCA15D5 To: ;tag=as6b91fc09 Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61d2824b" Content-Length: 0 <------------> [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81023 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '257253e4-96ab661-410d306a@192.168.1.231' in 6400 ms (Method: INVITE) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK450f669b;rport From: "asterisk" ;tag=as5f938435 To: "3001" ;tag=551EA012-C924FAD7 CSeq: 280 NOTIFY Call-ID: e928da31-250edcfa-a47cd01f@192.168.1.231 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Content-Length: 0 <-------------> [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK450f669b;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as5f938435 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 57]: To: "3001" ;tag=551EA012-C924FAD7 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 16]: CSeq: 280 NOTIFY [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: e928da31-250edcfa-a47cd01f@192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 22]: Event: message-summary [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: --- (11 headers 0 lines) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: = Looking for Call ID: e928da31-250edcfa-a47cd01f@192.168.1.231 (Checking To) --From tag as5f938435 --To-tag 551EA012-C924FAD7 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81016 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Stopping retransmission on 'e928da31-250edcfa-a47cd01f@192.168.1.231' of Request 280: Match Found [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> ACK sip:3002@212.62.4.230:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKae42d80eB02D4E3 From: "3001" ;tag=8707368-3FCA15D5 To: ;tag=as6b91fc09 CSeq: 1 ACK Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 49]: ACK sip:3002@212.62.4.230:5060;user=phone SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKae42d80eB02D4E3 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3001" ;tag=8707368-3FCA15D5 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 53]: To: ;tag=as6b91fc09 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: --- (12 headers 0 lines) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: = Looking for Call ID: 257253e4-96ab661-410d306a@192.168.1.231 (Checking From) --From tag 8707368-3FCA15D5 --To-tag as6b91fc09 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81023 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Stopping retransmission on '257253e4-96ab661-410d306a@192.168.1.231' of Response 1: Match Found [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> INVITE sip:3002@212.62.4.230:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320 From: "3001" ;tag=8707368-3FCA15D5 To: CSeq: 2 INVITE Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="3001", realm="asterisk", nonce="61d2824b", uri="sip:3002@212.62.4.230:5060;user=phone", response="68a84da083507a9e814960dc9716f103", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 297 v=0 o=- 1287579514 1287579514 IN IP4 192.168.1.231 s=Polycom IP Phone c=IN IP4 192.168.1.231 t=0 0 a=sendrecv m=audio 10008 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-------------> [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 52]: INVITE sip:3002@212.62.4.230:5060;user=phone SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3001" ;tag=8707368-3FCA15D5 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 38]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 14]: CSeq: 2 INVITE [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 26]: Supported: 100rel,replaces [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 11 [ 34]: Allow-Events: talk,hold,conference [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 12 [178]: Authorization: Digest username="3001", realm="asterisk", nonce="61d2824b", uri="sip:3002@212.62.4.230:5060;user=phone", response="68a84da083507a9e814960dc9716f103", algorithm=MD5 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 15 [ 19]: Content-Length: 297 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 16 [ 0]: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 0 [ 3]: v=0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 1 [ 46]: o=- 1287579514 1287579514 IN IP4 192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 5 [ 10]: a=sendrecv [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 6 [ 34]: m=audio 10008 RTP/AVP 9 0 8 18 127 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 7 [ 20]: a=rtpmap:9 G722/8000 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Body 12 [ 33]: a=rtpmap:127 telephone-event/8000 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: --- (16 headers 13 lines) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: = Looking for Call ID: 257253e4-96ab661-410d306a@192.168.1.231 (Checking From) --From tag 8707368-3FCA15D5 --To-tag [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method INVITE - callid 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Using INVITE request as basis request - 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '3001' AND host = 'dynamic' [Oct 20 13:58:21] WARNING[24099] chan_sip.c: Unknown dtmf mode '' on line 0, using rfc2833 [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '195.59.152.66' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '195.59.152.66' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 20 13:58:21] NOTICE[24099] chan_sip.c: Still have a QUALIFY dialog active, deleting [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Destroying SIP dialog 134be32f077ed6e807545b476a782103@212.62.4.230:5060 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '134be32f077ed6e807545b476a782103@212.62.4.230:5060' Method: OPTIONS [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 781a94577a23c62833a536bf79e28316@127.0.0.1:0 - OPTIONS (No RTP) [Oct 20 13:58:21] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method OPTIONS - callid 2951f72f6b7b01a02365b1c23da1731d@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 38]: OPTIONS sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1808ed8a;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as11ab8080 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 2951f72f6b7b01a02365b1c23da1731d@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:58:21 GMT [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1808ed8a;rport Max-Forwards: 70 From: "asterisk" ;tag=as11ab8080 To: Contact: Call-ID: 2951f72f6b7b01a02365b1c23da1731d@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81025 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 4b2f7052650b08e2078cca2167a45dc1@127.0.0.1:0 - NOTIFY (No RTP) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 20 13:58:21] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060' in 6400 ms (Method: NOTIFY) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method NOTIFY - callid 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 37]: NOTIFY sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK28f11555;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as1705a649 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 16]: CSeq: 102 NOTIFY [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 48]: Content-Type: application/simple-message-summary [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK28f11555;rport Max-Forwards: 70 From: "asterisk" ;tag=as1705a649 To: Contact: Call-ID: 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81028 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: -REALTIME- loading peer from database to memory. Name: 3001. Peer objects: -28 [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE host = '195.59.152.66' AND port = '17261' [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE ipaddr = '195.59.152.66' AND port = '17261' [Oct 20 13:58:21] WARNING[24099] chan_sip.c: Unknown dtmf mode '' on line 0, using rfc2833 [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '195.59.152.66' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '195.59.152.66' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 20 13:58:21] NOTICE[24099] chan_sip.c: Still have a QUALIFY dialog active, deleting [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Destroying SIP dialog 2951f72f6b7b01a02365b1c23da1731d@212.62.4.230:5060 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '2951f72f6b7b01a02365b1c23da1731d@212.62.4.230:5060' Method: OPTIONS [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 727b13be1800f5c05bbdccec3a9c6fa8@127.0.0.1:0 - OPTIONS (No RTP) [Oct 20 13:58:21] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method OPTIONS - callid 40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 38]: OPTIONS sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK2c7ebee6;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as0a8443a8 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:58:21 GMT [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK2c7ebee6;rport Max-Forwards: 70 From: "asterisk" ;tag=as0a8443a8 To: Contact: Call-ID: 40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81030 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 7822f8b720563c5033d6032d20636e13@127.0.0.1:0 - NOTIFY (No RTP) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 20 13:58:21] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '0d4e4944211af9116758fc4263785e40@212.62.4.230:5060' in 6400 ms (Method: NOTIFY) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Initializing initreq for method NOTIFY - callid 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 37]: NOTIFY sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK48bc47b6;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as277261f1 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 16]: CSeq: 102 NOTIFY [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 48]: Content-Type: application/simple-message-summary [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK48bc47b6;rport Max-Forwards: 70 From: "asterisk" ;tag=as277261f1 To: Contact: Call-ID: 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81033 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: -REALTIME- loading peer from database to memory. Name: 3001. Peer objects: -28 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting NAT on UDPTL to On [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x80859b8' [Oct 20 13:58:21] DEBUG[24099] res_rtp_asterisk.c: Allocated port 19602 for RTP instance '0x80859b8' [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: RTP instance '0x80859b8' is setup and ready to go [Oct 20 13:58:21] DEBUG[24099] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x80859b8' [Oct 20 13:58:21] VERBOSE[24099] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting NAT on RTP to On [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Setting NAT on UDPTL to On [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing session-level SDP o=- 1287579514 1287579514 IN IP4 192.168.1.231... UNSUPPORTED. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [Oct 20 13:58:21] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:21] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.231... OK. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found RTP audio format 9 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Setting payload 9 based on m type on 0x402ecda0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found RTP audio format 0 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Setting payload 0 based on m type on 0x402ecda0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found RTP audio format 8 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Setting payload 8 based on m type on 0x402ecda0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found RTP audio format 18 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Setting payload 18 based on m type on 0x402ecda0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found RTP audio format 127 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found audio description format G722 for ID 9 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found audio description format PCMU for ID 0 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found audio description format G729 for ID 18 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Found audio description format telephone-event for ID 127 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:127 telephone-event/8000... OK. [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Incorporating payload 0 on 0x402ecda0 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Incorporating payload 8 on 0x402ecda0 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Incorporating payload 9 on 0x402ecda0 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Incorporating payload 18 on 0x402ecda0 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Incorporating payload 127 on 0x402ecda0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 20 13:58:21] DEBUG[24099] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x80859b8' [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Peer audio RTP is at port 192.168.1.231:10008 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Copying payload 0 from 0x402ecda0 to 0x8085b80 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Copying payload 8 from 0x402ecda0 to 0x8085b80 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Copying payload 9 from 0x402ecda0 to 0x8085b80 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Copying payload 18 from 0x402ecda0 to 0x8085b80 [Oct 20 13:58:21] DEBUG[24099] rtp_engine.c: Copying payload 127 from 0x402ecda0 to 0x8085b80 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Peer doesn't provide T.38 UDPTL [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Checking SIP call limits for device 3001 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Updating call counter for incoming call [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Call from peer '3001' is 1 out of 2147483647 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Looking for 3002 in DLPN_All (domain 212.62.4.230:5060) [Oct 20 13:58:21] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3001 [Oct 20 13:58:21] DEBUG[24088] chan_sip.c: Checking device state for peer 3001 [Oct 20 13:58:21] DEBUG[24088] devicestate.c: Changing state for SIP/3001 - state 2 (In use) [Oct 20 13:58:21] DEBUG[24088] devicestate.c: device 'SIP/3001' state '2' [Oct 20 13:58:21] DEBUG[24123] app_queue.c: Device 'SIP/3001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '3002' AND context = 'DLPN_All' AND priority = '1' [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'DLPN_All' AND priority = '1' ORDER BY exten [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: This channel will not be able to handle video. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '3002' AND context = 'DLPN_All' AND priority = '1' [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'DLPN_All' AND priority = '1' ORDER BY exten [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP/3001-000000ac: New call is still down.... Trying... [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320;received=195.59.152.66;rport=17261 From: "3001" ;tag=8707368-3FCA15D5 To: Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:21] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3001 [Oct 20 13:58:21] DEBUG[24088] chan_sip.c: Checking device state for peer 3001 [Oct 20 13:58:21] DEBUG[24088] devicestate.c: Changing state for SIP/3001 - state 2 (In use) [Oct 20 13:58:21] DEBUG[24088] devicestate.c: device 'SIP/3001' state '2' [Oct 20 13:58:21] DEBUG[24123] app_queue.c: Device 'SIP/3001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '3002' AND context = 'DLPN_All' AND priority = '1' [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'DLPN_All' AND priority = '1' ORDER BY exten [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '3002' AND context = 'DLPN_All' AND priority = '1' [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'DLPN_All' AND priority = '1' ORDER BY exten [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '3002' AND context = 'DLPN_All' AND priority = '1' [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'DLPN_All' AND priority = '1' ORDER BY exten [Oct 20 13:58:21] DEBUG[27665] pbx.c: Result of 'EXTEN' is '3002' [Oct 20 13:58:21] VERBOSE[27665] pbx_realtime.c: -- Executing [3002@DLPN_All:1] Gosub("SIP/3001-000000ac", "internal,3002,1") [Oct 20 13:58:21] DEBUG[27665] app_stack.c: Channel SIP/3001-000000ac has no datastore, so we're allocating one. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '3002' AND context = 'internal' AND priority = '1' [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '3002' AND context = 'internal' AND priority = '1' [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '3002' AND context = 'internal' AND priority = '1' [Oct 20 13:58:21] VERBOSE[27665] pbx_realtime.c: -- Executing [3002@internal:1] Dial("SIP/3001-000000ac", "SIP/3002") [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Oct 20 13:58:21] VERBOSE[27665] netsock.c: == Using UDPTL CoS mark 5 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Allocating new SIP dialog for 3f0213af3816da592f4ab06c5d1e5fb2@127.0.0.1:0 - INVITE (No RTP) [Oct 20 13:58:21] DEBUG[27665] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7e97338' [Oct 20 13:58:21] DEBUG[27665] res_rtp_asterisk.c: Allocated port 13900 for RTP instance '0x7e97338' [Oct 20 13:58:21] DEBUG[27665] rtp_engine.c: RTP instance '0x7e97338' is setup and ready to go [Oct 20 13:58:21] DEBUG[27665] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7e97338' [Oct 20 13:58:21] VERBOSE[27665] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Setting NAT on RTP to On [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Setting NAT on UDPTL to On [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 20 13:58:21] DEBUG[27665] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Target address 195.59.152.66:40452 is not local, substituting externaddr [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: This channel will not be able to handle video. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '' AND context = 'DLPN_All' AND priority = '1' [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:21] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'DLPN_All' AND priority = '1' ORDER BY exten [Oct 20 13:58:21] DEBUG[27665] rtp_engine.c: Seeded SDP of 'SIP/3002-000000ad' with that of 'SIP/3001-000000ac' [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable DIALEDTIME. [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable ANSWEREDTIME. [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable DIALEDPEERNAME. [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable DIALEDPEERNUMBER. [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable DIALSTATUS. [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable ARGC. [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable SIPCALLID. [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable SIPDOMAIN. [Oct 20 13:58:21] DEBUG[27665] channel.c: Not copying variable SIPURI. [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Outgoing Call for 3002 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Updating call counter for outgoing call [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Call to peer '3002' is 1 out of 2147483647 [Oct 20 13:58:21] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3002 [Oct 20 13:58:21] DEBUG[24088] chan_sip.c: Checking device state for peer 3002 [Oct 20 13:58:21] DEBUG[24088] devicestate.c: Changing state for SIP/3002 - state 6 (Ringing) [Oct 20 13:58:21] DEBUG[24088] devicestate.c: device 'SIP/3002' state '6' [Oct 20 13:58:21] DEBUG[24123] app_queue.c: Device 'SIP/3002' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Oct 20 13:58:21] VERBOSE[27665] chan_sip.c: Audio is at 5060 [Oct 20 13:58:21] VERBOSE[27665] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 20 13:58:21] VERBOSE[27665] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 20 13:58:21] VERBOSE[27665] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: -- Done with adding codecs to SDP [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Initializing initreq for method INVITE - callid 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 0 [ 37]: INVITE sip:3002@192.168.1.230 SIP/2.0 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 3 [ 51]: From: "3001" ;tag=as7a93a611 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 5 [ 37]: Contact: [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 6 [ 59]: Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:58:21 GMT [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 12 [ 83]: Remote-Party-ID: "3001" ;party=calling;privacy=off;screen=no [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Oct 20 13:58:21] VERBOSE[27665] chan_sip.c: Reliably Transmitting (NAT) to 195.59.152.66:40452: INVITE sip:3002@192.168.1.230 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport Max-Forwards: 70 From: "3001" ;tag=as7a93a611 To: Contact: Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "3001" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 289 v=0 o=root 1402128267 1402128267 IN IP4 212.62.4.230 s=Asterisk PBX 1.8.0-rc5 c=IN IP4 212.62.4.230 t=0 0 m=audio 13900 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81036 [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:21] VERBOSE[27665] app_dial.c: -- Called 3002 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81021 (1) NOTIFY - 4 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #81021)) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK14ef7a9b;rport Max-Forwards: 70 From: "asterisk" ;tag=as57468f5e To: Contact: Call-ID: 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81028 (1) NOTIFY - 4 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #81028)) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK28f11555;rport Max-Forwards: 70 From: "asterisk" ;tag=as1705a649 To: Contact: Call-ID: 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81033 (1) NOTIFY - 4 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #81033)) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK48bc47b6;rport Max-Forwards: 70 From: "asterisk" ;tag=as277261f1 To: Contact: Call-ID: 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport From: "3001" ;tag=as7a93a611 To: "3002" ;tag=441BD874-34B1942B CSeq: 102 INVITE Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Content-Length: 0 <-------------> [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 51]: From: "3001" ;tag=as7a93a611 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 57]: To: "3002" ;tag=441BD874-34B1942B [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 59]: Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: --- (10 headers 0 lines) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: = Looking for Call ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 (Checking To) --From tag as7a93a611 --To-tag 441BD874-34B1942B [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: *** SIP TIMER: Cancelling retransmission #81036 - INVITE (got response) [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7406065a6e117a15456106a53e278961@212.62.4.230:5060' Request 102: Found [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP response 100 to standard invite [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport From: "3001" ;tag=as7a93a611 To: "3002" ;tag=441BD874-34B1942B CSeq: 102 INVITE Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Allow-Events: talk,hold,conference Accept-Language: en Content-Length: 0 <-------------> [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 2 [ 51]: From: "3001" ;tag=as7a93a611 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 3 [ 57]: To: "3002" ;tag=441BD874-34B1942B [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 5 [ 59]: Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 8 [ 34]: Allow-Events: talk,hold,conference [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: --- (11 headers 0 lines) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: = Looking for Call ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 (Checking To) --From tag as7a93a611 --To-tag 441BD874-34B1942B [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7406065a6e117a15456106a53e278961@212.62.4.230:5060' Request 102: Found [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP response 180 to standard invite [Oct 20 13:58:21] VERBOSE[27665] app_dial.c: -- SIP/3002-000000ad is ringing [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Early remote bridge setting SIP '257253e4-96ab661-410d306a@192.168.1.231' - Sending media to (null) [Oct 20 13:58:21] DEBUG[27665] rtp_engine.c: Setting early bridge SDP of 'SIP/3001-000000ac' with that of 'SIP/3002-000000ad' [Oct 20 13:58:21] VERBOSE[27665] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320;received=195.59.152.66;rport=17261 From: "3001" ;tag=8707368-3FCA15D5 To: ;tag=as1a8324fd Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 20 13:58:21] DEBUG[27665] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:21] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3002 [Oct 20 13:58:21] DEBUG[24088] chan_sip.c: Checking device state for peer 3002 [Oct 20 13:58:21] DEBUG[24088] devicestate.c: Changing state for SIP/3002 - state 6 (Ringing) [Oct 20 13:58:21] DEBUG[24088] devicestate.c: device 'SIP/3002' state '6' [Oct 20 13:58:21] DEBUG[24123] app_queue.c: Device 'SIP/3002' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81021 (2) NOTIFY - 4 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #81021)) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK14ef7a9b;rport Max-Forwards: 70 From: "asterisk" ;tag=as57468f5e To: Contact: Call-ID: 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81028 (2) NOTIFY - 4 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #81028)) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK28f11555;rport Max-Forwards: 70 From: "asterisk" ;tag=as1705a649 To: Contact: Call-ID: 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81033 (2) NOTIFY - 4 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #81033)) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK48bc47b6;rport Max-Forwards: 70 From: "asterisk" ;tag=as277261f1 To: Contact: Call-ID: 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81021 (3) NOTIFY - 4 [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #81021)) [Oct 20 13:58:21] VERBOSE[24099] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK14ef7a9b;rport Max-Forwards: 70 From: "asterisk" ;tag=as57468f5e To: Contact: Call-ID: 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:21] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK9b65f87b36A01D1C From: "3001" ;tag=DD0DFFFD-20410596 To: CSeq: 1 SUBSCRIBE Call-ID: cec1beb9-93b4eb62-705a70e7@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK9b65f87b36A01D1C [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3001" ;tag=DD0DFFFD-20410596 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: cec1beb9-93b4eb62-705a70e7@192.168.1.231 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: = Looking for Call ID: cec1beb9-93b4eb62-705a70e7@192.168.1.231 (Checking From) --From tag DD0DFFFD-20410596 --To-tag [Oct 20 13:58:22] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:17261 is not local, substituting externaddr [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for cec1beb9-93b4eb62-705a70e7@192.168.1.231 - SUBSCRIBE (No RTP) [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid cec1beb9-93b4eb62-705a70e7@192.168.1.231 [Oct 20 13:58:22] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:22] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK9b65f87b36A01D1C;received=195.59.152.66;rport=17261 From: "3001" ;tag=DD0DFFFD-20410596 To: ;tag=as77b9ce48 Call-ID: cec1beb9-93b4eb62-705a70e7@192.168.1.231 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bace10a" Content-Length: 0 <------------> [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog 'cec1beb9-93b4eb62-705a70e7@192.168.1.231' in 6400 ms (Method: SUBSCRIBE) [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81028 (3) NOTIFY - 4 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #81028)) [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK28f11555;rport Max-Forwards: 70 From: "asterisk" ;tag=as1705a649 To: Contact: Call-ID: 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81033 (3) NOTIFY - 4 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #81033)) [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK48bc47b6;rport Max-Forwards: 70 From: "asterisk" ;tag=as277261f1 To: Contact: Call-ID: 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK51b6fdd8CD3B3E85 From: "3001" ;tag=DD0DFFFD-20410596 To: CSeq: 2 SUBSCRIBE Call-ID: cec1beb9-93b4eb62-705a70e7@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Authorization: Digest username="3001", realm="asterisk", nonce="5bace10a", uri="sip:3001@212.62.4.230:5060", response="7251df27595bf9cb08bc75194478fe57", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK51b6fdd8CD3B3E85 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3001" ;tag=DD0DFFFD-20410596 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 2 SUBSCRIBE [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: cec1beb9-93b4eb62-705a70e7@192.168.1.231 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 12 [167]: Authorization: Digest username="3001", realm="asterisk", nonce="5bace10a", uri="sip:3001@212.62.4.230:5060", response="7251df27595bf9cb08bc75194478fe57", algorithm=MD5 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: = Looking for Call ID: cec1beb9-93b4eb62-705a70e7@192.168.1.231 (Checking From) --From tag DD0DFFFD-20410596 --To-tag [Oct 20 13:58:22] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:22] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:22] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:22] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Got a new subscription cec1beb9-93b4eb62-705a70e7@192.168.1.231 (possibly with auth) [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid cec1beb9-93b4eb62-705a70e7@192.168.1.231 [Oct 20 13:58:22] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:22] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: build_route: Retaining previous route: [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Looking for 3001 in DLPN_All (domain 212.62.4.230:5060) [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK51b6fdd8CD3B3E85;received=195.59.152.66;rport=17261 From: "3001" ;tag=DD0DFFFD-20410596 To: ;tag=as77b9ce48 Call-ID: cec1beb9-93b4eb62-705a70e7@192.168.1.231 CSeq: 2 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Destroying SIP dialog cec1beb9-93b4eb62-705a70e7@192.168.1.231 [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog 'cec1beb9-93b4eb62-705a70e7@192.168.1.231' Method: SUBSCRIBE [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81030:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK2c7ebee6;rport Max-Forwards: 70 From: "asterisk" ;tag=as0a8443a8 To: Contact: Call-ID: 40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81021 (4) NOTIFY - 4 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #81021)) [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK14ef7a9b;rport Max-Forwards: 70 From: "asterisk" ;tag=as57468f5e To: Contact: Call-ID: 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81028 (4) NOTIFY - 4 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #81028)) [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK28f11555;rport Max-Forwards: 70 From: "asterisk" ;tag=as1705a649 To: Contact: Call-ID: 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81033 (4) NOTIFY - 4 [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #81033)) [Oct 20 13:58:22] VERBOSE[24099] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK48bc47b6;rport Max-Forwards: 70 From: "asterisk" ;tag=as277261f1 To: Contact: Call-ID: 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:22] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:23] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81030:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:23] VERBOSE[24099] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK2c7ebee6;rport Max-Forwards: 70 From: "asterisk" ;tag=as0a8443a8 To: Contact: Call-ID: 40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:23] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81030:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:24] VERBOSE[24099] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK2c7ebee6;rport Max-Forwards: 70 From: "asterisk" ;tag=as0a8443a8 To: Contact: Call-ID: 40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81021 (5) NOTIFY - 4 [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #81021)) [Oct 20 13:58:24] VERBOSE[24099] chan_sip.c: Retransmitting #5 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK14ef7a9b;rport Max-Forwards: 70 From: "asterisk" ;tag=as57468f5e To: Contact: Call-ID: 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81028 (5) NOTIFY - 4 [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #81028)) [Oct 20 13:58:24] VERBOSE[24099] chan_sip.c: Retransmitting #5 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK28f11555;rport Max-Forwards: 70 From: "asterisk" ;tag=as1705a649 To: Contact: Call-ID: 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81033 (5) NOTIFY - 4 [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #81033)) [Oct 20 13:58:24] VERBOSE[24099] chan_sip.c: Retransmitting #5 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK48bc47b6;rport Max-Forwards: 70 From: "asterisk" ;tag=as277261f1 To: Contact: Call-ID: 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:24] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKd789d3daC43F5941 From: "3002" ;tag=B16D3D9F-AE9DDBEE To: CSeq: 1 SUBSCRIBE Call-ID: b678420-1d27c237-b05a7f46@192.168.1.230 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKd789d3daC43F5941 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3002" ;tag=B16D3D9F-AE9DDBEE [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: b678420-1d27c237-b05a7f46@192.168.1.230 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: = Looking for Call ID: b678420-1d27c237-b05a7f46@192.168.1.230 (Checking From) --From tag B16D3D9F-AE9DDBEE --To-tag [Oct 20 13:58:25] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:40452 is not local, substituting externaddr [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for b678420-1d27c237-b05a7f46@192.168.1.230 - SUBSCRIBE (No RTP) [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid b678420-1d27c237-b05a7f46@192.168.1.230 [Oct 20 13:58:25] DEBUG[24099] netsock2.c: Splitting '192.168.1.230' gives... [Oct 20 13:58:25] DEBUG[24099] netsock2.c: ...host '192.168.1.230' and port '(null)'. [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:40452 (NAT) [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Found peer '3002' for '3002' from 195.59.152.66:40452 [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:40452 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKd789d3daC43F5941;received=195.59.152.66;rport=40452 From: "3002" ;tag=B16D3D9F-AE9DDBEE To: ;tag=as13b2ac8d Call-ID: b678420-1d27c237-b05a7f46@192.168.1.230 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="695683ed" Content-Length: 0 <------------> [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog 'b678420-1d27c237-b05a7f46@192.168.1.230' in 6400 ms (Method: SUBSCRIBE) [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKc4df950d59943B4C From: "3002" ;tag=B16D3D9F-AE9DDBEE To: CSeq: 2 SUBSCRIBE Call-ID: b678420-1d27c237-b05a7f46@192.168.1.230 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Authorization: Digest username="3002", realm="asterisk", nonce="695683ed", uri="sip:3002@212.62.4.230:5060", response="797fdbc2986012ae971dffac56617b5d", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKc4df950d59943B4C [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3002" ;tag=B16D3D9F-AE9DDBEE [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 2 SUBSCRIBE [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: b678420-1d27c237-b05a7f46@192.168.1.230 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 12 [167]: Authorization: Digest username="3002", realm="asterisk", nonce="695683ed", uri="sip:3002@212.62.4.230:5060", response="797fdbc2986012ae971dffac56617b5d", algorithm=MD5 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: = Looking for Call ID: b678420-1d27c237-b05a7f46@192.168.1.230 (Checking From) --From tag B16D3D9F-AE9DDBEE --To-tag [Oct 20 13:58:25] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:25] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:25] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:25] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Got a new subscription b678420-1d27c237-b05a7f46@192.168.1.230 (possibly with auth) [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid b678420-1d27c237-b05a7f46@192.168.1.230 [Oct 20 13:58:25] DEBUG[24099] netsock2.c: Splitting '192.168.1.230' gives... [Oct 20 13:58:25] DEBUG[24099] netsock2.c: ...host '192.168.1.230' and port '(null)'. [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:40452 (NAT) [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: build_route: Retaining previous route: [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Found peer '3002' for '3002' from 195.59.152.66:40452 [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Looking for 3002 in DLPN_All (domain 212.62.4.230:5060) [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:40452 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKc4df950d59943B4C;received=195.59.152.66;rport=40452 From: "3002" ;tag=B16D3D9F-AE9DDBEE To: ;tag=as13b2ac8d Call-ID: b678420-1d27c237-b05a7f46@192.168.1.230 CSeq: 2 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Destroying SIP dialog b678420-1d27c237-b05a7f46@192.168.1.230 [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog 'b678420-1d27c237-b05a7f46@192.168.1.230' Method: SUBSCRIBE [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81030:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK2c7ebee6;rport Max-Forwards: 70 From: "asterisk" ;tag=as0a8443a8 To: Contact: Call-ID: 40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:25] NOTICE[24099] chan_sip.c: Peer '3001' is now UNREACHABLE! Last qualify: 35 [Oct 20 13:58:25] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:25] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Update SQL: UPDATE gag1_sip SET lastms = '-1' WHERE name = '3001' [Oct 20 13:58:25] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Updated 1 rows on table: gag1_sip [Oct 20 13:58:25] DEBUG[24099] chan_sip.c: Destroying SIP dialog 40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060 [Oct 20 13:58:25] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '40b18b2a4ba186546d27dbb45ccc13cc@212.62.4.230:5060' Method: OPTIONS [Oct 20 13:58:25] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3001 [Oct 20 13:58:25] DEBUG[24088] chan_sip.c: Checking device state for peer 3001 [Oct 20 13:58:25] DEBUG[24088] devicestate.c: Changing state for SIP/3001 - state 2 (In use) [Oct 20 13:58:25] DEBUG[24088] devicestate.c: device 'SIP/3001' state '2' [Oct 20 13:58:25] DEBUG[24123] app_queue.c: Device 'SIP/3001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81021 (6) NOTIFY - 4 [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #81021)) [Oct 20 13:58:27] VERBOSE[24099] chan_sip.c: Retransmitting #6 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK14ef7a9b;rport Max-Forwards: 70 From: "asterisk" ;tag=as57468f5e To: Contact: Call-ID: 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81028 (6) NOTIFY - 4 [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #81028)) [Oct 20 13:58:27] VERBOSE[24099] chan_sip.c: Retransmitting #6 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK28f11555;rport Max-Forwards: 70 From: "asterisk" ;tag=as1705a649 To: Contact: Call-ID: 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: SIP TIMER: Rescheduling retransmission #81033 (6) NOTIFY - 4 [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #81033)) [Oct 20 13:58:27] VERBOSE[24099] chan_sip.c: Retransmitting #6 (NAT) to 192.168.1.231:17261: NOTIFY sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK48bc47b6;rport Max-Forwards: 70 From: "asterisk" ;tag=as277261f1 To: Contact: Call-ID: 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: Re-scheduled destruction of SIP call 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 [Oct 20 13:58:27] WARNING[24099] chan_sip.c: Retransmission timeout reached on transmission 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog '67d20df46480b243457f3b9f0ba1eb61@sip01.lon01.gagenetworks.net' [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: Destroying SIP dialog 67d20df46480b243457f3b9f0ba1eb61@sip01.lon01.gagenetworks.net [Oct 20 13:58:27] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '67d20df46480b243457f3b9f0ba1eb61@sip01.lon01.gagenetworks.net' Method: OPTIONS [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: Re-scheduled destruction of SIP call 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 [Oct 20 13:58:27] WARNING[24099] chan_sip.c: Retransmission timeout reached on transmission 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [Oct 20 13:58:27] DEBUG[24099] chan_sip.c: Re-scheduled destruction of SIP call 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 [Oct 20 13:58:27] WARNING[24099] chan_sip.c: Retransmission timeout reached on transmission 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [Oct 20 13:58:28] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:14529 ---> SUBSCRIBE sip:6003@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKaf7aacd12B5CFCE0 From: "6003" ;tag=AA8F8ACB-AD6D040A To: CSeq: 1 SUBSCRIBE Call-ID: 5a970df7-f9f28bd6-8c40243d@192.168.1.150 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.3.1734 Accept-Language: en Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6003@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKaf7aacd12B5CFCE0 [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "6003" ;tag=AA8F8ACB-AD6D040A [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 3 [ 27]: To: [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: 5a970df7-f9f28bd6-8c40243d@192.168.1.150 [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 6 [ 38]: Contact: [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 8 [ 22]: Event: message-summary [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.3.1734 [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:28] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: = Looking for Call ID: 5a970df7-f9f28bd6-8c40243d@192.168.1.150 (Checking From) --From tag AA8F8ACB-AD6D040A --To-tag [Oct 20 13:58:28] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:14529 is not local, substituting externaddr [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 5a970df7-f9f28bd6-8c40243d@192.168.1.150 - SUBSCRIBE (No RTP) [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:28] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 5a970df7-f9f28bd6-8c40243d@192.168.1.150 [Oct 20 13:58:28] DEBUG[24099] netsock2.c: Splitting '192.168.1.150:5060' gives... [Oct 20 13:58:28] DEBUG[24099] netsock2.c: ...host '192.168.1.150' and port '5060'. [Oct 20 13:58:28] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:14529 (NAT) [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:28] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:28] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:28] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '6003' AND host = 'dynamic' [Oct 20 13:58:28] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:28] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '6003' [Oct 20 13:58:28] VERBOSE[24099] chan_sip.c: No matching peer for '6003' from '195.59.152.66:14529' [Oct 20 13:58:28] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:14529 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKaf7aacd12B5CFCE0;received=195.59.152.66;rport=14529 From: "6003" ;tag=AA8F8ACB-AD6D040A To: ;tag=as47db6d84 Call-ID: 5a970df7-f9f28bd6-8c40243d@192.168.1.150 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:14529 [Oct 20 13:58:28] NOTICE[24099] chan_sip.c: Received SIP subscribe for peer without mailbox: (null) [Oct 20 13:58:28] DEBUG[24099] chan_sip.c: Destroying SIP dialog 5a970df7-f9f28bd6-8c40243d@192.168.1.150 [Oct 20 13:58:28] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '5a970df7-f9f28bd6-8c40243d@192.168.1.150' Method: SUBSCRIBE [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKf596595aAC0C2AFF From: "3001" ;tag=44A7C154-24BDB611 To: CSeq: 1 SUBSCRIBE Call-ID: a91dc390-1d19115d-cd5b2176@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKf596595aAC0C2AFF [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3001" ;tag=44A7C154-24BDB611 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: a91dc390-1d19115d-cd5b2176@192.168.1.231 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: = Looking for Call ID: a91dc390-1d19115d-cd5b2176@192.168.1.231 (Checking From) --From tag 44A7C154-24BDB611 --To-tag [Oct 20 13:58:29] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:17261 is not local, substituting externaddr [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for a91dc390-1d19115d-cd5b2176@192.168.1.231 - SUBSCRIBE (No RTP) [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid a91dc390-1d19115d-cd5b2176@192.168.1.231 [Oct 20 13:58:29] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:29] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKf596595aAC0C2AFF;received=195.59.152.66;rport=17261 From: "3001" ;tag=44A7C154-24BDB611 To: ;tag=as120c5380 Call-ID: a91dc390-1d19115d-cd5b2176@192.168.1.231 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56b5363f" Content-Length: 0 <------------> [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog 'a91dc390-1d19115d-cd5b2176@192.168.1.231' in 6400 ms (Method: SUBSCRIBE) [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKe48c44abAFB9208C From: "3001" ;tag=44A7C154-24BDB611 To: CSeq: 2 SUBSCRIBE Call-ID: a91dc390-1d19115d-cd5b2176@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Authorization: Digest username="3001", realm="asterisk", nonce="56b5363f", uri="sip:3001@212.62.4.230:5060", response="9248eb6c3c3bad5c07e9aa1c65a5332f", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKe48c44abAFB9208C [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3001" ;tag=44A7C154-24BDB611 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 2 SUBSCRIBE [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: a91dc390-1d19115d-cd5b2176@192.168.1.231 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 12 [167]: Authorization: Digest username="3001", realm="asterisk", nonce="56b5363f", uri="sip:3001@212.62.4.230:5060", response="9248eb6c3c3bad5c07e9aa1c65a5332f", algorithm=MD5 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: = Looking for Call ID: a91dc390-1d19115d-cd5b2176@192.168.1.231 (Checking From) --From tag 44A7C154-24BDB611 --To-tag [Oct 20 13:58:29] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:29] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:29] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:29] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Got a new subscription a91dc390-1d19115d-cd5b2176@192.168.1.231 (possibly with auth) [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid a91dc390-1d19115d-cd5b2176@192.168.1.231 [Oct 20 13:58:29] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:29] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: build_route: Retaining previous route: [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Looking for 3001 in DLPN_All (domain 212.62.4.230:5060) [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKe48c44abAFB9208C;received=195.59.152.66;rport=17261 From: "3001" ;tag=44A7C154-24BDB611 To: ;tag=as120c5380 Call-ID: a91dc390-1d19115d-cd5b2176@192.168.1.231 CSeq: 2 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:29] DEBUG[24099] chan_sip.c: Destroying SIP dialog a91dc390-1d19115d-cd5b2176@192.168.1.231 [Oct 20 13:58:29] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog 'a91dc390-1d19115d-cd5b2176@192.168.1.231' Method: SUBSCRIBE [Oct 20 13:58:34] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog '33da456233a3bc88721a129540f85ce2@212.62.4.230:5060' [Oct 20 13:58:34] DEBUG[24099] chan_sip.c: Destroying SIP dialog 33da456233a3bc88721a129540f85ce2@212.62.4.230:5060 [Oct 20 13:58:34] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '33da456233a3bc88721a129540f85ce2@212.62.4.230:5060' Method: NOTIFY [Oct 20 13:58:34] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog '18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060' [Oct 20 13:58:34] DEBUG[24099] chan_sip.c: Destroying SIP dialog 18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060 [Oct 20 13:58:34] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '18ebdf596a815bb76b99e19f6fd040ed@212.62.4.230:5060' Method: NOTIFY [Oct 20 13:58:34] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog '0d4e4944211af9116758fc4263785e40@212.62.4.230:5060' [Oct 20 13:58:34] DEBUG[24099] chan_sip.c: Destroying SIP dialog 0d4e4944211af9116758fc4263785e40@212.62.4.230:5060 [Oct 20 13:58:34] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '0d4e4944211af9116758fc4263785e40@212.62.4.230:5060' Method: NOTIFY [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:14529 ---> SUBSCRIBE sip:6003@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK232e197865AC21CF From: "6003" ;tag=8A6CA422-2510E829 To: CSeq: 1 SUBSCRIBE Call-ID: bedafcee-cde72895-1829484@192.168.1.150 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.3.1734 Accept-Language: en Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6003@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK232e197865AC21CF [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "6003" ;tag=8A6CA422-2510E829 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 3 [ 27]: To: [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: bedafcee-cde72895-1829484@192.168.1.150 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 6 [ 38]: Contact: [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 8 [ 22]: Event: message-summary [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.3.1734 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: = Looking for Call ID: bedafcee-cde72895-1829484@192.168.1.150 (Checking From) --From tag 8A6CA422-2510E829 --To-tag [Oct 20 13:58:35] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:14529 is not local, substituting externaddr [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for bedafcee-cde72895-1829484@192.168.1.150 - SUBSCRIBE (No RTP) [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid bedafcee-cde72895-1829484@192.168.1.150 [Oct 20 13:58:35] DEBUG[24099] netsock2.c: Splitting '192.168.1.150:5060' gives... [Oct 20 13:58:35] DEBUG[24099] netsock2.c: ...host '192.168.1.150' and port '5060'. [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:14529 (NAT) [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:35] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:35] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '6003' AND host = 'dynamic' [Oct 20 13:58:35] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:35] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '6003' [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: No matching peer for '6003' from '195.59.152.66:14529' [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:14529 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK232e197865AC21CF;received=195.59.152.66;rport=14529 From: "6003" ;tag=8A6CA422-2510E829 To: ;tag=as4f40c4de Call-ID: bedafcee-cde72895-1829484@192.168.1.150 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:14529 [Oct 20 13:58:35] NOTICE[24099] chan_sip.c: Received SIP subscribe for peer without mailbox: (null) [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Destroying SIP dialog bedafcee-cde72895-1829484@192.168.1.150 [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog 'bedafcee-cde72895-1829484@192.168.1.150' Method: SUBSCRIBE [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 1955165419788031343c557351fcc97a@127.0.0.1:0 - OPTIONS (No RTP) [Oct 20 13:58:35] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Initializing initreq for method OPTIONS - callid 71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 0 [ 38]: OPTIONS sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK44ccccb6;rport [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as24031695 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:58:35 GMT [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 20 13:58:35] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK44ccccb6;rport Max-Forwards: 70 From: "asterisk" ;tag=as24031695 To: Contact: Call-ID: 71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81043 [Oct 20 13:58:35] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:36] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81043:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:36] VERBOSE[24099] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK44ccccb6;rport Max-Forwards: 70 From: "asterisk" ;tag=as24031695 To: Contact: Call-ID: 71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:36] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:36] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog 'dac0db36-f1a6376b-2ef9e84c@192.168.1.231' [Oct 20 13:58:36] DEBUG[24099] chan_sip.c: Destroying SIP dialog dac0db36-f1a6376b-2ef9e84c@192.168.1.231 [Oct 20 13:58:36] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog 'dac0db36-f1a6376b-2ef9e84c@192.168.1.231' Method: REGISTER [Oct 20 13:58:37] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:14529 ---> REGISTER sip:212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK42b087a7840D3006 From: "6003" ;tag=2E1DF297-DEA45B76 To: CSeq: 1 REGISTER Call-ID: 6536deed-5240d41c-13bab153@192.168.1.150 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 0 [ 38]: REGISTER sip:212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK42b087a7840D3006 [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "6003" ;tag=2E1DF297-DEA45B76 [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 3 [ 27]: To: [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 4 [ 16]: CSeq: 1 REGISTER [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: 6536deed-5240d41c-13bab153@192.168.1.150 [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 6 [138]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.3.1734 [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 10 [ 13]: Expires: 3600 [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Oct 20 13:58:37] VERBOSE[24099] chan_sip.c: --- (12 headers 0 lines) --- [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: = Looking for Call ID: 6536deed-5240d41c-13bab153@192.168.1.150 (Checking From) --From tag 2E1DF297-DEA45B76 --To-tag [Oct 20 13:58:37] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:14529 is not local, substituting externaddr [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 6536deed-5240d41c-13bab153@192.168.1.150 - REGISTER (No RTP) [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Initializing initreq for method REGISTER - callid 6536deed-5240d41c-13bab153@192.168.1.150 [Oct 20 13:58:37] DEBUG[24099] netsock2.c: Splitting '192.168.1.150:5060' gives... [Oct 20 13:58:37] DEBUG[24099] netsock2.c: ...host '192.168.1.150' and port '5060'. [Oct 20 13:58:37] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:14529 (NAT) [Oct 20 13:58:37] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:37] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '6003' AND host = 'dynamic' [Oct 20 13:58:37] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:37] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '6003' [Oct 20 13:58:37] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:14529 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK42b087a7840D3006;received=195.59.152.66;rport=14529 From: "6003" ;tag=2E1DF297-DEA45B76 To: ;tag=as4e570fe5 Call-ID: 6536deed-5240d41c-13bab153@192.168.1.150 CSeq: 1 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:14529 [Oct 20 13:58:37] NOTICE[24099] chan_sip.c: Registration from '' failed for '195.59.152.66:14529' - No matching peer found [Oct 20 13:58:37] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '6536deed-5240d41c-13bab153@192.168.1.150' in 32000 ms (Method: REGISTER) [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81043:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:37] VERBOSE[24099] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK44ccccb6;rport Max-Forwards: 70 From: "asterisk" ;tag=as24031695 To: Contact: Call-ID: 71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:37] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:38] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81043:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:38] VERBOSE[24099] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK44ccccb6;rport Max-Forwards: 70 From: "asterisk" ;tag=as24031695 To: Contact: Call-ID: 71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:38] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKa5d264cfD9E8B09E From: "3002" ;tag=9CC4F059-4E12DF8 To: CSeq: 1 SUBSCRIBE Call-ID: b6fb1b25-4a0b8c24-664e395b@192.168.1.230 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKa5d264cfD9E8B09E [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3002" ;tag=9CC4F059-4E12DF8 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: b6fb1b25-4a0b8c24-664e395b@192.168.1.230 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: = Looking for Call ID: b6fb1b25-4a0b8c24-664e395b@192.168.1.230 (Checking From) --From tag 9CC4F059-4E12DF8 --To-tag [Oct 20 13:58:39] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:40452 is not local, substituting externaddr [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for b6fb1b25-4a0b8c24-664e395b@192.168.1.230 - SUBSCRIBE (No RTP) [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid b6fb1b25-4a0b8c24-664e395b@192.168.1.230 [Oct 20 13:58:39] DEBUG[24099] netsock2.c: Splitting '192.168.1.230' gives... [Oct 20 13:58:39] DEBUG[24099] netsock2.c: ...host '192.168.1.230' and port '(null)'. [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:40452 (NAT) [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Found peer '3002' for '3002' from 195.59.152.66:40452 [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:40452 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKa5d264cfD9E8B09E;received=195.59.152.66;rport=40452 From: "3002" ;tag=9CC4F059-4E12DF8 To: ;tag=as1698411c Call-ID: b6fb1b25-4a0b8c24-664e395b@192.168.1.230 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="28ca6e79" Content-Length: 0 <------------> [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog 'b6fb1b25-4a0b8c24-664e395b@192.168.1.230' in 6400 ms (Method: SUBSCRIBE) [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKf3d0968aCCDBC571 From: "3002" ;tag=9CC4F059-4E12DF8 To: CSeq: 2 SUBSCRIBE Call-ID: b6fb1b25-4a0b8c24-664e395b@192.168.1.230 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Authorization: Digest username="3002", realm="asterisk", nonce="28ca6e79", uri="sip:3002@212.62.4.230:5060", response="0a55ae1c96f71f5b0db99e754c97d39d", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKf3d0968aCCDBC571 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3002" ;tag=9CC4F059-4E12DF8 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 2 SUBSCRIBE [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: b6fb1b25-4a0b8c24-664e395b@192.168.1.230 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 12 [167]: Authorization: Digest username="3002", realm="asterisk", nonce="28ca6e79", uri="sip:3002@212.62.4.230:5060", response="0a55ae1c96f71f5b0db99e754c97d39d", algorithm=MD5 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: = Looking for Call ID: b6fb1b25-4a0b8c24-664e395b@192.168.1.230 (Checking From) --From tag 9CC4F059-4E12DF8 --To-tag [Oct 20 13:58:39] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:39] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:39] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:39] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Got a new subscription b6fb1b25-4a0b8c24-664e395b@192.168.1.230 (possibly with auth) [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid b6fb1b25-4a0b8c24-664e395b@192.168.1.230 [Oct 20 13:58:39] DEBUG[24099] netsock2.c: Splitting '192.168.1.230' gives... [Oct 20 13:58:39] DEBUG[24099] netsock2.c: ...host '192.168.1.230' and port '(null)'. [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:40452 (NAT) [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: build_route: Retaining previous route: [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Found peer '3002' for '3002' from 195.59.152.66:40452 [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Looking for 3002 in DLPN_All (domain 212.62.4.230:5060) [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:40452 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKf3d0968aCCDBC571;received=195.59.152.66;rport=40452 From: "3002" ;tag=9CC4F059-4E12DF8 To: ;tag=as1698411c Call-ID: b6fb1b25-4a0b8c24-664e395b@192.168.1.230 CSeq: 2 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Destroying SIP dialog b6fb1b25-4a0b8c24-664e395b@192.168.1.230 [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog 'b6fb1b25-4a0b8c24-664e395b@192.168.1.230' Method: SUBSCRIBE [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81043:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK44ccccb6;rport Max-Forwards: 70 From: "asterisk" ;tag=as24031695 To: Contact: Call-ID: 71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:39] DEBUG[24099] chan_sip.c: Destroying SIP dialog 71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060 [Oct 20 13:58:39] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '71836b5b1bd1d13e7054316b1f7a6a2c@212.62.4.230:5060' Method: OPTIONS [Oct 20 13:58:40] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog '12fd97e506f21869282a809b6a6a48d8@212.62.8.164' [Oct 20 13:58:40] DEBUG[24099] chan_sip.c: Destroying SIP dialog 12fd97e506f21869282a809b6a6a48d8@212.62.8.164 [Oct 20 13:58:40] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '12fd97e506f21869282a809b6a6a48d8@212.62.8.164' Method: OPTIONS [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Auto destroying SIP dialog 'c1de59ec-3716363-d42eed2@192.168.1.230' [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Destroying SIP dialog c1de59ec-3716363-d42eed2@192.168.1.230 [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog 'c1de59ec-3716363-d42eed2@192.168.1.230' Method: REGISTER [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> CANCEL sip:3002@212.62.4.230:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320 From: "3001" ;tag=8707368-3FCA15D5 To: CSeq: 2 CANCEL Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Authorization: Digest username="3001", realm="asterisk", nonce="61d2824b", uri="sip:3002@212.62.4.230:5060;user=phone", response="f5a6b920ba3d688edf80d316b7ec4e70", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 0 [ 52]: CANCEL sip:3002@212.62.4.230:5060;user=phone SIP/2.0 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3001" ;tag=8707368-3FCA15D5 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 3 [ 38]: To: [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 4 [ 14]: CSeq: 2 CANCEL [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 8 [178]: Authorization: Digest username="3001", realm="asterisk", nonce="61d2824b", uri="sip:3002@212.62.4.230:5060;user=phone", response="f5a6b920ba3d688edf80d316b7ec4e70", algorithm=MD5 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: --- (11 headers 0 lines) --- [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: = Looking for Call ID: 257253e4-96ab661-410d306a@192.168.1.231 (Checking From) --From tag 8707368-3FCA15D5 --To-tag [Oct 20 13:58:41] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:41] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:41] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:41] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL [Oct 20 13:58:41] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:41] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Setting SIP_ALREADYGONE on dialog 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Updating call counter for incoming call [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Call from peer '3001' removed from call limit 2147483647 [Oct 20 13:58:41] DEBUG[24099] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x80859b8' [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: <--- Reliably Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320;received=195.59.152.66;rport=17261 From: "3001" ;tag=8707368-3FCA15D5 To: ;tag=as1a8324fd Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81048 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320;received=195.59.152.66;rport=17261 From: "3001" ;tag=8707368-3FCA15D5 To: ;tag=as1a8324fd Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 CSeq: 2 CANCEL Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:41] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3001 [Oct 20 13:58:41] DEBUG[24088] chan_sip.c: Checking device state for peer 3001 [Oct 20 13:58:41] DEBUG[24088] devicestate.c: Changing state for SIP/3001 - state 5 (Unavailable) [Oct 20 13:58:41] DEBUG[24088] devicestate.c: device 'SIP/3001' state '5' [Oct 20 13:58:41] DEBUG[27665] channel.c: Hanging up channel 'SIP/3002-000000ad' [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: Hangup call SIP/3002-000000ad, SIP callid 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: update_call_counter(3002) - decrement call limit counter on hangup [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: Updating call counter for outgoing call [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: Call to peer '3002' removed from call limit 2147483647 [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: Hanging up channel in state Ringing (not UP) [Oct 20 13:58:41] DEBUG[27665] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7e97338' [Oct 20 13:58:41] VERBOSE[27665] chan_sip.c: Scheduling destruction of SIP dialog '7406065a6e117a15456106a53e278961@212.62.4.230:5060' in 6400 ms (Method: INVITE) [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7406065a6e117a15456106a53e278961@212.62.4.230:5060' Request 102: Found [Oct 20 13:58:41] VERBOSE[27665] chan_sip.c: Reliably Transmitting (NAT) to 195.59.152.66:40452: CANCEL sip:3002@192.168.1.230 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport Max-Forwards: 70 From: "3001" ;tag=as7a93a611 To: Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81050 [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:41] VERBOSE[27665] chan_sip.c: Scheduling destruction of SIP dialog '7406065a6e117a15456106a53e278961@212.62.4.230:5060' in 6400 ms (Method: INVITE) [Oct 20 13:58:41] DEBUG[27665] app_dial.c: Exiting with DIALSTATUS=CANCEL. [Oct 20 13:58:41] DEBUG[27665] pbx.c: Spawn extension (internal,3002,1) exited non-zero on 'SIP/3001-000000ac' [Oct 20 13:58:41] VERBOSE[27665] pbx.c: == Spawn extension (internal, 3002, 1) exited non-zero on 'SIP/3001-000000ac' [Oct 20 13:58:41] DEBUG[27665] channel.c: Soft-Hanging up channel 'SIP/3001-000000ac' [Oct 20 13:58:41] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3002 [Oct 20 13:58:41] DEBUG[24088] chan_sip.c: Checking device state for peer 3002 [Oct 20 13:58:41] DEBUG[24088] devicestate.c: Changing state for SIP/3002 - state 1 (Not in use) [Oct 20 13:58:41] DEBUG[24088] devicestate.c: device 'SIP/3002' state '1' [Oct 20 13:58:41] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3002 [Oct 20 13:58:41] DEBUG[24088] chan_sip.c: Checking device state for peer 3002 [Oct 20 13:58:41] DEBUG[24088] devicestate.c: Changing state for SIP/3002 - state 1 (Not in use) [Oct 20 13:58:41] DEBUG[24088] devicestate.c: device 'SIP/3002' state '1' [Oct 20 13:58:41] DEBUG[24123] app_queue.c: Device 'SIP/3001' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Oct 20 13:58:41] DEBUG[24123] app_queue.c: Device 'SIP/3002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 20 13:58:41] DEBUG[24123] app_queue.c: Device 'SIP/3002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 20 13:58:41] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:41] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = 'h' AND context = 'internal' AND priority = '1' [Oct 20 13:58:41] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:41] DEBUG[27665] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'internal' AND priority = '1' ORDER BY exten [Oct 20 13:58:41] DEBUG[27665] channel.c: Hanging up channel 'SIP/3001-000000ac' [Oct 20 13:58:41] DEBUG[27665] chan_sip.c: Hangup call SIP/3001-000000ac, SIP callid 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:41] DEBUG[27665] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x80859b8' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is '2010-10-20 13:58:21' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is '"3001" <3001>' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is 'internal' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is 'SIP/3001-000000ac' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is 'SIP/3002-000000ad' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is 'Dial' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is 'SIP/3002' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is '20' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is '0' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is 'NO ANSWER' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is 'DOCUMENTATION' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is '(null)' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is 'PR-GN-NIMBUSTEST-V-01-1287579501.172' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is '(null)' [Oct 20 13:58:41] DEBUG[27665] pbx.c: Function result is '(null)' [Oct 20 13:58:41] DEBUG[27665] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2010-10-20 13:58:21','"3001" <3001>','internal','SIP/3001-000000ac','SIP/3002-000000ad','Dial','SIP/3002','20','0','NO ANSWER','DOCUMENTATION','','PR-GN-NIMBUSTEST-V-01-1287579501.172','','') [Oct 20 13:58:41] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3001 [Oct 20 13:58:41] DEBUG[24088] chan_sip.c: Checking device state for peer 3001 [Oct 20 13:58:41] DEBUG[24088] devicestate.c: Changing state for SIP/3001 - state 5 (Unavailable) [Oct 20 13:58:41] DEBUG[24088] devicestate.c: device 'SIP/3001' state '5' [Oct 20 13:58:41] DEBUG[24123] app_queue.c: Device 'SIP/3001' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport From: "3001" ;tag=as7a93a611 To: "3002" CSeq: 102 CANCEL Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Content-Length: 0 <-------------> [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 2 [ 51]: From: "3001" ;tag=as7a93a611 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 3 [ 35]: To: "3002" [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 4 [ 16]: CSeq: 102 CANCEL [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 5 [ 59]: Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: --- (10 headers 0 lines) --- [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: = Looking for Call ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 (Checking To) --From tag as7a93a611 --To-tag [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Acked pending invite 102 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81050 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Stopping retransmission on '7406065a6e117a15456106a53e278961@212.62.4.230:5060' of Request 102: Match Found [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport From: "3001" ;tag=as7a93a611 To: "3002" ;tag=441BD874-34B1942B CSeq: 102 INVITE Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Content-Length: 0 <-------------> [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 0 [ 29]: SIP/2.0 487 Request Cancelled [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 2 [ 51]: From: "3001" ;tag=as7a93a611 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 3 [ 57]: To: "3002" ;tag=441BD874-34B1942B [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 5 [ 59]: Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: --- (10 headers 0 lines) --- [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: = Looking for Call ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 (Checking To) --From tag as7a93a611 --To-tag 441BD874-34B1942B [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Stopping retransmission on '7406065a6e117a15456106a53e278961@212.62.4.230:5060' of Request 102: Match Found [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: SIP response 487 to standard invite [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: Transmitting (NAT) to 195.59.152.66:40452: ACK sip:3002@192.168.1.230 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK1826b7af;rport Max-Forwards: 70 From: "3001" ;tag=as7a93a611 To: ;tag=441BD874-34B1942B Contact: Call-ID: 7406065a6e117a15456106a53e278961@212.62.4.230:5060 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Trying to put 'ACK sip:300' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Updating call counter for outgoing call [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Call to peer '3002' removed from call limit 2147483647 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Destroying SIP dialog 7406065a6e117a15456106a53e278961@212.62.4.230:5060 [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '7406065a6e117a15456106a53e278961@212.62.4.230:5060' Method: INVITE [Oct 20 13:58:41] DEBUG[24099] rtp_engine.c: Destroyed RTP instance '0x7e97338' [Oct 20 13:58:41] DEBUG[24088] devicestate.c: No provider found, checking channel drivers for SIP - 3002 [Oct 20 13:58:41] DEBUG[24088] chan_sip.c: Checking device state for peer 3002 [Oct 20 13:58:41] DEBUG[24088] devicestate.c: Changing state for SIP/3002 - state 1 (Not in use) [Oct 20 13:58:41] DEBUG[24088] devicestate.c: device 'SIP/3002' state '1' [Oct 20 13:58:41] DEBUG[24123] app_queue.c: Device 'SIP/3002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> ACK sip:3002@212.62.4.230:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320 From: "3001" ;tag=8707368-3FCA15D5 To: ;tag=as1a8324fd CSeq: 2 ACK Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 0 [ 49]: ACK sip:3002@212.62.4.230:5060;user=phone SIP/2.0 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK5f421dcfBE708320 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3001" ;tag=8707368-3FCA15D5 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 3 [ 53]: To: ;tag=as1a8324fd [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 4 [ 11]: CSeq: 2 ACK [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: --- (12 headers 0 lines) --- [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: = Looking for Call ID: 257253e4-96ab661-410d306a@192.168.1.231 (Checking From) --From tag 8707368-3FCA15D5 --To-tag as1a8324fd [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81048 [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Stopping retransmission on '257253e4-96ab661-410d306a@192.168.1.231' of Response 2: Match Found [Oct 20 13:58:41] DEBUG[24099] chan_sip.c: Destroying SIP dialog 257253e4-96ab661-410d306a@192.168.1.231 [Oct 20 13:58:41] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '257253e4-96ab661-410d306a@192.168.1.231' Method: ACK [Oct 20 13:58:41] DEBUG[24099] rtp_engine.c: Destroyed RTP instance '0x80859b8' [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SUBSCRIBE sip:asterisk@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bK65e34afc8ABE58F3 From: "3002" ;tag=4EE01759-ADDFCCF8 To: ;tag=as6a4a15b8 CSeq: 359 SUBSCRIBE Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/simple-message-summary Authorization: Digest username="3002", realm="asterisk", nonce="4cb7eec8", uri="sip:asterisk@212.62.4.230:5060", response="812fc386e463793242c0087f334e2ed6", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 0 [ 48]: SUBSCRIBE sip:asterisk@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bK65e34afc8ABE58F3 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3002" ;tag=4EE01759-ADDFCCF8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 3 [ 42]: To: ;tag=as6a4a15b8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 4 [ 19]: CSeq: 359 SUBSCRIBE [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 8 [ 22]: Event: message-summary [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 12 [171]: Authorization: Digest username="3002", realm="asterisk", nonce="4cb7eec8", uri="sip:asterisk@212.62.4.230:5060", response="812fc386e463793242c0087f334e2ed6", algorithm=MD5 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: = Looking for Call ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 (Checking From) --From tag 4EE01759-ADDFCCF8 --To-tag as6a4a15b8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Got a re-subscribe on existing subscription e23ea225-2fdd8b24-5d24705b@192.168.1.230 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: Found peer '3002' for '3002' from 195.59.152.66:40452 [Oct 20 13:58:42] NOTICE[24099] chan_sip.c: Correct auth, but based on stale nonce received from '"3002" ;tag=4EE01759-ADDFCCF8' [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:40452 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bK65e34afc8ABE58F3;received=195.59.152.66;rport=40452 From: "3002" ;tag=4EE01759-ADDFCCF8 To: ;tag=as6a4a15b8 Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 CSeq: 359 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0744b3dd", stale=true Content-Length: 0 <------------> [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog 'e23ea225-2fdd8b24-5d24705b@192.168.1.230' in 6400 ms (Method: SUBSCRIBE) [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SUBSCRIBE sip:asterisk@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKe4ccece2B5095889 From: "3002" ;tag=4EE01759-ADDFCCF8 To: ;tag=as6a4a15b8 CSeq: 360 SUBSCRIBE Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/simple-message-summary Authorization: Digest username="3002", realm="asterisk", nonce="0744b3dd", uri="sip:asterisk@212.62.4.230:5060", response="5394487d34917f90e114e80d09cb3cb3", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 0 [ 48]: SUBSCRIBE sip:asterisk@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKe4ccece2B5095889 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3002" ;tag=4EE01759-ADDFCCF8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 3 [ 42]: To: ;tag=as6a4a15b8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 4 [ 19]: CSeq: 360 SUBSCRIBE [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 8 [ 22]: Event: message-summary [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 12 [171]: Authorization: Digest username="3002", realm="asterisk", nonce="0744b3dd", uri="sip:asterisk@212.62.4.230:5060", response="5394487d34917f90e114e80d09cb3cb3", algorithm=MD5 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: = Looking for Call ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 (Checking From) --From tag 4EE01759-ADDFCCF8 --To-tag as6a4a15b8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Got a re-subscribe on existing subscription e23ea225-2fdd8b24-5d24705b@192.168.1.230 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: Found peer '3002' for '3002' from 195.59.152.66:40452 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Adding subscription for mailbox notification - peer 3002 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog 'e23ea225-2fdd8b24-5d24705b@192.168.1.230' in 130000 ms (Method: SUBSCRIBE) [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:40452 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKe4ccece2B5095889;received=195.59.152.66;rport=40452 From: "3002" ;tag=4EE01759-ADDFCCF8 To: ;tag=as6a4a15b8 Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 CSeq: 360 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Content-Length: 0 <------------> [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 195.59.152.66:40452: NOTIFY sip:3002@192.168.1.230 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK19cf3c56;rport Max-Forwards: 70 Route: From: "asterisk" ;tag=as6a4a15b8 To: ;tag=4EE01759-ADDFCCF8 Contact: Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 CSeq: 282 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 97 Messages-Waiting: no Message-Account: sip:asterisk@212.62.4.230:5060 Voice-Message: 0/0 (0/0) --- [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81054 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK19cf3c56;rport From: "asterisk" ;tag=as6a4a15b8 To: "3002" ;tag=4EE01759-ADDFCCF8 CSeq: 282 NOTIFY Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Content-Length: 0 <-------------> [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK19cf3c56;rport [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as6a4a15b8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 3 [ 57]: To: "3002" ;tag=4EE01759-ADDFCCF8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 4 [ 16]: CSeq: 282 NOTIFY [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 7 [ 22]: Event: message-summary [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 20 13:58:42] VERBOSE[24099] chan_sip.c: --- (11 headers 0 lines) --- [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: = Looking for Call ID: e23ea225-2fdd8b24-5d24705b@192.168.1.230 (Checking To) --From tag as6a4a15b8 --To-tag 4EE01759-ADDFCCF8 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81054 [Oct 20 13:58:42] DEBUG[24099] chan_sip.c: Stopping retransmission on 'e23ea225-2fdd8b24-5d24705b@192.168.1.230' of Request 282: Match Found [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK6cf62d43DD501AC4 From: "3001" ;tag=2AFAAC17-AD0AB448 To: CSeq: 1 SUBSCRIBE Call-ID: 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK6cf62d43DD501AC4 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3001" ;tag=2AFAAC17-AD0AB448 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: = Looking for Call ID: 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 (Checking From) --From tag 2AFAAC17-AD0AB448 --To-tag [Oct 20 13:58:43] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:17261 is not local, substituting externaddr [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 - SUBSCRIBE (No RTP) [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 [Oct 20 13:58:43] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:43] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK6cf62d43DD501AC4;received=195.59.152.66;rport=17261 From: "3001" ;tag=2AFAAC17-AD0AB448 To: ;tag=as228eeebd Call-ID: 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a7f0448" Content-Length: 0 <------------> [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '3ac1f1c1-bfedce4a-74c5542f@192.168.1.231' in 6400 ms (Method: SUBSCRIBE) [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKfaefb0009BF7140D From: "3001" ;tag=2AFAAC17-AD0AB448 To: CSeq: 2 SUBSCRIBE Call-ID: 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Authorization: Digest username="3001", realm="asterisk", nonce="0a7f0448", uri="sip:3001@212.62.4.230:5060", response="485aa8c009ef797ec009e402bc8f139e", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKfaefb0009BF7140D [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3001" ;tag=2AFAAC17-AD0AB448 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 2 SUBSCRIBE [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 12 [167]: Authorization: Digest username="3001", realm="asterisk", nonce="0a7f0448", uri="sip:3001@212.62.4.230:5060", response="485aa8c009ef797ec009e402bc8f139e", algorithm=MD5 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: = Looking for Call ID: 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 (Checking From) --From tag 2AFAAC17-AD0AB448 --To-tag [Oct 20 13:58:43] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:43] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:43] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:43] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Got a new subscription 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 (possibly with auth) [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 [Oct 20 13:58:43] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:43] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: build_route: Retaining previous route: [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Looking for 3001 in DLPN_All (domain 212.62.4.230:5060) [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bKfaefb0009BF7140D;received=195.59.152.66;rport=17261 From: "3001" ;tag=2AFAAC17-AD0AB448 To: ;tag=as228eeebd Call-ID: 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 CSeq: 2 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:43] DEBUG[24099] chan_sip.c: Destroying SIP dialog 3ac1f1c1-bfedce4a-74c5542f@192.168.1.231 [Oct 20 13:58:43] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '3ac1f1c1-bfedce4a-74c5542f@192.168.1.231' Method: SUBSCRIBE [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKb3410ba8134E03FF From: "3002" ;tag=ED71EFF6-36CE9F3D To: CSeq: 1 SUBSCRIBE Call-ID: 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKb3410ba8134E03FF [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3002" ;tag=ED71EFF6-36CE9F3D [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: = Looking for Call ID: 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 (Checking From) --From tag ED71EFF6-36CE9F3D --To-tag [Oct 20 13:58:46] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:40452 is not local, substituting externaddr [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 - SUBSCRIBE (No RTP) [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 [Oct 20 13:58:46] DEBUG[24099] netsock2.c: Splitting '192.168.1.230' gives... [Oct 20 13:58:46] DEBUG[24099] netsock2.c: ...host '192.168.1.230' and port '(null)'. [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:40452 (NAT) [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Found peer '3002' for '3002' from 195.59.152.66:40452 [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:40452 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bKb3410ba8134E03FF;received=195.59.152.66;rport=40452 From: "3002" ;tag=ED71EFF6-36CE9F3D To: ;tag=as0df4c4ad Call-ID: 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="786fc4a3" Content-Length: 0 <------------> [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230' in 6400 ms (Method: SUBSCRIBE) [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:40452 ---> SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bK248c568bE306913A From: "3002" ;tag=ED71EFF6-36CE9F3D To: CSeq: 2 SUBSCRIBE Call-ID: 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Authorization: Digest username="3002", realm="asterisk", nonce="786fc4a3", uri="sip:3002@212.62.4.230:5060", response="0f053087f97b4778f98a810ae9b7f4b5", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3002@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bK248c568bE306913A [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "3002" ;tag=ED71EFF6-36CE9F3D [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 2 SUBSCRIBE [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 12 [167]: Authorization: Digest username="3002", realm="asterisk", nonce="786fc4a3", uri="sip:3002@212.62.4.230:5060", response="0f053087f97b4778f98a810ae9b7f4b5", algorithm=MD5 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: = Looking for Call ID: 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 (Checking From) --From tag ED71EFF6-36CE9F3D --To-tag [Oct 20 13:58:46] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:46] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:46] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:46] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Got a new subscription 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 (possibly with auth) [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 [Oct 20 13:58:46] DEBUG[24099] netsock2.c: Splitting '192.168.1.230' gives... [Oct 20 13:58:46] DEBUG[24099] netsock2.c: ...host '192.168.1.230' and port '(null)'. [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:40452 (NAT) [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: build_route: Retaining previous route: [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Found peer '3002' for '3002' from 195.59.152.66:40452 [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Looking for 3002 in DLPN_All (domain 212.62.4.230:5060) [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:40452 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.230;branch=z9hG4bK248c568bE306913A;received=195.59.152.66;rport=40452 From: "3002" ;tag=ED71EFF6-36CE9F3D To: ;tag=as0df4c4ad Call-ID: 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 CSeq: 2 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:40452 [Oct 20 13:58:46] DEBUG[24099] chan_sip.c: Destroying SIP dialog 86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230 [Oct 20 13:58:46] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '86ffbd4e-b04fa155-7d5cf7d4@192.168.1.230' Method: SUBSCRIBE [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:14529 ---> SUBSCRIBE sip:6003@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK6deea51e9728C745 From: "6003" ;tag=801F1581-9E9C9810 To: CSeq: 1 SUBSCRIBE Call-ID: d14b5b4-d45ce22b-8673906a@192.168.1.150 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.3.1734 Accept-Language: en Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6003@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK6deea51e9728C745 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 2 [ 58]: From: "6003" ;tag=801F1581-9E9C9810 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 3 [ 27]: To: [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 5 [ 48]: Call-ID: d14b5b4-d45ce22b-8673906a@192.168.1.150 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 6 [ 38]: Contact: [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 8 [ 22]: Event: message-summary [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.3.1734 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: = Looking for Call ID: d14b5b4-d45ce22b-8673906a@192.168.1.150 (Checking From) --From tag 801F1581-9E9C9810 --To-tag [Oct 20 13:58:49] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:14529 is not local, substituting externaddr [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for d14b5b4-d45ce22b-8673906a@192.168.1.150 - SUBSCRIBE (No RTP) [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid d14b5b4-d45ce22b-8673906a@192.168.1.150 [Oct 20 13:58:49] DEBUG[24099] netsock2.c: Splitting '192.168.1.150:5060' gives... [Oct 20 13:58:49] DEBUG[24099] netsock2.c: ...host '192.168.1.150' and port '5060'. [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:14529 (NAT) [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '6003' AND host = 'dynamic' [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_sip WHERE name = '6003' [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: No matching peer for '6003' from '195.59.152.66:14529' [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:14529 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK6deea51e9728C745;received=195.59.152.66;rport=14529 From: "6003" ;tag=801F1581-9E9C9810 To: ;tag=as50576402 Call-ID: d14b5b4-d45ce22b-8673906a@192.168.1.150 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:14529 [Oct 20 13:58:49] NOTICE[24099] chan_sip.c: Received SIP subscribe for peer without mailbox: (null) [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Destroying SIP dialog d14b5b4-d45ce22b-8673906a@192.168.1.150 [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog 'd14b5b4-d45ce22b-8673906a@192.168.1.150' Method: SUBSCRIBE [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:10.50.0.39:5060 ---> OPTIONS sip:10.50.0.47 SIP/2.0 Via: SIP/2.0/UDP 10.50.0.39:5060;branch=z9hG4bK11575283;rport Max-Forwards: 70 From: "asterisk" ;tag=as62e1e740 To: Contact: Call-ID: 4a1d39fc2273949a3c30403e1f6a1ca9@sip01.lon01.gagenetworks.net CSeq: 102 OPTIONS User-Agent: Gage Networks sip01.lon01 Date: Wed, 20 Oct 2010 12:59:07 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 0 [ 30]: OPTIONS sip:10.50.0.47 SIP/2.0 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.50.0.39:5060;branch=z9hG4bK11575283;rport [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 3 [ 75]: From: "asterisk" ;tag=as62e1e740 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 4 [ 20]: To: [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 5 [ 34]: Contact: [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 6 [ 70]: Call-ID: 4a1d39fc2273949a3c30403e1f6a1ca9@sip01.lon01.gagenetworks.net [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 8 [ 37]: User-Agent: Gage Networks sip01.lon01 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:59:07 GMT [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: = Looking for Call ID: 4a1d39fc2273949a3c30403e1f6a1ca9@sip01.lon01.gagenetworks.net (Checking From) --From tag as62e1e740 --To-tag [Oct 20 13:58:49] DEBUG[24099] acl.c: For destination '10.50.0.39', our source address is '10.50.0.47'. [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.50.0.47:5060 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 4a1d39fc2273949a3c30403e1f6a1ca9@sip01.lon01.gagenetworks.net - OPTIONS (No RTP) [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: Looking for in default (domain 10.50.0.47) [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '' AND context = 'default' AND priority = '1' [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'default' AND priority = '1' ORDER BY exten [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '' AND context = 'default' AND priority = '1' [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:49] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'default' AND priority = '1' ORDER BY exten [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 10.50.0.39:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.50.0.39:5060;branch=z9hG4bK11575283;received=10.50.0.39;rport=5060 From: "asterisk" ;tag=as62e1e740 To: ;tag=as73873155 Call-ID: 4a1d39fc2273949a3c30403e1f6a1ca9@sip01.lon01.gagenetworks.net CSeq: 102 OPTIONS Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.50.0.39:5060 [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '4a1d39fc2273949a3c30403e1f6a1ca9@sip01.lon01.gagenetworks.net' in 32000 ms (Method: OPTIONS) [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: SIP message could not be handled, bad request: 4a1d39fc2273949a3c30403e1f6a1ca9@sip01.lon01.gagenetworks.net [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 665ef63f1610904e671362333af35eac@127.0.0.1:0 - OPTIONS (No RTP) [Oct 20 13:58:49] DEBUG[24099] acl.c: For destination '192.168.1.231', our source address is '10.50.0.47'. [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Target address 192.168.1.231:17261 is not local, substituting externaddr [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Initializing initreq for method OPTIONS - callid 26bb91f656d471db6730d726188801a8@212.62.4.230:5060 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 0 [ 38]: OPTIONS sip:3001@192.168.1.231 SIP/2.0 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK7f5be423;rport [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as535647eb [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 4 [ 28]: To: [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 5 [ 41]: Contact: [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 6 [ 59]: Call-ID: 26bb91f656d471db6730d726188801a8@212.62.4.230:5060 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:58:49 GMT [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 20 13:58:49] VERBOSE[24099] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK7f5be423;rport Max-Forwards: 70 From: "asterisk" ;tag=as535647eb To: Contact: Call-ID: 26bb91f656d471db6730d726188801a8@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81058 [Oct 20 13:58:49] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:49] VERBOSE[27662] asterisk.c: -- Remote UNIX connection disconnected [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK6d859542C9878347 From: "3001" ;tag=3B48FFC-2D9EF619 To: CSeq: 1 SUBSCRIBE Call-ID: 48ca96b8-80c803e5-8ff805de@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK6d859542C9878347 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3001" ;tag=3B48FFC-2D9EF619 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: 48ca96b8-80c803e5-8ff805de@192.168.1.231 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: = Looking for Call ID: 48ca96b8-80c803e5-8ff805de@192.168.1.231 (Checking From) --From tag 3B48FFC-2D9EF619 --To-tag [Oct 20 13:58:50] DEBUG[24099] acl.c: For destination '195.59.152.66', our source address is '10.50.0.47'. [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Target address 195.59.152.66:17261 is not local, substituting externaddr [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.62.4.230:5060 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 48ca96b8-80c803e5-8ff805de@192.168.1.231 - SUBSCRIBE (No RTP) [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 48ca96b8-80c803e5-8ff805de@192.168.1.231 [Oct 20 13:58:50] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:50] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: build_route: Contact hop: [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: list_route: hop: [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK6d859542C9878347;received=195.59.152.66;rport=17261 From: "3001" ;tag=3B48FFC-2D9EF619 To: ;tag=as39a7130f Call-ID: 48ca96b8-80c803e5-8ff805de@192.168.1.231 CSeq: 1 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75927ad0" Content-Length: 0 <------------> [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '48ca96b8-80c803e5-8ff805de@192.168.1.231' in 6400 ms (Method: SUBSCRIBE) [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:195.59.152.66:17261 ---> SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK8d5a6373A57A6034 From: "3001" ;tag=3B48FFC-2D9EF619 To: CSeq: 2 SUBSCRIBE Call-ID: 48ca96b8-80c803e5-8ff805de@192.168.1.231 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: as-feature-event User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 Accept-Language: en Accept: application/x-as-feature-event+xml Authorization: Digest username="3001", realm="asterisk", nonce="75927ad0", uri="sip:3001@212.62.4.230:5060", response="106370de43cd4b4d0fbeebf47f6f4673", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:3001@212.62.4.230:5060 SIP/2.0 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK8d5a6373A57A6034 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 2 [ 57]: From: "3001" ;tag=3B48FFC-2D9EF619 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 3 [ 32]: To: [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 4 [ 17]: CSeq: 2 SUBSCRIBE [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 5 [ 49]: Call-ID: 48ca96b8-80c803e5-8ff805de@192.168.1.231 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 6 [ 33]: Contact: [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 8 [ 23]: Event: as-feature-event [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.1.0769 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 11 [ 42]: Accept: application/x-as-feature-event+xml [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 12 [167]: Authorization: Digest username="3001", realm="asterisk", nonce="75927ad0", uri="sip:3001@212.62.4.230:5060", response="106370de43cd4b4d0fbeebf47f6f4673", algorithm=MD5 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: --- (16 headers 0 lines) --- [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: = Looking for Call ID: 48ca96b8-80c803e5-8ff805de@192.168.1.231 (Checking From) --From tag 3B48FFC-2D9EF619 --To-tag [Oct 20 13:58:50] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:50] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:50] DEBUG[24099] netsock2.c: Splitting '212.62.4.230:5060' gives... [Oct 20 13:58:50] DEBUG[24099] netsock2.c: ...host '212.62.4.230' and port '5060'. [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Got a new subscription 48ca96b8-80c803e5-8ff805de@192.168.1.231 (possibly with auth) [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Creating new subscription [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 48ca96b8-80c803e5-8ff805de@192.168.1.231 [Oct 20 13:58:50] DEBUG[24099] netsock2.c: Splitting '192.168.1.231' gives... [Oct 20 13:58:50] DEBUG[24099] netsock2.c: ...host '192.168.1.231' and port '(null)'. [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Sending to 195.59.152.66:17261 (NAT) [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: build_route: Retaining previous route: [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Found peer '3001' for '3001' from 195.59.152.66:17261 [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Looking for 3001 in DLPN_All (domain 212.62.4.230:5060) [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 195.59.152.66:17261 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.231;branch=z9hG4bK8d5a6373A57A6034;received=195.59.152.66;rport=17261 From: "3001" ;tag=3B48FFC-2D9EF619 To: ;tag=as39a7130f Call-ID: 48ca96b8-80c803e5-8ff805de@192.168.1.231 CSeq: 2 SUBSCRIBE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 195.59.152.66:17261 [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Destroying SIP dialog 48ca96b8-80c803e5-8ff805de@192.168.1.231 [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '48ca96b8-80c803e5-8ff805de@192.168.1.231' Method: SUBSCRIBE [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81058:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:50] VERBOSE[24099] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK7f5be423;rport Max-Forwards: 70 From: "asterisk" ;tag=as535647eb To: Contact: Call-ID: 26bb91f656d471db6730d726188801a8@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:50] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:51] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81058:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:51] VERBOSE[24099] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK7f5be423;rport Max-Forwards: 70 From: "asterisk" ;tag=as535647eb To: Contact: Call-ID: 26bb91f656d471db6730d726188801a8@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:51] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:52] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81058:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:52] VERBOSE[24099] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK7f5be423;rport Max-Forwards: 70 From: "asterisk" ;tag=as535647eb To: Contact: Call-ID: 26bb91f656d471db6730d726188801a8@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:52] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:53] DEBUG[24099] chan_sip.c: SIP TIMER: Not rescheduling id #81058:OPTIONS (Method 3) (No timer T1) [Oct 20 13:58:53] VERBOSE[24099] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.231:17261: OPTIONS sip:3001@192.168.1.231 SIP/2.0 Via: SIP/2.0/UDP 212.62.4.230:5060;branch=z9hG4bK7f5be423;rport Max-Forwards: 70 From: "asterisk" ;tag=as535647eb To: Contact: Call-ID: 26bb91f656d471db6730d726188801a8@212.62.4.230:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 20 Oct 2010 12:58:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Oct 20 13:58:53] DEBUG[24099] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.231:17261 [Oct 20 13:58:53] DEBUG[24099] chan_sip.c: Destroying SIP dialog 26bb91f656d471db6730d726188801a8@212.62.4.230:5060 [Oct 20 13:58:53] VERBOSE[24099] chan_sip.c: Really destroying SIP dialog '26bb91f656d471db6730d726188801a8@212.62.4.230:5060' Method: OPTIONS [Oct 20 13:58:55] VERBOSE[24099] chan_sip.c: <--- SIP read from UDP:10.50.0.39:5060 ---> OPTIONS sip:10.50.0.47 SIP/2.0 Via: SIP/2.0/UDP 10.50.0.39:5060;branch=z9hG4bK6ddb8fe3;rport Max-Forwards: 70 From: "asterisk" ;tag=as7d35be7c To: Contact: Call-ID: 2cc4eb5d5869c9172a4c3dbb4a7fcd6b@sip01.lon01.gagenetworks.net CSeq: 102 OPTIONS User-Agent: Gage Networks sip01.lon01 Date: Wed, 20 Oct 2010 12:59:14 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 0 [ 30]: OPTIONS sip:10.50.0.47 SIP/2.0 [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.50.0.39:5060;branch=z9hG4bK6ddb8fe3;rport [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 3 [ 75]: From: "asterisk" ;tag=as7d35be7c [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 4 [ 20]: To: [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 5 [ 34]: Contact: [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 6 [ 70]: Call-ID: 2cc4eb5d5869c9172a4c3dbb4a7fcd6b@sip01.lon01.gagenetworks.net [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 8 [ 37]: User-Agent: Gage Networks sip01.lon01 [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 9 [ 35]: Date: Wed, 20 Oct 2010 12:59:14 GMT [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 20 13:58:55] VERBOSE[24099] chan_sip.c: --- (15 headers 0 lines) --- [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: = Looking for Call ID: 2cc4eb5d5869c9172a4c3dbb4a7fcd6b@sip01.lon01.gagenetworks.net (Checking From) --From tag as7d35be7c --To-tag [Oct 20 13:58:55] DEBUG[24099] acl.c: For destination '10.50.0.39', our source address is '10.50.0.47'. [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.50.0.47:5060 [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Allocating new SIP dialog for 2cc4eb5d5869c9172a4c3dbb4a7fcd6b@sip01.lon01.gagenetworks.net - OPTIONS (No RTP) [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Oct 20 13:58:55] VERBOSE[24099] chan_sip.c: Looking for in default (domain 10.50.0.47) [Oct 20 13:58:55] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:55] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '' AND context = 'default' AND priority = '1' [Oct 20 13:58:55] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:55] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'default' AND priority = '1' ORDER BY exten [Oct 20 13:58:55] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:55] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten = '' AND context = 'default' AND priority = '1' [Oct 20 13:58:55] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Connection okay. [Oct 20 13:58:55] DEBUG[24099] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM gag1_exten WHERE exten LIKE '\\_%' AND context = 'default' AND priority = '1' ORDER BY exten [Oct 20 13:58:55] VERBOSE[24099] chan_sip.c: <--- Transmitting (NAT) to 10.50.0.39:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.50.0.39:5060;branch=z9hG4bK6ddb8fe3;received=10.50.0.39;rport=5060 From: "asterisk" ;tag=as7d35be7c To: ;tag=as28b554b1 Call-ID: 2cc4eb5d5869c9172a4c3dbb4a7fcd6b@sip01.lon01.gagenetworks.net CSeq: 102 OPTIONS Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.50.0.39:5060 [Oct 20 13:58:55] VERBOSE[24099] chan_sip.c: Scheduling destruction of SIP dialog '2cc4eb5d5869c9172a4c3dbb4a7fcd6b@sip01.lon01.gagenetworks.net' in 32000 ms (Method: OPTIONS) [Oct 20 13:58:55] DEBUG[24099] chan_sip.c: SIP message could not be handled, bad request: 2cc4eb5d5869c9172a4c3dbb4a7fcd6b@sip01.lon01.gagenetworks.net