[2011-02-14 20:01:36.533] VERBOSE[13628] chan_sip.c: Reloading SIP [2011-02-14 20:01:36.533] VERBOSE[13628] config.c: == Parsing '/etc/asterisk/sip.conf': [2011-02-14 20:01:36.533] DEBUG[13628] config.c: Parsing /etc/asterisk/sip.conf [2011-02-14 20:01:36.533] VERBOSE[13628] config.c: == Found [2011-02-14 20:01:36.533] DEBUG[13628] chan_sip.c: --------------- SIP reload started [2011-02-14 20:01:36.533] DEBUG[13628] chan_sip.c: --------------- Done destroying registry list [2011-02-14 20:01:36.533] DEBUG[13628] config.c: extract addr from 0.0.0.0 gives 0.0.0.0:0(0) [2011-02-14 20:01:36.533] DEBUG[13628] chan_sip.c: Setting SIP channel User-Agent Name to Asterisk UCS [2011-02-14 20:01:36.533] DEBUG[13628] acl.c: 10.0.0.0:0/255.0.0.0:0 sense 0 appended to acl for peer [2011-02-14 20:01:36.533] DEBUG[13628] acl.c: 192.168.0.0:0/255.255.0.0:0 sense 0 appended to acl for peer [2011-02-14 20:01:36.533] DEBUG[13628] acl.c: Multiple addresses. Using the first only [2011-02-14 20:01:36.534] VERBOSE[13628] netsock2.c: == Using SIP CoS mark 4 [2011-02-14 20:01:36.534] DEBUG[13628] tcptls.c: Nothing changed in SIP TCP server [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: SIP TCP server started [2011-02-14 20:01:36.534] VERBOSE[13628] dnsmgr.c: > doing dnsmgr_lookup for '80.250.3.91' [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] VERBOSE[13628] dnsmgr.c: > doing dnsmgr_lookup for '192.168.1.122' [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: SIP Seeding peer from astdb: 'michal' at michal@192.168.111.17 for 900 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Allocating new SIP dialog for 3d9af6c2527ae4d52de678d72b65d8b3@[::1]:0 - OPTIONS (No RTP) [2011-02-14 20:01:36.534] DEBUG[13628] acl.c: For destination '192.168.111.17', our source address is '192.168.111.1'. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Initializing initreq for method OPTIONS - callid 2be5a4357ae9867e69d3056960ef87c8@192.168.111.1:5060 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 0 [ 46]: OPTIONS sip:michal@192.168.111.17:5060 SIP/2.0 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK7d40579c [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 3 [ 60]: From: "asterisk" ;tag=as0a5349e9 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 4 [ 36]: To: [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 5 [ 42]: Contact: [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 6 [ 60]: Call-ID: 2be5a4357ae9867e69d3056960ef87c8@192.168.111.1:5060 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk UCS [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 9 [ 35]: Date: Mon, 14 Feb 2011 19:01:36 GMT [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2011-02-14 20:01:36.534] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.17:5060: OPTIONS sip:michal@192.168.111.17:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK7d40579c Max-Forwards: 70 From: "asterisk" ;tag=as0a5349e9 To: Contact: Call-ID: 2be5a4357ae9867e69d3056960ef87c8@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:01:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #23 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.111.17:5060 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] db.c: Unable to find key '000E08CE46CE_0' in family 'SIP/Registry' [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: SIP Seeding peer from astdb: '00260BD8E682_0' at 777@192.168.111.15 for 900 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Allocating new SIP dialog for 390206a06504b9bb1b58fcf8190896cb@[::1]:0 - OPTIONS (No RTP) [2011-02-14 20:01:36.534] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Initializing initreq for method OPTIONS - callid 526cd090323326e61b8c334b7c015ae1@192.168.111.1:5060 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 0 [ 57]: OPTIONS sip:777@192.168.111.15:5060;transport=udp SIP/2.0 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK6485753a [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 3 [ 60]: From: "asterisk" ;tag=as35e5ead0 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 4 [ 47]: To: [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 5 [ 42]: Contact: [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 6 [ 60]: Call-ID: 526cd090323326e61b8c334b7c015ae1@192.168.111.1:5060 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk UCS [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 9 [ 35]: Date: Mon, 14 Feb 2011 19:01:36 GMT [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2011-02-14 20:01:36.534] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.15:5060: OPTIONS sip:777@192.168.111.15:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK6485753a Max-Forwards: 70 From: "asterisk" ;tag=as35e5ead0 To: Contact: Call-ID: 526cd090323326e61b8c334b7c015ae1@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:01:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #26 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [2011-02-14 20:01:36.534] DEBUG[13628] chan_sip.c: SIP Seeding peer from astdb: 'zdenek' at zdenek@192.168.111.22 for 900 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Allocating new SIP dialog for 5b3aded04d6014ba3e27de5a013d95f7@[::1]:0 - OPTIONS (No RTP) [2011-02-14 20:01:36.535] DEBUG[13628] acl.c: For destination '192.168.111.22', our source address is '192.168.111.1'. [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Initializing initreq for method OPTIONS - callid 7acd03d3290f0e76519db1586d08f871@192.168.111.1:5060 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 0 [ 46]: OPTIONS sip:zdenek@192.168.111.22:5063 SIP/2.0 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK464d3a01 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 3 [ 60]: From: "asterisk" ;tag=as2b0e27ad [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 4 [ 36]: To: [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 5 [ 42]: Contact: [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 6 [ 60]: Call-ID: 7acd03d3290f0e76519db1586d08f871@192.168.111.1:5060 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk UCS [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 9 [ 35]: Date: Mon, 14 Feb 2011 19:01:36 GMT [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2011-02-14 20:01:36.535] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.22:5063: OPTIONS sip:zdenek@192.168.111.22:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK464d3a01 Max-Forwards: 70 From: "asterisk" ;tag=as2b0e27ad To: Contact: Call-ID: 7acd03d3290f0e76519db1586d08f871@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:01:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #29 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.111.22:5063 [2011-02-14 20:01:36.535] VERBOSE[13628] config.c: == Parsing '/etc/asterisk/sip_notify.conf': [2011-02-14 20:01:36.535] DEBUG[13628] config.c: Parsing /etc/asterisk/sip_notify.conf [2011-02-14 20:01:36.535] VERBOSE[13628] config.c: == Found [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: SIP reload_config done...Runtime= 0 sec [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: Asterisk Schedule Dump (9 in Q, 31 Total, 2 Cache, 10 high-water) [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: ============================================================= [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |ID Callback Data Time (sec:ms) | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: +-----+-----------------+-----------------+-----------------+ [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0023 | 0x7fb5d374f680 | 0xd4fd70 | 000000 : 999153 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0029 | 0x7fb5d374f680 | 0xf12bd0 | 000000 : 999848 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0026 | 0x7fb5d374f680 | 0xf0a290 | 000000 : 999539 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0024 | 0x7fb5d37426c0 | 0xe5bf08 | 000003 : 999239 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0030 | 0x7fb5d37426c0 | 0xe687c8 | 000003 : 999886 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0025 | 0x7fb5d374bf00 | 0xe5bf08 | 000909 : 999242 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0028 | 0x7fb5d374bf00 | 0xe64498 | 000909 : 999587 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0027 | 0x7fb5d37426c0 | 0xe64498 | 000003 : 999585 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: |0031 | 0x7fb5d374bf00 | 0xe687c8 | 000909 : 999888 | [2011-02-14 20:01:36.535] DEBUG[13628] sched.c: ============================================================= [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: --------------- Done destroying pruned peers [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: do_reload finished. peer poke/prune reg contact time = 0 sec. [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: --------------- SIP reload done [2011-02-14 20:01:36.535] NOTICE[13628] chan_sip.c: Still have a QUALIFY dialog active, deleting [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Destroying SIP dialog 2be5a4357ae9867e69d3056960ef87c8@192.168.111.1:5060 [2011-02-14 20:01:36.535] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '2be5a4357ae9867e69d3056960ef87c8@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Allocating new SIP dialog for 272e8aff65e196f81f12e567775510f1@[::1]:0 - OPTIONS (No RTP) [2011-02-14 20:01:36.535] DEBUG[13628] acl.c: For destination '192.168.111.17', our source address is '192.168.111.1'. [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Initializing initreq for method OPTIONS - callid 629495d612a75b6a025f4a750008fd61@192.168.111.1:5060 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 0 [ 46]: OPTIONS sip:michal@192.168.111.17:5060 SIP/2.0 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK6eae8f89 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 3 [ 60]: From: "asterisk" ;tag=as538c7bc0 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 4 [ 36]: To: [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 5 [ 42]: Contact: [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 6 [ 60]: Call-ID: 629495d612a75b6a025f4a750008fd61@192.168.111.1:5060 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk UCS [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 9 [ 35]: Date: Mon, 14 Feb 2011 19:01:36 GMT [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2011-02-14 20:01:36.535] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.17:5060: OPTIONS sip:michal@192.168.111.17:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK6eae8f89 Max-Forwards: 70 From: "asterisk" ;tag=as538c7bc0 To: Contact: Call-ID: 629495d612a75b6a025f4a750008fd61@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:01:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #38 [2011-02-14 20:01:36.535] DEBUG[13628] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.111.17:5060 [2011-02-14 20:01:36.545] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.17:5060 ---> SIP/2.0 200 OK To: ;tag=bc0c28782f1665a0i0 From: "asterisk" ;tag=as0a5349e9 Call-ID: 2be5a4357ae9867e69d3056960ef87c8@192.168.111.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK7d40579c Server: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 1 [ 59]: To: ;tag=bc0c28782f1665a0i0 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 2 [ 60]: From: "asterisk" ;tag=as0a5349e9 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 3 [ 60]: Call-ID: 2be5a4357ae9867e69d3056960ef87c8@192.168.111.1:5060 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 4 [ 17]: CSeq: 102 OPTIONS [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK7d40579c [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 6 [ 34]: Server: Linksys/SPA3102-5.1.10(GW) [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 8 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 9 [ 29]: Supported: x-sipura, replaces [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 10 [ 0]: [2011-02-14 20:01:36.546] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 2be5a4357ae9867e69d3056960ef87c8@192.168.111.1:5060 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 431 [2011-02-14 20:01:36.546] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> SIP/2.0 200 OK To: ;tag=9d9812c22b8579e0i3 From: "asterisk" ;tag=as2b0e27ad Call-ID: 7acd03d3290f0e76519db1586d08f871@192.168.111.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK464d3a01 Server: Cisco/SPA504G-7.4.3a Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces <-------------> [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 1 [ 59]: To: ;tag=9d9812c22b8579e0i3 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 2 [ 60]: From: "asterisk" ;tag=as2b0e27ad [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 3 [ 60]: Call-ID: 7acd03d3290f0e76519db1586d08f871@192.168.111.1:5060 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 4 [ 17]: CSeq: 102 OPTIONS [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK464d3a01 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 6 [ 28]: Server: Cisco/SPA504G-7.4.3a [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 8 [ 69]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 9 [ 19]: Supported: replaces [2011-02-14 20:01:36.546] DEBUG[13628] chan_sip.c: Header 10 [ 0]: [2011-02-14 20:01:36.546] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:01:36.547] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 [2011-02-14 20:01:36.547] DEBUG[13628] chan_sip.c: Stopping retransmission on '7acd03d3290f0e76519db1586d08f871@192.168.111.1:5060' of Request 102: Match Found [2011-02-14 20:01:36.547] DEBUG[13628] chan_sip.c: Destroying SIP dialog 7acd03d3290f0e76519db1586d08f871@192.168.111.1:5060 [2011-02-14 20:01:36.547] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '7acd03d3290f0e76519db1586d08f871@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:01:36.550] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.17:5060 ---> SIP/2.0 200 OK To: ;tag=bc0c28782f1665a0i0 From: "asterisk" ;tag=as538c7bc0 Call-ID: 629495d612a75b6a025f4a750008fd61@192.168.111.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK6eae8f89 Server: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 1 [ 59]: To: ;tag=bc0c28782f1665a0i0 [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 2 [ 60]: From: "asterisk" ;tag=as538c7bc0 [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 3 [ 60]: Call-ID: 629495d612a75b6a025f4a750008fd61@192.168.111.1:5060 [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 4 [ 17]: CSeq: 102 OPTIONS [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK6eae8f89 [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 6 [ 34]: Server: Linksys/SPA3102-5.1.10(GW) [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 8 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 9 [ 29]: Supported: x-sipura, replaces [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Header 10 [ 0]: [2011-02-14 20:01:36.550] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38 [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Stopping retransmission on '629495d612a75b6a025f4a750008fd61@192.168.111.1:5060' of Request 102: Match Found [2011-02-14 20:01:36.550] DEBUG[13628] chan_sip.c: Destroying SIP dialog 629495d612a75b6a025f4a750008fd61@192.168.111.1:5060 [2011-02-14 20:01:36.550] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '629495d612a75b6a025f4a750008fd61@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:01:36.562] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51682 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK6485753a From: "asterisk" ;tag=as35e5ead0 To: ;tag=00260bd8e68208be369d945d-36d370f0 Call-ID: 526cd090323326e61b8c334b7c015ae1@192.168.111.1:5060 Date: Mon, 14 Feb 2011 19:01:36 GMT CSeq: 102 OPTIONS Server: Cisco-CP7945G/8.5.3 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces,join,norefersub Content-Length: 287 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 17408 0 IN IP4 192.168.111.15 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 116 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK6485753a [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 2 [ 60]: From: "asterisk" ;tag=as35e5ead0 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 3 [ 85]: To: ;tag=00260bd8e68208be369d945d-36d370f0 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 4 [ 60]: Call-ID: 526cd090323326e61b8c334b7c015ae1@192.168.111.1:5060 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 5 [ 35]: Date: Mon, 14 Feb 2011 19:01:36 GMT [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 7 [ 27]: Server: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 8 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 9 [ 31]: Allow-Events: kpml,dialog,refer [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 10 [ 61]: Accept: application/sdp,multipart/mixed,multipart/alternative [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 11 [ 25]: Accept-Encoding: identity [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 12 [ 19]: Accept-Language: en [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 13 [ 35]: Supported: replaces,join,norefersub [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 14 [ 19]: Content-Length: 287 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 16 [ 46]: Content-Disposition: session;handling=optional [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Header 17 [ 0]: [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 0 [ 3]: v=0 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 1 [ 43]: o=Cisco-SIPUA 17408 0 IN IP4 192.168.111.15 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 2 [ 10]: s=SIP Call [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 3 [ 5]: t=0 0 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 4 [ 32]: m=audio 0 RTP/AVP 0 8 18 116 101 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 5 [ 20]: a=rtpmap:0 PCMU/8000 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 8 [ 19]: a=fmtp:18 annexb=no [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 9 [ 22]: a=rtpmap:116 iLBC/8000 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 10 [ 18]: a=fmtp:116 mode=20 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-15 [2011-02-14 20:01:36.563] VERBOSE[13628] chan_sip.c: --- (17 headers 13 lines) --- [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #26 [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Stopping retransmission on '526cd090323326e61b8c334b7c015ae1@192.168.111.1:5060' of Request 102: Match Found [2011-02-14 20:01:36.563] DEBUG[13628] chan_sip.c: Destroying SIP dialog 526cd090323326e61b8c334b7c015ae1@192.168.111.1:5060 [2011-02-14 20:01:36.563] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '526cd090323326e61b8c334b7c015ae1@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:01:40.043] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> INVITE sip:222@192.168.111.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK5656874f From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:40 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7945G/8.5.3 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "777" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 328 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 27464 0 IN IP4 192.168.111.15 s=SIP Call t=0 0 m=audio 24554 RTP/AVP 0 8 18 116 101 c=IN IP4 192.168.111.15 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 0 [ 36]: INVITE sip:222@192.168.111.1 SIP/2.0 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK5656874f [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 3 [ 27]: To: [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:40 GMT [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 7 [ 16]: CSeq: 101 INVITE [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 9 [ 52]: Contact: [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 10 [ 12]: Expires: 180 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 12 [ 75]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 13 [113]: Remote-Party-ID: "777" ;party=calling;id-type=subscriber;privacy=off;screen=yes [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 14 [ 35]: Supported: replaces,join,norefersub [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 15 [ 25]: Allow-Events: kpml,dialog [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 16 [ 19]: Content-Length: 328 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 17 [ 29]: Content-Type: application/sdp [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 18 [ 46]: Content-Disposition: session;handling=optional [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Header 19 [ 0]: [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 0 [ 3]: v=0 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 1 [ 43]: o=Cisco-SIPUA 27464 0 IN IP4 192.168.111.15 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 2 [ 10]: s=SIP Call [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 3 [ 5]: t=0 0 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 4 [ 36]: m=audio 24554 RTP/AVP 0 8 18 116 101 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 5 [ 23]: c=IN IP4 192.168.111.15 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 10 [ 22]: a=rtpmap:116 iLBC/8000 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 11 [ 18]: a=fmtp:116 mode=20 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-15 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Body 14 [ 10]: a=sendrecv [2011-02-14 20:01:40.043] VERBOSE[13628] chan_sip.c: --- (19 headers 15 lines) --- [2011-02-14 20:01:40.043] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Allocating new SIP dialog for 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 - INVITE (No RTP) [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-02-14 20:01:40.043] DEBUG[13628] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces,join,norefersub" [2011-02-14 20:01:40.043] DEBUG[13628] sip/reqresp_parser.c: Found SIP option: -replaces- [2011-02-14 20:01:40.043] DEBUG[13628] sip/reqresp_parser.c: Matched SIP option: replaces [2011-02-14 20:01:40.043] DEBUG[13628] sip/reqresp_parser.c: Found SIP option: -join- [2011-02-14 20:01:40.043] DEBUG[13628] sip/reqresp_parser.c: Matched SIP option: join [2011-02-14 20:01:40.043] DEBUG[13628] sip/reqresp_parser.c: Found SIP option: -norefersub- [2011-02-14 20:01:40.043] DEBUG[13628] sip/reqresp_parser.c: Matched SIP option: norefersub [2011-02-14 20:01:40.043] VERBOSE[13628] chan_sip.c: Sending to 192.168.111.15:5060 (no NAT) [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Initializing initreq for method INVITE - callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:40.043] VERBOSE[13628] chan_sip.c: Using INVITE request as basis request - 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:40.043] VERBOSE[13628] chan_sip.c: Found peer '00260BD8E682_0' for '00260BD8E682_0' from 192.168.111.15:51177 [2011-02-14 20:01:40.043] VERBOSE[13628] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.111.15:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK5656874f;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as111aebf3 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 101 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3708fdc0" Content-Length: 0 <------------> [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #43 [2011-02-14 20:01:40.043] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:40.043] VERBOSE[13628] chan_sip.c: Scheduling destruction of SIP dialog '00260bd8-e682002b-93700064-08db6b13@192.168.111.15' in 6400 ms (Method: INVITE) [2011-02-14 20:01:40.049] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51683 ---> ACK sip:222@192.168.111.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK5656874f From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as111aebf3 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Date: Mon, 14 Feb 2011 19:01:40 GMT CSeq: 101 ACK Content-Length: 0 <-------------> [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 0 [ 33]: ACK sip:222@192.168.111.1 SIP/2.0 [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK5656874f [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as111aebf3 [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 5 [ 35]: Date: Mon, 14 Feb 2011 19:01:40 GMT [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 6 [ 13]: CSeq: 101 ACK [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Header 8 [ 0]: [2011-02-14 20:01:40.049] VERBOSE[13628] chan_sip.c: --- (8 headers 0 lines) --- [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #43 [2011-02-14 20:01:40.049] DEBUG[13628] chan_sip.c: Stopping retransmission on '00260bd8-e682002b-93700064-08db6b13@192.168.111.15' of Response 101: Match Found [2011-02-14 20:01:40.052] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> INVITE sip:222@192.168.111.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:40 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7945G/8.5.3 Contact: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1",response="400da357a8d40649421bead6ae4a8588",nonce="3708fdc0",algorithm=MD5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "777" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 328 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 27464 0 IN IP4 192.168.111.15 s=SIP Call t=0 0 m=audio 24554 RTP/AVP 0 8 18 116 101 c=IN IP4 192.168.111.15 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 0 [ 36]: INVITE sip:222@192.168.111.1 SIP/2.0 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 3 [ 27]: To: [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:40 GMT [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 9 [ 52]: Contact: [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 10 [167]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1",response="400da357a8d40649421bead6ae4a8588",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 11 [ 12]: Expires: 180 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 12 [ 23]: Accept: application/sdp [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 13 [ 75]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 14 [113]: Remote-Party-ID: "777" ;party=calling;id-type=subscriber;privacy=off;screen=yes [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 15 [ 35]: Supported: replaces,join,norefersub [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 16 [ 25]: Allow-Events: kpml,dialog [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 17 [ 19]: Content-Length: 328 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 18 [ 29]: Content-Type: application/sdp [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 19 [ 46]: Content-Disposition: session;handling=optional [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Header 20 [ 0]: [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 0 [ 3]: v=0 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 1 [ 43]: o=Cisco-SIPUA 27464 0 IN IP4 192.168.111.15 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 2 [ 10]: s=SIP Call [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 3 [ 5]: t=0 0 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 4 [ 36]: m=audio 24554 RTP/AVP 0 8 18 116 101 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 5 [ 23]: c=IN IP4 192.168.111.15 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 10 [ 22]: a=rtpmap:116 iLBC/8000 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 11 [ 18]: a=fmtp:116 mode=20 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-15 [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Body 14 [ 10]: a=sendrecv [2011-02-14 20:01:40.052] VERBOSE[13628] chan_sip.c: --- (20 headers 15 lines) --- [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-02-14 20:01:40.052] VERBOSE[13628] chan_sip.c: Sending to 192.168.111.15:5060 (no NAT) [2011-02-14 20:01:40.052] DEBUG[13628] chan_sip.c: Initializing initreq for method INVITE - callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:40.052] VERBOSE[13628] chan_sip.c: Using INVITE request as basis request - 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:40.052] VERBOSE[13628] chan_sip.c: Found peer '00260BD8E682_0' for '00260BD8E682_0' from 192.168.111.15:51177 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fb5ae8ba358' [2011-02-14 20:01:40.053] DEBUG[13628] res_rtp_asterisk.c: Allocated port 16170 for RTP instance '0x7fb5ae8ba358' [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: RTP instance '0x7fb5ae8ba358' is setup and ready to go [2011-02-14 20:01:40.053] DEBUG[13628] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7fb5ae8ba358' [2011-02-14 20:01:40.053] VERBOSE[13628] netsock2.c: == Using SIP RTP CoS mark 5 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Setting NAT on RTP to Off [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing session-level SDP o=Cisco-SIPUA 27464 0 IN IP4 192.168.111.15... UNSUPPORTED. [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found RTP audio format 0 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Setting payload 0 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found RTP audio format 8 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Setting payload 8 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found RTP audio format 18 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Setting payload 18 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found RTP audio format 116 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Setting payload 116 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found RTP audio format 101 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Setting payload 101 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 192.168.111.15... OK. [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found audio description format PCMU for ID 0 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found audio description format PCMA for ID 8 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found audio description format G729 for ID 18 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found audio description format iLBC for ID 116 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:116 iLBC/8000... OK. [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=fmtp:116 mode=20... UNSUPPORTED. [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Found audio description format telephone-event for ID 101 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Incorporating payload 0 on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Incorporating payload 8 on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Incorporating payload 18 on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Incorporating payload 101 on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Incorporating payload 116 on 0x7fb5b30b5f80 [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Capabilities: us - 0x8 (alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2011-02-14 20:01:40.053] DEBUG[13628] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fb5ae8ba358' [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Peer audio RTP is at port 192.168.111.15:24554 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Copying payload 0 from 0x7fb5b30b5f80 to 0x7fb5ae8ba520 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Copying payload 8 from 0x7fb5b30b5f80 to 0x7fb5ae8ba520 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Copying payload 18 from 0x7fb5b30b5f80 to 0x7fb5ae8ba520 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Copying payload 101 from 0x7fb5b30b5f80 to 0x7fb5ae8ba520 [2011-02-14 20:01:40.053] DEBUG[13628] rtp_engine.c: Copying payload 116 from 0x7fb5b30b5f80 to 0x7fb5ae8ba520 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: We're settling with these formats: 0x8 (alaw) [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Checking SIP call limits for device 777 [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Updating call counter for incoming call [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: Looking for 222 in filter-2 (domain 192.168.111.1) [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: *** Our native formats are 0x8 (alaw) [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: *** Our capabilities are 0x8 (alaw) [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: This channel will not be able to handle video. [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: build_route: Contact hop: [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: list_route: hop: [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: SIP/00260BD8E682_0-00000000: New call is still down.... Trying... [2011-02-14 20:01:40.053] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [2011-02-14 20:01:40.053] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:40.053] DEBUG[13619] devicestate.c: No provider found, checking channel drivers for SIP - 00260BD8E682_0 [2011-02-14 20:01:40.053] DEBUG[13619] chan_sip.c: Checking device state for peer 00260BD8E682_0 [2011-02-14 20:01:40.054] DEBUG[13619] devicestate.c: Changing state for SIP/00260BD8E682_0 - state 1 (Not in use) [2011-02-14 20:01:40.054] DEBUG[13619] devicestate.c: device 'SIP/00260BD8E682_0' state '1' [2011-02-14 20:01:40.054] DEBUG[13635] app_queue.c: Device 'SIP/00260BD8E682_0' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Launching 'NoOp' [2011-02-14 20:01:40.055] VERBOSE[13654] pbx.c: -- Executing [222@filter-2:1] NoOp("SIP/00260BD8E682_0-00000000", "") in new stack [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Launching 'Set' [2011-02-14 20:01:40.055] VERBOSE[13654] pbx.c: -- Executing [222@filter-2:2] Set("SIP/00260BD8E682_0-00000000", "EXT_ID_DST=4") in new stack [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Launching 'Set' [2011-02-14 20:01:40.055] VERBOSE[13654] pbx.c: -- Executing [222@filter-2:3] Set("SIP/00260BD8E682_0-00000000", "CALL_DST=1") in new stack [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Launching 'Set' [2011-02-14 20:01:40.055] VERBOSE[13654] pbx.c: -- Executing [222@filter-2:4] Set("SIP/00260BD8E682_0-00000000", "GROUP_ID_DST=3") in new stack [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Function result is '777' [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Launching 'Set' [2011-02-14 20:01:40.055] VERBOSE[13654] pbx.c: -- Executing [222@filter-2:5] Set("SIP/00260BD8E682_0-00000000", "CDR(userfield)=2,3,1,4,777,222,1,1,") in new stack [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Launching 'Set' [2011-02-14 20:01:40.055] VERBOSE[13654] pbx.c: -- Executing [222@filter-2:6] Set("SIP/00260BD8E682_0-00000000", "CONNECTEDLINE(name)=Michal Kudlič") in new stack [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Launching 'Set' [2011-02-14 20:01:40.055] VERBOSE[13654] pbx.c: -- Executing [222@filter-2:7] Set("SIP/00260BD8E682_0-00000000", "CONNECTEDLINE(num)=222") in new stack [2011-02-14 20:01:40.055] DEBUG[13654] pbx.c: Launching 'Dial' [2011-02-14 20:01:40.055] VERBOSE[13654] pbx.c: -- Executing [222@filter-2:8] Dial("SIP/00260BD8E682_0-00000000", "SIP/michal,,t") in new stack [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: Allocating new SIP dialog for 446961f80ff77a817f28108e7a065a99@[::1]:0 - INVITE (No RTP) [2011-02-14 20:01:40.055] DEBUG[13654] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xf121d8' [2011-02-14 20:01:40.055] DEBUG[13654] res_rtp_asterisk.c: Allocated port 11176 for RTP instance '0xf121d8' [2011-02-14 20:01:40.055] DEBUG[13654] rtp_engine.c: RTP instance '0xf121d8' is setup and ready to go [2011-02-14 20:01:40.055] DEBUG[13654] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xf121d8' [2011-02-14 20:01:40.055] VERBOSE[13654] netsock2.c: == Using SIP RTP CoS mark 5 [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: Setting NAT on RTP to Off [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2011-02-14 20:01:40.055] DEBUG[13654] acl.c: For destination '192.168.111.17', our source address is '192.168.111.1'. [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: *** Our native formats are 0x8 (alaw) [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: *** Our capabilities are 0x8 (alaw) [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [2011-02-14 20:01:40.055] DEBUG[13654] chan_sip.c: This channel will not be able to handle video. [2011-02-14 20:01:40.056] DEBUG[13654] rtp_engine.c: Seeded SDP of 'SIP/michal-00000001' with that of 'SIP/00260BD8E682_0-00000000' [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable DIALEDTIME. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable ANSWEREDTIME. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable DIALEDPEERNAME. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable DIALEDPEERNUMBER. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable DIALSTATUS. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable GROUP_ID_DST. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable CALL_DST. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable EXT_ID_DST. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable GROUP_ID_SRC. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable CALL_SRC. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable EXT_ID_SRC. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable EXT_NAME. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable EXT. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable SIPCALLID. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable SIPDOMAIN. [2011-02-14 20:01:40.056] DEBUG[13654] channel.c: Not copying variable SIPURI. [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Outgoing Call for michal [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Updating call counter for outgoing call [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: False Text flag: False [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [2011-02-14 20:01:40.056] VERBOSE[13654] chan_sip.c: Audio is at 5060 [2011-02-14 20:01:40.056] VERBOSE[13654] chan_sip.c: Adding codec 0x8 (alaw) to SDP [2011-02-14 20:01:40.056] VERBOSE[13654] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: -- Done with adding codecs to SDP [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Initializing initreq for method INVITE - callid 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 0 [ 45]: INVITE sip:michal@192.168.111.17:5060 SIP/2.0 [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 3 [ 57]: From: "Jan Klepal" ;tag=as19b4b5f0 [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 4 [ 36]: To: [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 5 [ 37]: Contact: [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 6 [ 60]: Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk UCS [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 9 [ 35]: Date: Mon, 14 Feb 2011 19:01:40 GMT [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 12 [ 89]: Remote-Party-ID: "Jan Klepal" ;party=calling;privacy=off;screen=no [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-02-14 20:01:40.056] VERBOSE[13654] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.17:5060: INVITE sip:michal@192.168.111.17:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 Max-Forwards: 70 From: "Jan Klepal" ;tag=as19b4b5f0 To: Contact: Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 CSeq: 102 INVITE User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:01:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "Jan Klepal" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 230 v=0 o=root 1407100411 1407100411 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 11176 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #46 [2011-02-14 20:01:40.056] DEBUG[13654] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.111.17:5060 [2011-02-14 20:01:40.056] VERBOSE[13654] app_dial.c: -- Called michal [2011-02-14 20:01:40.067] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.17:5060 ---> SIP/2.0 100 Trying To: From: "Jan Klepal" ;tag=as19b4b5f0 Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 Server: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 <-------------> [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 1 [ 36]: To: [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 2 [ 57]: From: "Jan Klepal" ;tag=as19b4b5f0 [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 3 [ 60]: Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 6 [ 34]: Server: Linksys/SPA3102-5.1.10(GW) [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: Header 8 [ 0]: [2011-02-14 20:01:40.067] VERBOSE[13628] chan_sip.c: --- (8 headers 0 lines) --- [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: *** SIP TIMER: Cancelling retransmission #46 - INVITE (got response) [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060' Request 102: Found [2011-02-14 20:01:40.067] DEBUG[13628] chan_sip.c: SIP response 100 to standard invite [2011-02-14 20:01:40.073] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.17:5060 ---> SIP/2.0 180 Ringing To: ;tag=541822245c9d4949i0 From: "Jan Klepal" ;tag=as19b4b5f0 Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 Contact: Externi Volani Server: Linksys/SPA3102-5.1.10(GW) Remote-Party-ID: Externi Volani ;screen=yes;party=called Content-Length: 0 <-------------> [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 1 [ 59]: To: ;tag=541822245c9d4949i0 [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 2 [ 57]: From: "Jan Klepal" ;tag=as19b4b5f0 [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 3 [ 60]: Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 6 [ 56]: Contact: Externi Volani [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 7 [ 34]: Server: Linksys/SPA3102-5.1.10(GW) [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 8 [ 82]: Remote-Party-ID: Externi Volani ;screen=yes;party=called [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: Header 10 [ 0]: [2011-02-14 20:01:40.073] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060' Request 102: Found [2011-02-14 20:01:40.073] DEBUG[13628] chan_sip.c: SIP response 180 to standard invite [2011-02-14 20:01:40.073] DEBUG[13619] devicestate.c: No provider found, checking channel drivers for SIP - michal [2011-02-14 20:01:40.073] DEBUG[13619] chan_sip.c: Checking device state for peer michal [2011-02-14 20:01:40.073] DEBUG[13619] devicestate.c: Changing state for SIP/michal - state 1 (Not in use) [2011-02-14 20:01:40.073] DEBUG[13619] devicestate.c: device 'SIP/michal' state '1' [2011-02-14 20:01:40.073] VERBOSE[13654] app_dial.c: -- SIP/michal-00000001 is ringing [2011-02-14 20:01:40.073] DEBUG[13654] rtp_engine.c: Setting early bridge SDP of 'SIP/00260BD8E682_0-00000000' with that of 'SIP/michal-00000001' [2011-02-14 20:01:40.073] DEBUG[13635] app_queue.c: Device 'SIP/michal' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-02-14 20:01:40.073] VERBOSE[13654] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michal Kudlič" ;party=called;privacy=off;screen=no Content-Length: 0 <------------> [2011-02-14 20:01:40.073] DEBUG[13654] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:43.664] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> INVITE sip:*8@192.168.111.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-e92895be From: ;tag=9eded97e1481156eo3 To: Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 101 INVITE Max-Forwards: 70 Contact: Expires: 240 User-Agent: Cisco/SPA504G-7.4.3a Content-Length: 395 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 7765 7765 IN IP4 192.168.111.22 s=- c=IN IP4 192.168.111.22 t=0 0 m=audio 16454 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 0 [ 35]: INVITE sip:*8@192.168.111.1 SIP/2.0 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-e92895be [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 2 [ 55]: From: ;tag=9eded97e1481156eo3 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 3 [ 26]: To: [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 4 [ 41]: Call-ID: cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 5 [ 16]: CSeq: 101 INVITE [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 7 [ 41]: Contact: [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 8 [ 12]: Expires: 240 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 9 [ 32]: User-Agent: Cisco/SPA504G-7.4.3a [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 10 [ 19]: Content-Length: 395 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 11 [ 69]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 12 [ 19]: Supported: replaces [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Header 14 [ 0]: [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 0 [ 3]: v=0 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 1 [ 35]: o=- 7765 7765 IN IP4 192.168.111.22 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 2 [ 3]: s=- [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.111.22 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 5 [ 45]: m=audio 16454 RTP/AVP 0 2 8 9 18 96 97 98 101 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 7 [ 23]: a=rtpmap:2 G726-32/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 9 [ 20]: a=rtpmap:9 G722/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 10 [ 22]: a=rtpmap:18 G729a/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 11 [ 24]: a=rtpmap:96 G726-40/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 12 [ 24]: a=rtpmap:97 G726-24/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 13 [ 24]: a=rtpmap:98 G726-16/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-15 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 16 [ 10]: a=ptime:30 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Body 17 [ 10]: a=sendrecv [2011-02-14 20:01:43.664] VERBOSE[13628] chan_sip.c: --- (14 headers 18 lines) --- [2011-02-14 20:01:43.664] DEBUG[13628] acl.c: For destination '192.168.111.22', our source address is '192.168.111.1'. [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Allocating new SIP dialog for cc43f6be-197c992e@192.168.111.22 - INVITE (No RTP) [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-02-14 20:01:43.664] DEBUG[13628] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces" [2011-02-14 20:01:43.664] DEBUG[13628] sip/reqresp_parser.c: Found SIP option: -replaces- [2011-02-14 20:01:43.664] DEBUG[13628] sip/reqresp_parser.c: Matched SIP option: replaces [2011-02-14 20:01:43.664] VERBOSE[13628] chan_sip.c: Sending to 192.168.111.22:5063 (no NAT) [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Initializing initreq for method INVITE - callid cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.664] VERBOSE[13628] chan_sip.c: Using INVITE request as basis request - cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.664] VERBOSE[13628] chan_sip.c: Found peer 'zdenek' for 'zdenek' from 192.168.111.22:5063 [2011-02-14 20:01:43.664] VERBOSE[13628] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.111.22:5063 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-e92895be;received=192.168.111.22 From: ;tag=9eded97e1481156eo3 To: ;tag=as04959fae Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 101 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f49c32d" Content-Length: 0 <------------> [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #49 [2011-02-14 20:01:43.664] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.111.22:5063 [2011-02-14 20:01:43.664] VERBOSE[13628] chan_sip.c: Scheduling destruction of SIP dialog 'cc43f6be-197c992e@192.168.111.22' in 6400 ms (Method: INVITE) [2011-02-14 20:01:43.670] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> ACK sip:*8@192.168.111.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-e92895be From: ;tag=9eded97e1481156eo3 To: ;tag=as04959fae Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 101 ACK Max-Forwards: 70 Contact: User-Agent: Cisco/SPA504G-7.4.3a Content-Length: 0 <-------------> [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 0 [ 32]: ACK sip:*8@192.168.111.1 SIP/2.0 [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-e92895be [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 2 [ 55]: From: ;tag=9eded97e1481156eo3 [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 3 [ 41]: To: ;tag=as04959fae [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 4 [ 41]: Call-ID: cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 5 [ 13]: CSeq: 101 ACK [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 7 [ 41]: Contact: [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 8 [ 32]: User-Agent: Cisco/SPA504G-7.4.3a [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Header 10 [ 0]: [2011-02-14 20:01:43.670] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #49 [2011-02-14 20:01:43.670] DEBUG[13628] chan_sip.c: Stopping retransmission on 'cc43f6be-197c992e@192.168.111.22' of Response 101: Match Found [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> INVITE sip:*8@192.168.111.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-3ce76872 From: ;tag=9eded97e1481156eo3 To: Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="zdenek",realm="asterisk",nonce="6f49c32d",uri="sip:*8@192.168.111.1",algorithm=MD5,response="1941ebe568e55de0f12f6e6c375341ff" Contact: Expires: 240 User-Agent: Cisco/SPA504G-7.4.3a Content-Length: 395 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 7765 7765 IN IP4 192.168.111.22 s=- c=IN IP4 192.168.111.22 t=0 0 m=audio 16454 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 0 [ 35]: INVITE sip:*8@192.168.111.1 SIP/2.0 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-3ce76872 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 2 [ 55]: From: ;tag=9eded97e1481156eo3 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 3 [ 26]: To: [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 4 [ 41]: Call-ID: cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 7 [158]: Authorization: Digest username="zdenek",realm="asterisk",nonce="6f49c32d",uri="sip:*8@192.168.111.1",algorithm=MD5,response="1941ebe568e55de0f12f6e6c375341ff" [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 8 [ 41]: Contact: [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 9 [ 12]: Expires: 240 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 10 [ 32]: User-Agent: Cisco/SPA504G-7.4.3a [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 11 [ 19]: Content-Length: 395 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 12 [ 69]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 13 [ 19]: Supported: replaces [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Header 15 [ 0]: [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 0 [ 3]: v=0 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 1 [ 35]: o=- 7765 7765 IN IP4 192.168.111.22 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 2 [ 3]: s=- [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.111.22 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 5 [ 45]: m=audio 16454 RTP/AVP 0 2 8 9 18 96 97 98 101 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 7 [ 23]: a=rtpmap:2 G726-32/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 9 [ 20]: a=rtpmap:9 G722/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 10 [ 22]: a=rtpmap:18 G729a/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 11 [ 24]: a=rtpmap:96 G726-40/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 12 [ 24]: a=rtpmap:97 G726-24/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 13 [ 24]: a=rtpmap:98 G726-16/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-15 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 16 [ 10]: a=ptime:30 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Body 17 [ 10]: a=sendrecv [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: --- (15 headers 18 lines) --- [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Sending to 192.168.111.22:5063 (no NAT) [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Initializing initreq for method INVITE - callid cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Using INVITE request as basis request - cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found peer 'zdenek' for 'zdenek' from 192.168.111.22:5063 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xf26e68' [2011-02-14 20:01:43.675] DEBUG[13628] res_rtp_asterisk.c: Allocated port 11228 for RTP instance '0xf26e68' [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: RTP instance '0xf26e68' is setup and ready to go [2011-02-14 20:01:43.675] DEBUG[13628] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xf26e68' [2011-02-14 20:01:43.675] VERBOSE[13628] netsock2.c: == Using SIP RTP CoS mark 5 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Setting NAT on RTP to Off [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing session-level SDP o=- 7765 7765 IN IP4 192.168.111.22... UNSUPPORTED. [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.111.22... OK. [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 0 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Setting payload 0 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 2 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 8 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Setting payload 8 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 9 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Setting payload 9 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 18 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Setting payload 18 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 96 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 97 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Setting payload 97 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 98 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Setting payload 98 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found RTP audio format 101 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Setting payload 101 based on m type on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format PCMU for ID 0 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format G726-32 for ID 2 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format PCMA for ID 8 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format G722 for ID 9 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format G729a for ID 18 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729a/8000... OK. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format G726-40 for ID 96 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format G726-24 for ID 97 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format G726-16 for ID 98 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Found audio description format telephone-event for ID 101 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Incorporating payload 0 on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Incorporating payload 2 on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Incorporating payload 8 on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Incorporating payload 9 on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Incorporating payload 18 on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Incorporating payload 97 on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Incorporating payload 98 on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Incorporating payload 101 on 0x7fb5b30b5f80 [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Capabilities: us - 0x8 (alaw), peer - audio=0x101d0c (ulaw|alaw|g726|g729|ilbc|g722|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2011-02-14 20:01:43.675] DEBUG[13628] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xf26e68' [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Peer audio RTP is at port 192.168.111.22:16454 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Copying payload 0 from 0x7fb5b30b5f80 to 0xf27030 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Copying payload 2 from 0x7fb5b30b5f80 to 0xf27030 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Copying payload 8 from 0x7fb5b30b5f80 to 0xf27030 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Copying payload 9 from 0x7fb5b30b5f80 to 0xf27030 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Copying payload 18 from 0x7fb5b30b5f80 to 0xf27030 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Copying payload 97 from 0x7fb5b30b5f80 to 0xf27030 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Copying payload 98 from 0x7fb5b30b5f80 to 0xf27030 [2011-02-14 20:01:43.675] DEBUG[13628] rtp_engine.c: Copying payload 101 from 0x7fb5b30b5f80 to 0xf27030 [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: We're settling with these formats: 0x8 (alaw) [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Checking SIP call limits for device zdenek [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Updating call counter for incoming call [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: Looking for *8 in filter-2 (domain 192.168.111.1) [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: *** Our native formats are 0x8 (alaw) [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: *** Our capabilities are 0x8 (alaw) [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: This channel will not be able to handle video. [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: build_route: Contact hop: [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: list_route: hop: [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: SIP/zdenek-00000002: New call is still down.... Trying... [2011-02-14 20:01:43.675] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.22:5063 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-3ce76872;received=192.168.111.22 From: ;tag=9eded97e1481156eo3 To: Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [2011-02-14 20:01:43.675] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.111.22:5063 [2011-02-14 20:01:43.676] DEBUG[13628] features.c: Call pickup on chan 'SIP/michal-00000001' by 'SIP/zdenek-00000002' [2011-02-14 20:01:43.676] DEBUG[13628] chan_sip.c: SIP answering channel: SIP/zdenek-00000002 [2011-02-14 20:01:43.676] DEBUG[13628] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-02-14 20:01:43.676] DEBUG[13628] chan_sip.c: Setting framing from config on incoming call [2011-02-14 20:01:43.676] DEBUG[13628] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [2011-02-14 20:01:43.676] DEBUG[13628] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2011-02-14 20:01:43.676] VERBOSE[13628] chan_sip.c: Audio is at 5060 [2011-02-14 20:01:43.676] VERBOSE[13628] chan_sip.c: Adding codec 0x8 (alaw) to SDP [2011-02-14 20:01:43.676] VERBOSE[13628] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-02-14 20:01:43.676] DEBUG[13628] chan_sip.c: -- Done with adding codecs to SDP [2011-02-14 20:01:43.676] DEBUG[13628] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [2011-02-14 20:01:43.676] VERBOSE[13628] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.111.22:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-3ce76872;received=192.168.111.22 From: ;tag=9eded97e1481156eo3 To: ;tag=as05e8dcb7 Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Jan Klepal" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 230 v=0 o=root 1659769998 1659769998 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 11228 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [2011-02-14 20:01:43.676] DEBUG[13628] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #52 [2011-02-14 20:01:43.676] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.22:5063 [2011-02-14 20:01:43.676] DEBUG[13619] devicestate.c: No provider found, checking channel drivers for SIP - zdenek [2011-02-14 20:01:43.676] DEBUG[13619] chan_sip.c: Checking device state for peer zdenek [2011-02-14 20:01:43.676] DEBUG[13619] devicestate.c: Changing state for SIP/zdenek - state 1 (Not in use) [2011-02-14 20:01:43.676] DEBUG[13619] devicestate.c: device 'SIP/zdenek' state '1' [2011-02-14 20:01:43.676] DEBUG[13619] devicestate.c: No provider found, checking channel drivers for SIP - zdenek [2011-02-14 20:01:43.676] DEBUG[13619] chan_sip.c: Checking device state for peer zdenek [2011-02-14 20:01:43.676] DEBUG[13619] devicestate.c: Changing state for SIP/zdenek - state 1 (Not in use) [2011-02-14 20:01:43.676] DEBUG[13619] devicestate.c: device 'SIP/zdenek' state '1' [2011-02-14 20:01:43.676] DEBUG[13635] app_queue.c: Device 'SIP/zdenek' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-02-14 20:01:43.676] DEBUG[13635] app_queue.c: Device 'SIP/zdenek' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-02-14 20:01:43.891] DEBUG[13628] channel.c: Planning to masquerade channel SIP/zdenek-00000002 into the structure of SIP/michal-00000001 [2011-02-14 20:01:43.891] DEBUG[13628] channel.c: Done planning to masquerade channel SIP/zdenek-00000002 into the structure of SIP/michal-00000001 [2011-02-14 20:01:43.891] DEBUG[13654] channel.c: Actually Masquerading SIP/zdenek-00000002(6) into the structure of SIP/michal-00000001(5) [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: SIP Fixup: New owner for dialogue 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060: SIP/zdenek-00000002 (Old parent: SIP/zdenek-00000002) [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: Hangup call SIP/zdenek-00000002, SIP callid 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: Hanging up channel in state Ringing (not UP) [2011-02-14 20:01:43.891] DEBUG[13654] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xf121d8' [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: Scheduling destruction of SIP dialog '0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060' in 6400 ms (Method: INVITE) [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060' Request 102: Found [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.17:5060: CANCEL sip:michal@192.168.111.17:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 Max-Forwards: 70 From: "Jan Klepal" ;tag=as19b4b5f0 To: Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 CSeq: 102 CANCEL User-Agent: Asterisk UCS Content-Length: 0 --- [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #55 [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 192.168.111.17:5060 [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: Scheduling destruction of SIP dialog '0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060' in 6400 ms (Method: INVITE) [2011-02-14 20:01:43.891] DEBUG[13654] channel.c: Putting channel SIP/zdenek-00000002 in alaw/alaw formats [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: SIP Fixup: New owner for dialogue cc43f6be-197c992e@192.168.111.22: SIP/zdenek-00000002 (Old parent: SIP/michal-00000001) [2011-02-14 20:01:43.891] DEBUG[13654] channel.c: Released clone lock on 'SIP/michal-00000001' [2011-02-14 20:01:43.891] DEBUG[13654] channel.c: Done Masquerading SIP/zdenek-00000002 (6) [2011-02-14 20:01:43.891] DEBUG[13654] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [2011-02-14 20:01:43.891] VERBOSE[13654] app_dial.c: -- SIP/zdenek-00000002 answered SIP/00260BD8E682_0-00000000 [2011-02-14 20:01:43.891] DEBUG[13628] channel.c: Hanging up zombie 'SIP/michal-00000001' [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: SIP answering channel: SIP/00260BD8E682_0-00000000 [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: SIP TIMER: Rescheduling retransmission #52 (1) SIP/2.0 - 1 [2011-02-14 20:01:43.891] DEBUG[13654] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #52)) [2011-02-14 20:01:43.891] DEBUG[13619] devicestate.c: No provider found, checking channel drivers for SIP - michal [2011-02-14 20:01:43.891] DEBUG[13619] chan_sip.c: Checking device state for peer michal [2011-02-14 20:01:43.891] VERBOSE[13628] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.111.22:5063: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-3ce76872;received=192.168.111.22 From: ;tag=9eded97e1481156eo3 To: ;tag=as05e8dcb7 Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Jan Klepal" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 230 v=0 o=root 1659769998 1659769998 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 11228 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.22:5063 [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: Setting framing from config on incoming call [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: Audio is at 5060 [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: Adding codec 0x8 (alaw) to SDP [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: -- Done with adding codecs to SDP [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [2011-02-14 20:01:43.891] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> ACK sip:*8@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-7beeb30c From: ;tag=9eded97e1481156eo3 To: ;tag=as05e8dcb7 Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="zdenek",realm="asterisk",nonce="6f49c32d",uri="sip:*8@192.168.111.1",algorithm=MD5,response="1941ebe568e55de0f12f6e6c375341ff" Contact: User-Agent: Cisco/SPA504G-7.4.3a Content-Length: 0 <-------------> [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 0 [ 37]: ACK sip:*8@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.111.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michal Kudlič" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 228 v=0 o=root 725020826 725020826 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 16170 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-7beeb30c [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 2 [ 55]: From: ;tag=9eded97e1481156eo3 [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 3 [ 41]: To: ;tag=as05e8dcb7 [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 4 [ 41]: Call-ID: cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #57 [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 5 [ 13]: CSeq: 102 ACK [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 7 [158]: Authorization: Digest username="zdenek",realm="asterisk",nonce="6f49c32d",uri="sip:*8@192.168.111.1",algorithm=MD5,response="1941ebe568e55de0f12f6e6c375341ff" [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 8 [ 41]: Contact: [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 9 [ 32]: User-Agent: Cisco/SPA504G-7.4.3a [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:01:43.891] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #52 [2011-02-14 20:01:43.891] DEBUG[13654] features.c: bridge answer set, chan answer set [2011-02-14 20:01:43.891] DEBUG[13628] chan_sip.c: Stopping retransmission on 'cc43f6be-197c992e@192.168.111.22' of Response 102: Match Found [2011-02-14 20:01:43.891] DEBUG[13654] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-02-14 20:01:43.891] DEBUG[13654] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-02-14 20:01:43.891] DEBUG[13619] devicestate.c: Changing state for SIP/michal - state 1 (Not in use) [2011-02-14 20:01:43.891] DEBUG[13619] devicestate.c: device 'SIP/michal' state '1' [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: Strict routing enforced for session 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:43.891] DEBUG[13619] devicestate.c: No provider found, checking channel drivers for SIP - 00260BD8E682_0 [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: set_destination: Parsing for address/port to send to [2011-02-14 20:01:43.891] DEBUG[13619] chan_sip.c: Checking device state for peer 00260BD8E682_0 [2011-02-14 20:01:43.891] DEBUG[13619] devicestate.c: Changing state for SIP/00260BD8E682_0 - state 1 (Not in use) [2011-02-14 20:01:43.891] DEBUG[13619] devicestate.c: device 'SIP/00260BD8E682_0' state '1' [2011-02-14 20:01:43.891] DEBUG[13635] app_queue.c: Device 'SIP/michal' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-02-14 20:01:43.891] DEBUG[13635] app_queue.c: Device 'SIP/00260BD8E682_0' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: set_destination: set destination to 192.168.111.15:5060 [2011-02-14 20:01:43.891] VERBOSE[13654] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.15:5060: UPDATE sip:777@192.168.111.15:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2965ec21 Max-Forwards: 70 From: ;tag=as697780e7 To: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 Contact: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 UPDATE User-Agent: Asterisk UCS Remote-Party-ID: "Zdeněk Brož" ;party=called;privacy=off;screen=no X-Asterisk-rpid-update: Yes Content-Length: 0 --- [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #58 [2011-02-14 20:01:43.891] DEBUG[13654] chan_sip.c: Trying to put 'UPDATE sip:' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:43.892] DEBUG[13654] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [2011-02-14 20:01:43.892] DEBUG[13654] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [2011-02-14 20:01:43.892] DEBUG[13654] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7fb5ae8ba358' [2011-02-14 20:01:43.892] DEBUG[13654] channel.c: Dropping duplicate answer! [2011-02-14 20:01:43.895] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.17:5060 ---> SIP/2.0 487 Request Terminated To: ;tag=541822245c9d4949i0 From: "Jan Klepal" ;tag=as19b4b5f0 Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 Server: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 <-------------> [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 0 [ 30]: SIP/2.0 487 Request Terminated [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 1 [ 59]: To: ;tag=541822245c9d4949i0 [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 2 [ 57]: From: "Jan Klepal" ;tag=as19b4b5f0 [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 3 [ 60]: Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 6 [ 34]: Server: Linksys/SPA3102-5.1.10(GW) [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Header 8 [ 0]: [2011-02-14 20:01:43.895] VERBOSE[13628] chan_sip.c: --- (8 headers 0 lines) --- [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Acked pending invite 102 [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Stopping retransmission on '0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060' of Request 102: Match Found [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: SIP response 487 to standard invite [2011-02-14 20:01:43.895] VERBOSE[13628] chan_sip.c: Transmitting (no NAT) to 192.168.111.17:5060: ACK sip:michal@192.168.111.17:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 Max-Forwards: 70 From: "Jan Klepal" ;tag=as19b4b5f0 To: ;tag=541822245c9d4949i0 Contact: Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 CSeq: 102 ACK User-Agent: Asterisk UCS Content-Length: 0 --- [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Trying to put 'ACK sip:mic' onto UDP socket destined for 192.168.111.17:5060 [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Updating call counter for outgoing call [2011-02-14 20:01:43.895] DEBUG[13628] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:43.897] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.17:5060 ---> SIP/2.0 200 OK To: ;tag=541822245c9d4949i0 From: "Jan Klepal" ;tag=as19b4b5f0 Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 CSeq: 102 CANCEL Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 Server: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 <-------------> [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 1 [ 59]: To: ;tag=541822245c9d4949i0 [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 2 [ 57]: From: "Jan Klepal" ;tag=as19b4b5f0 [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 3 [ 60]: Call-ID: 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 4 [ 16]: CSeq: 102 CANCEL [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2945c1e5 [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 6 [ 34]: Server: Linksys/SPA3102-5.1.10(GW) [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Header 8 [ 0]: [2011-02-14 20:01:43.897] VERBOSE[13628] chan_sip.c: --- (8 headers 0 lines) --- [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #55 [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Stopping retransmission on '0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060' of Request 102: Match Found [2011-02-14 20:01:43.897] DEBUG[13628] chan_sip.c: Destroying SIP dialog 0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060 [2011-02-14 20:01:43.897] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '0a178c0d78d33919795f54bf4b77a5db@192.168.111.1:5060' Method: INVITE [2011-02-14 20:01:43.897] DEBUG[13628] rtp_engine.c: Destroyed RTP instance '0xf121d8' [2011-02-14 20:01:43.904] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2965ec21 From: ;tag=as697780e7 To: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Date: Mon, 14 Feb 2011 19:01:43 GMT CSeq: 102 UPDATE Server: Cisco-CP7945G/8.5.3 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Content-Length: 0 <-------------> [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2965ec21 [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 2 [ 44]: From: ;tag=as697780e7 [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 3 [ 82]: To: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 5 [ 35]: Date: Mon, 14 Feb 2011 19:01:43 GMT [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 6 [ 16]: CSeq: 102 UPDATE [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 7 [ 27]: Server: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 8 [ 52]: Contact: [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 9 [ 75]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:01:43.904] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Acked pending invite 102 [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #58 [2011-02-14 20:01:43.904] DEBUG[13628] chan_sip.c: Stopping retransmission on '00260bd8-e682002b-93700064-08db6b13@192.168.111.15' of Request 102: Match Found [2011-02-14 20:01:43.904] VERBOSE[13628] chan_sip.c: SIP Response message for INCOMING dialog UPDATE arrived [2011-02-14 20:01:43.910] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> ACK sip:*8@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-7beeb30c From: ;tag=9eded97e1481156eo3 To: ;tag=as05e8dcb7 Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="zdenek",realm="asterisk",nonce="6f49c32d",uri="sip:*8@192.168.111.1",algorithm=MD5,response="1941ebe568e55de0f12f6e6c375341ff" Contact: User-Agent: Cisco/SPA504G-7.4.3a Content-Length: 0 <-------------> [2011-02-14 20:01:43.910] DEBUG[13628] chan_sip.c: Header 0 [ 37]: ACK sip:*8@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.111.22:5063;branch=z9hG4bK-7beeb30c [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 2 [ 55]: From: ;tag=9eded97e1481156eo3 [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 3 [ 41]: To: ;tag=as05e8dcb7 [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 4 [ 41]: Call-ID: cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 5 [ 13]: CSeq: 102 ACK [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 7 [158]: Authorization: Digest username="zdenek",realm="asterisk",nonce="6f49c32d",uri="sip:*8@192.168.111.1",algorithm=MD5,response="1941ebe568e55de0f12f6e6c375341ff" [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 8 [ 41]: Contact: [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 9 [ 32]: User-Agent: Cisco/SPA504G-7.4.3a [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:01:43.911] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:01:43.911] DEBUG[13628] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-02-14 20:01:43.985] DEBUG[13654] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [2011-02-14 20:01:43.985] DEBUG[13654] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [2011-02-14 20:01:43.991] DEBUG[13628] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (1) SIP/2.0 - 1 [2011-02-14 20:01:43.991] DEBUG[13628] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #57)) [2011-02-14 20:01:43.991] VERBOSE[13628] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.111.15:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michal Kudlič" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 228 v=0 o=root 725020826 725020826 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 16170 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-02-14 20:01:43.991] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:44.070] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> ACK sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK53a59f09 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:44 GMT CSeq: 102 ACK User-Agent: Cisco-CP7945G/8.5.3 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1",response="400da357a8d40649421bead6ae4a8588",nonce="3708fdc0",algorithm=MD5 Remote-Party-ID: "777" ;party=calling;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 0 [ 38]: ACK sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK53a59f09 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:44 GMT [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 9 [167]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1",response="400da357a8d40649421bead6ae4a8588",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 10 [113]: Remote-Party-ID: "777" ;party=calling;id-type=subscriber;privacy=off;screen=yes [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: Header 12 [ 0]: [2011-02-14 20:01:44.070] VERBOSE[13628] chan_sip.c: --- (12 headers 0 lines) --- [2011-02-14 20:01:44.070] DEBUG[13628] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-02-14 20:01:44.191] DEBUG[13628] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (2) SIP/2.0 - 1 [2011-02-14 20:01:44.191] DEBUG[13628] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #57)) [2011-02-14 20:01:44.191] VERBOSE[13628] chan_sip.c: Retransmitting #2 (no NAT) to 192.168.111.15:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michal Kudlič" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 228 v=0 o=root 725020826 725020826 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 16170 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-02-14 20:01:44.191] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:44.591] DEBUG[13628] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (3) SIP/2.0 - 1 [2011-02-14 20:01:44.592] DEBUG[13628] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #57)) [2011-02-14 20:01:44.592] VERBOSE[13628] chan_sip.c: Retransmitting #3 (no NAT) to 192.168.111.15:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michal Kudlič" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 228 v=0 o=root 725020826 725020826 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 16170 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-02-14 20:01:44.592] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:45.391] DEBUG[13628] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (4) SIP/2.0 - 1 [2011-02-14 20:01:45.391] DEBUG[13628] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #57)) [2011-02-14 20:01:45.391] VERBOSE[13628] chan_sip.c: Retransmitting #4 (no NAT) to 192.168.111.15:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michal Kudlič" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 228 v=0 o=root 725020826 725020826 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 16170 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-02-14 20:01:45.391] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:46.991] DEBUG[13628] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (5) SIP/2.0 - 1 [2011-02-14 20:01:46.991] DEBUG[13628] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #57)) [2011-02-14 20:01:46.991] VERBOSE[13628] chan_sip.c: Retransmitting #5 (no NAT) to 192.168.111.15:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michal Kudlič" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 228 v=0 o=root 725020826 725020826 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 16170 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-02-14 20:01:46.991] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:50.191] DEBUG[13628] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (6) SIP/2.0 - 1 [2011-02-14 20:01:50.191] DEBUG[13628] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #57)) [2011-02-14 20:01:50.191] VERBOSE[13628] chan_sip.c: Retransmitting #6 (no NAT) to 192.168.111.15:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK06e54fe8;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 102 INVITE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michal Kudlič" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 228 v=0 o=root 725020826 725020826 IN IP4 192.168.111.1 s=Asterisk UCS c=IN IP4 192.168.111.1 t=0 0 m=audio 16170 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-02-14 20:01:50.191] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.111.15:5060 [2011-02-14 20:01:50.292] WARNING[13628] chan_sip.c: Retransmission timeout reached on transmission 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [2011-02-14 20:01:50.292] WARNING[13628] chan_sip.c: Hanging up call 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 - no reply to our critical packet (see doc/sip-retransmit.txt). [2011-02-14 20:01:50.292] DEBUG[13628] chan_sip.c: Setting SIP_ALREADYGONE on dialog 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:50.292] DEBUG[13654] channel.c: Didn't get a frame from channel: SIP/00260BD8E682_0-00000000 [2011-02-14 20:01:50.292] DEBUG[13654] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-02-14 20:01:50.292] DEBUG[13654] channel.c: Bridge stops bridging channels SIP/00260BD8E682_0-00000000 and SIP/zdenek-00000002 [2011-02-14 20:01:50.292] DEBUG[13654] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is '2011-02-14 20:01:40' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is '"Jan Klepal" <777>' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is 'filter-2' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is 'SIP/00260BD8E682_0-00000000' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is 'SIP/zdenek-00000002' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is 'Dial' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is 'SIP/michal,,t' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is '10' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is '7' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is 'ANSWERED' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is 'DOCUMENTATION' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is '(null)' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is '1297710100.0' [2011-02-14 20:01:50.292] DEBUG[13654] pbx.c: Function result is '2,3,1,4,777,222,1,1,' [2011-02-14 20:01:50.292] DEBUG[13654] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield) VALUES ('2011-02-14 20:01:40','"Jan Klepal" <777>','filter-2','SIP/00260BD8E682_0-00000000','SIP/zdenek-00000002','Dial','SIP/michal,,t','10','7','ANSWERED','DOCUMENTATION','','1297710100.0','2,3,1,4,777,222,1,1,') [2011-02-14 20:01:50.405] DEBUG[13654] cdr_pgsql.c: inserting a CDR record. [2011-02-14 20:01:50.420] DEBUG[13654] channel.c: Hanging up channel 'SIP/zdenek-00000002' [2011-02-14 20:01:50.420] DEBUG[13654] chan_sip.c: Hangup call SIP/zdenek-00000002, SIP callid cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:50.420] DEBUG[13654] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xf26e68' [2011-02-14 20:01:50.420] VERBOSE[13654] chan_sip.c: Scheduling destruction of SIP dialog 'cc43f6be-197c992e@192.168.111.22' in 6400 ms (Method: ACK) [2011-02-14 20:01:50.420] DEBUG[13654] chan_sip.c: Strict routing enforced for session cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:50.420] VERBOSE[13654] chan_sip.c: set_destination: Parsing for address/port to send to [2011-02-14 20:01:50.420] VERBOSE[13654] chan_sip.c: set_destination: set destination to 192.168.111.22:5063 [2011-02-14 20:01:50.420] VERBOSE[13654] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.22:5063: BYE sip:zdenek@192.168.111.22:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK655c7488 Max-Forwards: 70 From: ;tag=as05e8dcb7 To: ;tag=9eded97e1481156eo3 Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 102 BYE User-Agent: Asterisk UCS Proxy-Authorization: Digest username="zdenek", realm="asterisk", algorithm=MD5, uri="192.168.111.1", nonce="", response="370a018a3764cc0101ecc8f5f2452a59" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [2011-02-14 20:01:50.420] DEBUG[13654] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #61 [2011-02-14 20:01:50.420] DEBUG[13654] chan_sip.c: Trying to put 'BYE sip:zde' onto UDP socket destined for 192.168.111.22:5063 [2011-02-14 20:01:50.421] DEBUG[13654] app_dial.c: Exiting with DIALSTATUS=ANSWER. [2011-02-14 20:01:50.421] DEBUG[13619] devicestate.c: No provider found, checking channel drivers for SIP - zdenek [2011-02-14 20:01:50.421] DEBUG[13619] chan_sip.c: Checking device state for peer zdenek [2011-02-14 20:01:50.421] DEBUG[13654] pbx.c: Spawn extension (filter-2,222,8) exited non-zero on 'SIP/00260BD8E682_0-00000000' [2011-02-14 20:01:50.421] DEBUG[13619] devicestate.c: Changing state for SIP/zdenek - state 1 (Not in use) [2011-02-14 20:01:50.421] DEBUG[13619] devicestate.c: device 'SIP/zdenek' state '1' [2011-02-14 20:01:50.421] VERBOSE[13654] pbx.c: == Spawn extension (filter-2, 222, 8) exited non-zero on 'SIP/00260BD8E682_0-00000000' [2011-02-14 20:01:50.421] DEBUG[13635] app_queue.c: Device 'SIP/zdenek' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-02-14 20:01:50.421] DEBUG[13654] channel.c: Soft-Hanging up channel 'SIP/00260BD8E682_0-00000000' [2011-02-14 20:01:50.421] DEBUG[13654] channel.c: Hanging up channel 'SIP/00260BD8E682_0-00000000' [2011-02-14 20:01:50.421] DEBUG[13654] chan_sip.c: Hangup call SIP/00260BD8E682_0-00000000, SIP callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:50.421] DEBUG[13654] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fb5ae8ba358' [2011-02-14 20:01:50.421] DEBUG[13619] devicestate.c: No provider found, checking channel drivers for SIP - 00260BD8E682_0 [2011-02-14 20:01:50.421] DEBUG[13619] chan_sip.c: Checking device state for peer 00260BD8E682_0 [2011-02-14 20:01:50.421] DEBUG[13619] devicestate.c: Changing state for SIP/00260BD8E682_0 - state 1 (Not in use) [2011-02-14 20:01:50.421] DEBUG[13619] devicestate.c: device 'SIP/00260BD8E682_0' state '1' [2011-02-14 20:01:50.421] DEBUG[13635] app_queue.c: Device 'SIP/00260BD8E682_0' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-02-14 20:01:50.427] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> SIP/2.0 200 OK To: ;tag=9eded97e1481156eo3 From: ;tag=as05e8dcb7 Call-ID: cc43f6be-197c992e@192.168.111.22 CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK655c7488 Server: Cisco/SPA504G-7.4.3a Content-Length: 0 <-------------> [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 1 [ 53]: To: ;tag=9eded97e1481156eo3 [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 2 [ 43]: From: ;tag=as05e8dcb7 [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 3 [ 41]: Call-ID: cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 4 [ 13]: CSeq: 102 BYE [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK655c7488 [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 6 [ 28]: Server: Cisco/SPA504G-7.4.3a [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Header 8 [ 0]: [2011-02-14 20:01:50.427] VERBOSE[13628] chan_sip.c: --- (8 headers 0 lines) --- [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #61 [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Stopping retransmission on 'cc43f6be-197c992e@192.168.111.22' of Request 102: Match Found [2011-02-14 20:01:50.427] VERBOSE[13628] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Destroying SIP dialog 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:50.427] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '00260bd8-e682002b-93700064-08db6b13@192.168.111.15' Method: INVITE [2011-02-14 20:01:50.427] DEBUG[13628] rtp_engine.c: Destroyed RTP instance '0x7fb5ae8ba358' [2011-02-14 20:01:50.427] DEBUG[13628] chan_sip.c: Destroying SIP dialog cc43f6be-197c992e@192.168.111.22 [2011-02-14 20:01:50.427] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog 'cc43f6be-197c992e@192.168.111.22' Method: ACK [2011-02-14 20:01:50.427] DEBUG[13628] rtp_engine.c: Destroyed RTP instance '0xf26e68' [2011-02-14 20:01:51.836] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 0 [ 38]: BYE sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:51 GMT [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 103 BYE [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 10 [172]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:01:51.836] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:01:51.836] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:01:51.837] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:01:51.837] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UNKNOWN with address 192.168.111.1:5060 [2011-02-14 20:01:51.837] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:01:51.837] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.111.15:51177 [2011-02-14 20:01:51.837] DEBUG[13628] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:51.837] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 590 [2011-02-14 20:01:52.311] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 0 [ 38]: BYE sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:51 GMT [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 103 BYE [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 10 [172]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:01:52.311] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:01:52.311] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:52.311] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.111.15:51177 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:52.311] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 590 [2011-02-14 20:01:53.301] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 0 [ 38]: BYE sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:51 GMT [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 103 BYE [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 10 [172]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:01:53.301] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:01:53.301] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:53.301] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.111.15:51177 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:53.301] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 590 [2011-02-14 20:01:55.311] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 0 [ 38]: BYE sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:51 GMT [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 103 BYE [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 10 [172]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:01:55.311] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:01:55.311] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:55.311] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.111.15:51177 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:55.311] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 590 [2011-02-14 20:01:59.301] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 0 [ 38]: BYE sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:51 GMT [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 103 BYE [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 10 [172]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:01:59.301] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:01:59.301] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:01:59.301] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.111.15:51177 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:01:59.301] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 590 [2011-02-14 20:02:03.321] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 0 [ 38]: BYE sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:51 GMT [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 103 BYE [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 10 [172]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:02:03.321] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:02:03.321] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:02:03.321] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.111.15:51177 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:02:03.321] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 590 [2011-02-14 20:02:07.311] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 0 [ 38]: BYE sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:51 GMT [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 103 BYE [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 10 [172]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:02:07.311] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:02:07.311] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:02:07.311] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.111.15:51177 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:02:07.311] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 590 [2011-02-14 20:02:11.301] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 0 [ 38]: BYE sip:222@192.168.111.1:5060 SIP/2.0 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 2 [ 84]: From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 3 [ 42]: To: ;tag=as697780e7 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 4 [ 59]: Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 6 [ 35]: Date: Mon, 14 Feb 2011 19:01:51 GMT [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 7 [ 13]: CSeq: 103 BYE [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 8 [ 31]: User-Agent: Cisco-CP7945G/8.5.3 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 10 [172]: Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Header 11 [ 0]: [2011-02-14 20:02:11.301] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:02:11.301] DEBUG[13628] acl.c: For destination '192.168.111.15', our source address is '192.168.111.1'. [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.111.1:5060 [2011-02-14 20:02:11.301] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.111.15:51177 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 [2011-02-14 20:02:11.301] DEBUG[13628] chan_sip.c: Invalid SIP message - rejected , no callid, len 590 [2011-02-14 20:02:15.321] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:02:15.321] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:02:15.321] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:02:19.311] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:02:19.311] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:02:19.311] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:02:23.301] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51177 ---> BYE sip:222@192.168.111.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 Max-Forwards: 70 Date: Mon, 14 Feb 2011 19:01:51 GMT CSeq: 103 BYE User-Agent: Cisco-CP7945G/8.5.3 Content-Length: 0 Authorization: Digest username="00260BD8E682_0",realm="asterisk",uri="sip:222@192.168.111.1:5060",response="ed18d516bdf700399381cfb650236fa1",nonce="3708fdc0",algorithm=MD5 <-------------> [2011-02-14 20:02:23.301] VERBOSE[13628] chan_sip.c: --- (11 headers 0 lines) --- [2011-02-14 20:02:23.301] VERBOSE[13628] chan_sip.c: <--- Transmitting (no NAT) to 192.168.111.15:51177 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.111.15:5060;branch=z9hG4bK8d219706;received=192.168.111.15 From: "777" ;tag=00260bd8e68208bf4bfe6e0a-48e843f9 To: ;tag=as697780e7 Call-ID: 00260bd8-e682002b-93700064-08db6b13@192.168.111.15 CSeq: 103 BYE Server: Asterisk UCS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-02-14 20:02:36.546] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.22:5063: OPTIONS sip:zdenek@192.168.111.22:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK12474b3d Max-Forwards: 70 From: "asterisk" ;tag=as7b1e8d3e To: Contact: Call-ID: 20e10e50515f99df79a6fd0553fa9274@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:02:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:02:36.549] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.17:5060: OPTIONS sip:michal@192.168.111.17:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK3f1b3ad6 Max-Forwards: 70 From: "asterisk" ;tag=as3ae7f020 To: Contact: Call-ID: 33f0ed165be6bf9b14d2faf74c2f0e91@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:02:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:02:36.555] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.17:5060 ---> SIP/2.0 200 OK To: ;tag=bc0c28782f1665a0i0 From: "asterisk" ;tag=as3ae7f020 Call-ID: 33f0ed165be6bf9b14d2faf74c2f0e91@192.168.111.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK3f1b3ad6 Server: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2011-02-14 20:02:36.555] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:02:36.555] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '33f0ed165be6bf9b14d2faf74c2f0e91@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:02:36.557] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> SIP/2.0 200 OK To: ;tag=9d9812c22b8579e0i3 From: "asterisk" ;tag=as7b1e8d3e Call-ID: 20e10e50515f99df79a6fd0553fa9274@192.168.111.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK12474b3d Server: Cisco/SPA504G-7.4.3a Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces <-------------> [2011-02-14 20:02:36.557] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:02:36.557] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '20e10e50515f99df79a6fd0553fa9274@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:02:36.562] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.15:5060: OPTIONS sip:777@192.168.111.15:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK64edf1c8 Max-Forwards: 70 From: "asterisk" ;tag=as350b7afb To: Contact: Call-ID: 67ae32170ae915ff093d45d52b88c9af@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:02:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:02:36.593] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51684 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK64edf1c8 From: "asterisk" ;tag=as350b7afb To: ;tag=00260bd8e68208c05baf044b-f8f5a488 Call-ID: 67ae32170ae915ff093d45d52b88c9af@192.168.111.1:5060 Date: Mon, 14 Feb 2011 19:02:36 GMT CSeq: 102 OPTIONS Server: Cisco-CP7945G/8.5.3 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces,join,norefersub Content-Length: 287 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 17788 0 IN IP4 192.168.111.15 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 116 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2011-02-14 20:02:36.593] VERBOSE[13628] chan_sip.c: --- (17 headers 13 lines) --- [2011-02-14 20:02:36.593] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '67ae32170ae915ff093d45d52b88c9af@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:03:36.555] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.17:5060: OPTIONS sip:michal@192.168.111.17:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2fa306e5 Max-Forwards: 70 From: "asterisk" ;tag=as6a0eae64 To: Contact: Call-ID: 15bc6c7329741e18116452f429b14513@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:03:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:03:36.557] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.22:5063: OPTIONS sip:zdenek@192.168.111.22:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK27568fc3 Max-Forwards: 70 From: "asterisk" ;tag=as1cfb1781 To: Contact: Call-ID: 2a202364306485cd0ea3939f604f6f21@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:03:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:03:36.564] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.17:5060 ---> SIP/2.0 200 OK To: ;tag=bc0c28782f1665a0i0 From: "asterisk" ;tag=as6a0eae64 Call-ID: 15bc6c7329741e18116452f429b14513@192.168.111.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK2fa306e5 Server: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2011-02-14 20:03:36.564] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:03:36.564] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '15bc6c7329741e18116452f429b14513@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:03:36.567] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.22:5063 ---> SIP/2.0 200 OK To: ;tag=9d9812c22b8579e0i3 From: "asterisk" ;tag=as1cfb1781 Call-ID: 2a202364306485cd0ea3939f604f6f21@192.168.111.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK27568fc3 Server: Cisco/SPA504G-7.4.3a Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces <-------------> [2011-02-14 20:03:36.567] VERBOSE[13628] chan_sip.c: --- (10 headers 0 lines) --- [2011-02-14 20:03:36.567] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '2a202364306485cd0ea3939f604f6f21@192.168.111.1:5060' Method: OPTIONS [2011-02-14 20:03:36.593] VERBOSE[13628] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.111.15:5060: OPTIONS sip:777@192.168.111.15:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK3255d418 Max-Forwards: 70 From: "asterisk" ;tag=as1a9e4c89 To: Contact: Call-ID: 776045a7090de4f54c73380671d6fcf0@192.168.111.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk UCS Date: Mon, 14 Feb 2011 19:03:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-02-14 20:03:36.622] VERBOSE[13628] chan_sip.c: <--- SIP read from UDP:192.168.111.15:51685 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.111.1:5060;branch=z9hG4bK3255d418 From: "asterisk" ;tag=as1a9e4c89 To: ;tag=00260bd8e68208c1356b5bd6-95a16441 Call-ID: 776045a7090de4f54c73380671d6fcf0@192.168.111.1:5060 Date: Mon, 14 Feb 2011 19:03:36 GMT CSeq: 102 OPTIONS Server: Cisco-CP7945G/8.5.3 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces,join,norefersub Content-Length: 287 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 20198 0 IN IP4 192.168.111.15 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 116 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2011-02-14 20:03:36.622] VERBOSE[13628] chan_sip.c: --- (17 headers 13 lines) --- [2011-02-14 20:03:36.622] VERBOSE[13628] chan_sip.c: Really destroying SIP dialog '776045a7090de4f54c73380671d6fcf0@192.168.111.1:5060' Method: OPTIONS