[2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981086 (1) REGISTER - 2 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #5981086)) [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 0 [ 52]: INVITE sip:5321@192.168.15.251:5060;user=phone SIP/2.0 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bKc4341f7FFFA3D1E [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 2 [ 57]: From: "WT1" ;tag=E3B34B85-105452F2 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 3 [ 38]: To: [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 10 [ 26]: Supported: 100rel,replaces [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 11 [ 34]: Allow-Events: talk,hold,conference [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 14 [ 19]: Content-Length: 270 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Header 15 [ 0]: [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 1 [ 45]: o=- 1320791388 1320791388 IN IP4 192.168.15.184 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.184 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 5 [ 10]: a=sendrecv [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 6 [ 31]: m=audio 2230 RTP/AVP 0 8 18 101 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 10 [ 19]: a=fmtp:18 annexb=no [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:06] DEBUG[2629] acl.c: For destination '192.168.15.184', our source address is '192.168.15.251'. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.251:5060 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for bbc7e461-d5bca2bb-bed7a630@192.168.15.184 - INVITE (No RTP) [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-11-08 17:30:06] DEBUG[2629] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,replaces" [2011-11-08 17:30:06] DEBUG[2629] sip/reqresp_parser.c: Found SIP option: -100rel- [2011-11-08 17:30:06] DEBUG[2629] sip/reqresp_parser.c: Matched SIP option: 100rel [2011-11-08 17:30:06] DEBUG[2629] sip/reqresp_parser.c: Found SIP option: -replaces- [2011-11-08 17:30:06] DEBUG[2629] sip/reqresp_parser.c: Matched SIP option: replaces [2011-11-08 17:30:06] DEBUG[2629] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:06] DEBUG[2629] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Initializing initreq for method INVITE - callid bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2aaacc025808' [2011-11-08 17:30:06] DEBUG[2629] res_rtp_asterisk.c: Allocated port 11254 for RTP instance '0x2aaacc025808' [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: RTP instance '0x2aaacc025808' is setup and ready to go [2011-11-08 17:30:06] DEBUG[2629] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x2aaacc025808' [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Setting NAT on RTP to On [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing session-level SDP o=- 1320791388 1320791388 IN IP4 192.168.15.184... UNSUPPORTED. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [2011-11-08 17:30:06] DEBUG[2629] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:06] DEBUG[2629] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.15.184... OK. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Setting payload 0 based on m type on 0x40299570 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Setting payload 8 based on m type on 0x40299570 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Setting payload 18 based on m type on 0x40299570 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Setting payload 101 based on m type on 0x40299570 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Incorporating payload 0 on 0x40299570 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Incorporating payload 8 on 0x40299570 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Incorporating payload 18 on 0x40299570 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Incorporating payload 101 on 0x40299570 [2011-11-08 17:30:06] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2aaacc025808' [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Copying payload 0 from 0x40299570 to 0x2aaacc0259d0 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Copying payload 8 from 0x40299570 to 0x2aaacc0259d0 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Copying payload 18 from 0x40299570 to 0x2aaacc0259d0 [2011-11-08 17:30:06] DEBUG[2629] rtp_engine.c: Copying payload 101 from 0x40299570 to 0x2aaacc0259d0 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Checking SIP call limits for device 5261 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Updating call counter for incoming call [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Call from peer '5261' is 1 out of 100 [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: *** Our native formats are 0x4 (ulaw) [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/5261-0001cd99 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 5261 CallerIDName: WT1 AccountCode: Exten: 5321 Context: DLPN_LUSI_Internet_Unrestricted Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:06] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 2 (In use) [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: device 'SIP/5261' state '2' [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5261 Context: default Hint: SIP/5261 Status: 1 [2011-11-08 17:30:06] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: *** Our capabilities are 0x80e (gsm|ulaw|alaw|g726) [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: This channel will not be able to handle video. [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: build_route: Contact hop: [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: SIP/5261-0001cd99: New call is still down.... Trying... [2011-11-08 17:30:06] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:06] DEBUG[2500] app_queue.c: Extension '5261@default' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:06] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: SIPURI Value: sip:5261@192.168.15.184 Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: SIPDOMAIN Value: 192.168.15.251:5060 Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[24823] pbx.c: Result of 'HINT' is 'SIP/5321' [2011-11-08 17:30:06] DEBUG[24823] pbx.c: Launching 'Dial' [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: SIPCALLID Value: bbc7e461-d5bca2bb-bed7a630@192.168.15.184 Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 2 (In use) [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5261-0001cd99 Uniqueid: 1320791406.126035 Channeltype: SIP SIPcallid: bbc7e461-d5bca2bb-bed7a630@192.168.15.184 SIPfullcontact: sip:5261@192.168.15.184 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5261-0001cd99 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 5261 CallerIDName: WT1 ConnectedLineNum: ConnectedLineName: Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: device 'SIP/5261' state '2' [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd99 Context: DLPN_LUSI_Internet_Unrestricted Extension: 5321 Priority: 1 Application: Dial AppData: SIP/5321 Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Allocating new SIP dialog for 691ce2166673f3a622e299b7485606a8@192.168.7.11:0 - INVITE (No RTP) [2011-11-08 17:30:06] DEBUG[24823] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x20571658' [2011-11-08 17:30:06] DEBUG[24823] res_rtp_asterisk.c: Allocated port 14242 for RTP instance '0x20571658' [2011-11-08 17:30:06] DEBUG[24823] rtp_engine.c: RTP instance '0x20571658' is setup and ready to go [2011-11-08 17:30:06] DEBUG[24823] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x206a7f28' [2011-11-08 17:30:06] DEBUG[24823] res_rtp_asterisk.c: Allocated port 12066 for RTP instance '0x206a7f28' [2011-11-08 17:30:06] DEBUG[24823] rtp_engine.c: RTP instance '0x206a7f28' is setup and ready to go [2011-11-08 17:30:06] DEBUG[24823] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x206a7f28' [2011-11-08 17:30:06] DEBUG[24823] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x20571658' [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Setting NAT on RTP to On [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Setting NAT on VRTP to On [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2011-11-08 17:30:06] DEBUG[24823] acl.c: For destination '192.168.15.187', our source address is '192.168.15.251'. [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.251:5060 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: *** Our native formats are 0x4 (ulaw) [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: *** Our capabilities are 0x380006 (gsm|ulaw|h263|h263p|h264) [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: This channel can handle video! HOLLYWOOD next! [2011-11-08 17:30:06] DEBUG[24823] rtp_engine.c: Seeded SDP of 'SIP/5321-0001cd9a' with that of 'SIP/5261-0001cd99' [2011-11-08 17:30:06] DEBUG[24823] channel.c: Not copying variable DIALEDTIME. [2011-11-08 17:30:06] DEBUG[24823] channel.c: Not copying variable ANSWEREDTIME. [2011-11-08 17:30:06] DEBUG[24823] channel.c: Not copying variable DIALEDPEERNAME. [2011-11-08 17:30:06] DEBUG[24823] channel.c: Not copying variable DIALEDPEERNUMBER. [2011-11-08 17:30:06] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: DIALSTATUS Value: Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: DIALEDPEERNUMBER Value: Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: DIALEDPEERNAME Value: Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: ANSWEREDTIME Value: Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: DIALEDTIME Value: Uniqueid: 1320791406.126035 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/5321-0001cd9a ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 5321 CallerIDName: User1 AccountCode: Exten: Context: DLPN_DialPlan1 Uniqueid: 1320791406.126036 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9a Variable: SIPCALLID Value: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 Uniqueid: 1320791406.126036 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5321-0001cd9a Uniqueid: 1320791406.126036 Channeltype: SIP SIPcallid: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 SIPfullcontact: sip:5321@192.168.15.187 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5321-0001cd9a Channeltype: SIP SIPcallid: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 SIPfullcontact: sip:5321@192.168.15.187 Peername: 5321 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9a Variable: DIALEDPEERNUMBER Value: 5321 Uniqueid: 1320791406.126036 [2011-11-08 17:30:06] DEBUG[24823] channel.c: Not copying variable DIALSTATUS. [2011-11-08 17:30:06] DEBUG[24823] channel.c: Not copying variable SIPCALLID. [2011-11-08 17:30:06] DEBUG[24823] channel.c: Not copying variable SIPDOMAIN. [2011-11-08 17:30:06] DEBUG[24823] channel.c: Not copying variable SIPURI. [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Outgoing Call for 5321 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Call to peer '5321' is 1 out of 100 [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:06] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 6 (Ringing) [2011-11-08 17:30:06] DEBUG[2499] devicestate.c: device 'SIP/5321' state '6' [2011-11-08 17:30:06] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '6' (Ringing) [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 6 Paused: 0 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: This call needs video offers! [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: ** Our capability: 0x380006 (gsm|ulaw|h263|h263p|h264) Video flag: False Text flag: False [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: -- Done with adding codecs to SDP [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Done building SDP. Settling with this capability: 0x380006 (gsm|ulaw|h263|h263p|h264) [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Initializing initreq for method INVITE - callid 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 0 [ 36]: INVITE sip:5321@192.168.15.187 SIP/2.0 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.15.251:5060;branch=z9hG4bK7202bbc3;rport [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 4 [ 27]: To: [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 5 [ 37]: Contact: [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 6 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 9 [ 35]: Date: Tue, 08 Nov 2011 22:30:06 GMT [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 12 [ 82]: Remote-Party-ID: "WT1" ;party=calling;privacy=off;screen=no [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981088 [2011-11-08 17:30:06] DEBUG[24823] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/5261-0001cd99 Destination: SIP/5321-0001cd9a CallerIDNum: 5261 CallerIDName: WT1 ConnectedLineNum: ConnectedLineName: UniqueID: 1320791406.126035 DestUniqueID: 1320791406.126036 Dialstring: 5321 [2011-11-08 17:30:06] DEBUG[2500] app_queue.c: Extension '5321@default' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [2011-11-08 17:30:06] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5321 Context: default Hint: SIP/5321 Status: 8 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.15.251:5060;branch=z9hG4bK7202bbc3;rport [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 2 [ 50]: From: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 3 [ 49]: To: ;tag=5455E8AA-EF355777 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 5 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 10 [ 0]: [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5981088 - INVITE (got response) [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' Request 102: Found [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: SIP response 100 to standard invite [2011-11-08 17:30:07] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: ~HASH~SIP_CAUSE~SIP/5321-0001cd9a~ Value: SIP 100 Trying Uniqueid: 1320791406.126035 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981086 (2) REGISTER - 2 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #5981086)) [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.15.251:5060;branch=z9hG4bK7202bbc3;rport [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 2 [ 50]: From: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 3 [ 49]: To: ;tag=5455E8AA-EF355777 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 5 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 8 [ 34]: Allow-Events: talk,hold,conference [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' Request 102: Found [2011-11-08 17:30:07] DEBUG[2629] chan_sip.c: SIP response 180 to standard invite [2011-11-08 17:30:07] DEBUG[24823] rtp_engine.c: Setting early bridge SDP of 'SIP/5261-0001cd99' with that of 'SIP/5321-0001cd9a' [2011-11-08 17:30:07] DEBUG[24823] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:07] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:07] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:07] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 6 (Ringing) [2011-11-08 17:30:07] DEBUG[2499] devicestate.c: device 'SIP/5321' state '6' [2011-11-08 17:30:07] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5321-0001cd9a ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 5321 CallerIDName: User1 ConnectedLineNum: 5261 ConnectedLineName: WT1 Uniqueid: 1320791406.126036 [2011-11-08 17:30:07] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: ~HASH~SIP_CAUSE~SIP/5321-0001cd9a~ Value: SIP 180 Ringing Uniqueid: 1320791406.126035 [2011-11-08 17:30:07] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '6' (Ringing) [2011-11-08 17:30:07] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 6 Paused: 0 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981086 (3) REGISTER - 2 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #5981086)) [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 0 [ 38]: REGISTER sip:192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.149;branch=z9hG4bKe297faf3B4E7EEB8 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 2 [ 67]: From: "Keith Jackson" ;tag=8E06C3BA-6B94B80D [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 3 [ 27]: To: [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 4 [ 19]: CSeq: 5157 REGISTER [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: 98cce35e-c38be244-dc5be10f@192.168.15.149 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 6 [132]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 9 [179]: Authorization: Digest username="5223", realm="asterisk-phone.lusi.on.ca", nonce="279de0ea", uri="sip:192.168.15.251:5060", response="d054194690529044582a32dba6ade25e", algorithm=MD5 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 11 [ 12]: Expires: 300 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 13 [ 0]: [2011-11-08 17:30:09] DEBUG[2629] acl.c: For destination '192.168.15.149', our source address is '192.168.15.251'. [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.251:5060 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for 98cce35e-c38be244-dc5be10f@192.168.15.149 - REGISTER (No RTP) [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Initializing initreq for method REGISTER - callid 98cce35e-c38be244-dc5be10f@192.168.15.149 [2011-11-08 17:30:09] DEBUG[2629] netsock2.c: Splitting '192.168.15.149' into... [2011-11-08 17:30:09] DEBUG[2629] netsock2.c: ...host '192.168.15.149' and port ''. [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.15.149:5060 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 0 [ 38]: REGISTER sip:192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.149;branch=z9hG4bK3f0eed51E9C0E10E [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 2 [ 67]: From: "Keith Jackson" ;tag=8E06C3BA-6B94B80D [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 3 [ 27]: To: [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 4 [ 19]: CSeq: 5158 REGISTER [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: 98cce35e-c38be244-dc5be10f@192.168.15.149 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 6 [132]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 9 [179]: Authorization: Digest username="5223", realm="asterisk-phone.lusi.on.ca", nonce="55995194", uri="sip:192.168.15.251:5060", response="f69752b8e9f79c6b4dfea7284128202e", algorithm=MD5 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 11 [ 12]: Expires: 300 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Header 13 [ 0]: [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Initializing initreq for method REGISTER - callid 98cce35e-c38be244-dc5be10f@192.168.15.149 [2011-11-08 17:30:09] DEBUG[2629] netsock2.c: Splitting '192.168.15.149' into... [2011-11-08 17:30:09] DEBUG[2629] netsock2.c: ...host '192.168.15.149' and port ''. [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Store REGISTER's src-IP:port for call routing. [2011-11-08 17:30:09] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.149:5060 [2011-11-08 17:30:09] DEBUG[14661] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/5223 PeerStatus: Registered Address: 192.168.15.149:5060 [2011-11-08 17:30:09] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5223 [2011-11-08 17:30:09] DEBUG[2499] chan_sip.c: Checking device state for peer 5223 [2011-11-08 17:30:09] DEBUG[2499] devicestate.c: Changing state for SIP/5223 - state 1 (Not in use) [2011-11-08 17:30:09] DEBUG[2499] devicestate.c: device 'SIP/5223' state '1' [2011-11-08 17:30:09] DEBUG[2756] app_queue.c: Device 'SIP/5223' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: Auto destroying SIP dialog '2fae34f245c9966f50972c297f68efc0@192.168.7.11' [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: Destroying SIP dialog 2fae34f245c9966f50972c297f68efc0@192.168.7.11 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: ---------- SIP HISTORY for '2fae34f245c9966f50972c297f68efc0@192.168.7.11' [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: * SIP Call [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 001. AuthResp Auth response sent for 126982 in realm toronto2.voip.ms - nc 3 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 002. ReTx 1000 REGISTER sip:toronto2.voip.ms SIP/2.0 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 003. ReTx 2000 REGISTER sip:toronto2.voip.ms SIP/2.0 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 004. ReTx 4000 REGISTER sip:toronto2.voip.ms SIP/2.0 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 005. ReTx 4000 REGISTER sip:toronto2.voip.ms SIP/2.0 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 006. ReTx 4000 REGISTER sip:toronto2.voip.ms SIP/2.0 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 007. Ignore Ignoring this retransmit [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 008. AuthResp Auth response sent for 126982 in realm toronto2.voip.ms - nc 1 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 009. Ignore Ignoring this retransmit [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: 010. AutoDestroy 2fae34f245c9966f50972c297f68efc0@192.168.7.11 [2011-11-08 17:30:10] DEBUG[2629] chan_sip.c: ---------- END SIP HISTORY for '2fae34f245c9966f50972c297f68efc0@192.168.7.11' [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.15.251:5060;branch=z9hG4bK7202bbc3;rport [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 2 [ 50]: From: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 3 [ 49]: To: ;tag=5455E8AA-EF355777 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 5 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 11 [ 19]: Content-Length: 316 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 12 [ 0]: [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 1 [ 45]: o=- 1320791148 1320791148 IN IP4 192.168.15.187 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.187 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 5 [ 26]: m=audio 2236 RTP/AVP 0 101 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 6 [ 10]: a=sendrecv [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 9 [ 26]: m=video 0 RTP/AVP 34 98 99 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 10 [ 10]: a=sendrecv [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 11 [ 22]: a=rtpmap:34 H263/90000 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 12 [ 27]: a=rtpmap:98 h263-1998/90000 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Body 13 [ 22]: a=rtpmap:99 H264/90000 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Acked pending invite 102 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Stopping retransmission on '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' of Request 102: Match Found [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: SIP response 200 to standard invite [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Processing session-level SDP o=- 1320791148 1320791148 IN IP4 192.168.15.187... UNSUPPORTED. [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [2011-11-08 17:30:11] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:11] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.15.187... OK. [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-11-08 17:30:11] DEBUG[2629] rtp_engine.c: Setting payload 0 based on m type on 0x40298c30 [2011-11-08 17:30:11] DEBUG[2629] rtp_engine.c: Setting payload 101 based on m type on 0x40298c30 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-11-08 17:30:11] DEBUG[2629] rtp_engine.c: Incorporating payload 0 on 0x40298c30 [2011-11-08 17:30:11] DEBUG[2629] rtp_engine.c: Incorporating payload 101 on 0x40298c30 [2011-11-08 17:30:11] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x20571658' [2011-11-08 17:30:11] DEBUG[2629] rtp_engine.c: Copying payload 0 from 0x40298c30 to 0x20571820 [2011-11-08 17:30:11] DEBUG[2629] rtp_engine.c: Copying payload 101 from 0x40298c30 to 0x20571820 [2011-11-08 17:30:11] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x206a7f28' [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: We have an owner, now see if we need to change this call [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: build_route: Contact hop: [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Strict routing enforced for session 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:11] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:11] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Trying to put 'ACK sip:532' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5321-0001cd9a Channeltype: SIP Uniqueid: 1320791406.126036 SIPcallid: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 SIPfullcontact: sip:5321@192.168.15.187 Peername: 5321 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: ~HASH~SIP_CAUSE~SIP/5321-0001cd9a~ Value: SIP 200 OK Uniqueid: 1320791406.126035 [2011-11-08 17:30:11] DEBUG[24823] rtp_engine.c: Setting early bridge SDP of 'SIP/5261-0001cd99' with that of 'SIP/5321-0001cd9a' [2011-11-08 17:30:11] DEBUG[24823] chan_sip.c: SIP answering channel: SIP/5261-0001cd99 [2011-11-08 17:30:11] DEBUG[24823] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:11] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 2 (In use) [2011-11-08 17:30:11] DEBUG[24823] chan_sip.c: Setting framing from config on incoming call [2011-11-08 17:30:11] DEBUG[24823] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [2011-11-08 17:30:11] DEBUG[24823] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2011-11-08 17:30:11] DEBUG[24823] chan_sip.c: -- Done with adding codecs to SDP [2011-11-08 17:30:11] DEBUG[24823] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: device 'SIP/5321' state '2' [2011-11-08 17:30:11] DEBUG[24823] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981094 [2011-11-08 17:30:11] DEBUG[24823] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5321-0001cd9a ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 5321 CallerIDName: User1 ConnectedLineNum: 5261 ConnectedLineName: WT1 Uniqueid: 1320791406.126036 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1320791406.126035 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: DIALEDPEERNAME Value: SIP/5321-0001cd9a Uniqueid: 1320791406.126035 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: DIALEDPEERNUMBER Value: 5321 Uniqueid: 1320791406.126035 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: BRIDGEPEER Value: SIP/5321-0001cd9a Uniqueid: 1320791406.126035 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9a Variable: BRIDGEPEER Value: SIP/5261-0001cd99 Uniqueid: 1320791406.126036 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5261-0001cd99 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 5261 CallerIDName: WT1 ConnectedLineNum: 5321 ConnectedLineName: User1 Uniqueid: 1320791406.126035 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/5321-0001cd9a Uniqueid: 1320791406.126036 AccountCode: OldAccountCode: [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 2 Paused: 0 [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:11] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 2 (In use) [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: device 'SIP/5321' state '2' [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:11] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 2 (In use) [2011-11-08 17:30:11] DEBUG[2499] devicestate.c: device 'SIP/5261' state '2' [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5321 Context: default Hint: SIP/5321 Status: 1 [2011-11-08 17:30:11] DEBUG[24823] features.c: bridge answer set, chan answer set [2011-11-08 17:30:11] DEBUG[24823] features.c: Removing dialed interfaces datastore on SIP/5321-0001cd9a since we're bridging [2011-11-08 17:30:11] DEBUG[24823] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:11] DEBUG[24823] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/5261-0001cd99 Channel2: SIP/5321-0001cd9a Uniqueid1: 1320791406.126035 Uniqueid2: 1320791406.126036 CallerID1: 5261 CallerID2: 5321 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: BRIDGEPEER Value: SIP/5321-0001cd9a Uniqueid: 1320791406.126035 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: BRIDGEPVTCALLID Value: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 Uniqueid: 1320791406.126035 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9a Variable: BRIDGEPEER Value: SIP/5261-0001cd99 Uniqueid: 1320791406.126036 [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9a Variable: BRIDGEPVTCALLID Value: bbc7e461-d5bca2bb-bed7a630@192.168.15.184 Uniqueid: 1320791406.126036 [2011-11-08 17:30:11] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '2' (In use) [2011-11-08 17:30:11] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '2' (In use) [2011-11-08 17:30:11] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:11] DEBUG[2500] app_queue.c: Extension '5321@default' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:11] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 2 Paused: 0 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 0 [ 38]: ACK sip:5321@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bK40028be4D7C5893F [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 2 [ 57]: From: "WT1" ;tag=E3B34B85-105452F2 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 3 [ 53]: To: ;tag=as4ae76676 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Header 12 [ 0]: [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981094 [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Stopping retransmission on 'bbc7e461-d5bca2bb-bed7a630@192.168.15.184' of Response 1: Match Found [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'bbc7e461-d5bca2bb-bed7a630@192.168.15.184' Method: ACK [2011-11-08 17:30:11] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' Method: INVITE [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: Auto destroying SIP dialog '10c01f9a-94f40040-f057de79@192.168.15.170' [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: Destroying SIP dialog 10c01f9a-94f40040-f057de79@192.168.15.170 [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: ---------- SIP HISTORY for '10c01f9a-94f40040-f057de79@192.168.15.170' [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: * SIP Call [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: 001. AuthChal Auth challenge sent for - nc 0 [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: 002. RegRequest Succeeded : Account [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: 003. AuthOK Auth challenge successful for 5226 [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: 004. CancelDestroy [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: 005. RegRequest Succeeded : Account [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: 006. AutoDestroy 10c01f9a-94f40040-f057de79@192.168.15.170 [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: ---------- END SIP HISTORY for '10c01f9a-94f40040-f057de79@192.168.15.170' [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'bbc7e461-d5bca2bb-bed7a630@192.168.15.184' Method: ACK [2011-11-08 17:30:12] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' Method: INVITE [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'bbc7e461-d5bca2bb-bed7a630@192.168.15.184' Method: ACK [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' Method: INVITE [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'bbc7e461-d5bca2bb-bed7a630@192.168.15.184' Method: ACK [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' Method: INVITE [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981086 (4) REGISTER - 2 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #5981086)) [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'bbc7e461-d5bca2bb-bed7a630@192.168.15.184' Method: ACK [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' Method: INVITE [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 0 [ 41]: INVITE sip:5261@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187;branch=z9hG4bKe34501dd52C2A310 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 2 [ 51]: From: ;tag=5455E8AA-EF355777 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 3 [ 48]: To: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 5 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 10 [ 26]: Supported: 100rel,replaces [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 11 [ 34]: Allow-Events: talk,hold,conference [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 14 [ 19]: Content-Length: 328 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 15 [ 0]: [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 1 [ 45]: o=- 1320791148 1320791149 IN IP4 192.168.15.187 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.187 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 5 [ 10]: a=sendonly [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 6 [ 26]: m=audio 2236 RTP/AVP 0 101 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 7 [ 10]: a=sendonly [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 10 [ 26]: m=video 0 RTP/AVP 34 98 99 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 11 [ 10]: a=sendrecv [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 12 [ 22]: a=rtpmap:34 H263/90000 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 13 [ 27]: a=rtpmap:98 h263-1998/90000 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Body 14 [ 22]: a=rtpmap:99 H264/90000 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-11-08 17:30:13] DEBUG[2629] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,replaces" [2011-11-08 17:30:13] DEBUG[2629] sip/reqresp_parser.c: Found SIP option: -100rel- [2011-11-08 17:30:13] DEBUG[2629] sip/reqresp_parser.c: Matched SIP option: 100rel [2011-11-08 17:30:13] DEBUG[2629] sip/reqresp_parser.c: Found SIP option: -replaces- [2011-11-08 17:30:13] DEBUG[2629] sip/reqresp_parser.c: Matched SIP option: replaces [2011-11-08 17:30:13] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:13] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Initializing initreq for method INVITE - callid 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing session-level SDP o=- 1320791148 1320791149 IN IP4 192.168.15.187... UNSUPPORTED. [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [2011-11-08 17:30:13] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:13] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.15.187... OK. [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing session-level SDP a=sendonly... OK. [2011-11-08 17:30:13] DEBUG[2629] rtp_engine.c: Setting payload 0 based on m type on 0x40299570 [2011-11-08 17:30:13] DEBUG[2629] rtp_engine.c: Setting payload 101 based on m type on 0x40299570 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-11-08 17:30:13] DEBUG[2629] rtp_engine.c: Incorporating payload 0 on 0x40299570 [2011-11-08 17:30:13] DEBUG[2629] rtp_engine.c: Incorporating payload 101 on 0x40299570 [2011-11-08 17:30:13] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x20571658' [2011-11-08 17:30:13] DEBUG[2629] rtp_engine.c: Copying payload 0 from 0x40299570 to 0x20571820 [2011-11-08 17:30:13] DEBUG[2629] rtp_engine.c: Copying payload 101 from 0x40299570 to 0x20571820 [2011-11-08 17:30:13] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x206a7f28' [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: We have an owner, now see if we need to change this call [2011-11-08 17:30:13] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x20571658' [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Got a SIP re-invite for call 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: SIP/5321-0001cd9a: This call is UP.... [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Setting framing from config on incoming call [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: -- Done with adding codecs to SDP [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981095 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'bbc7e461-d5bca2bb-bed7a630@192.168.15.184' Method: ACK [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' Method: INVITE [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:13] DEBUG[24823] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:13] DEBUG[14661] manager.c: Examining event: Event: Hold Privilege: call,all Status: On Channel: SIP/5321-0001cd9a Uniqueid: 1320791406.126036 [2011-11-08 17:30:13] DEBUG[14661] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/5261-0001cd99 UniqueID: 1320791406.126035 Class: default [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_musiconhold.c: SIP/5261-0001cd99 Opened file 0 '/var/lib/asterisk/moh/macroform-cold_day' [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 0 [ 38]: ACK sip:5261@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187;branch=z9hG4bK598ecc43AE697D76 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 2 [ 51]: From: ;tag=5455E8AA-EF355777 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 3 [ 48]: To: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 5 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Header 12 [ 0]: [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981095 [2011-11-08 17:30:13] DEBUG[2629] chan_sip.c: Stopping retransmission on '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' of Response 1: Match Found [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:13] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 0 [ 38]: REGISTER sip:192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.15.139;branch=z9hG4bK2303638BA7DF06F [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 2 [ 65]: From: "User37" ;tag=D9258B89-11310632 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 3 [ 27]: To: [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 4 [ 19]: CSeq: 5157 REGISTER [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: 5e4b484d-d1aa508b-9d6f5b94@192.168.15.139 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 6 [132]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 9 [179]: Authorization: Digest username="5237", realm="asterisk-phone.lusi.on.ca", nonce="65ca9f7f", uri="sip:192.168.15.251:5060", response="671472ac517f3d033214e7a307f44d07", algorithm=MD5 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 11 [ 12]: Expires: 300 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 13 [ 0]: [2011-11-08 17:30:14] DEBUG[2629] acl.c: For destination '192.168.15.139', our source address is '192.168.15.251'. [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.251:5060 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for 5e4b484d-d1aa508b-9d6f5b94@192.168.15.139 - REGISTER (No RTP) [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Initializing initreq for method REGISTER - callid 5e4b484d-d1aa508b-9d6f5b94@192.168.15.139 [2011-11-08 17:30:14] DEBUG[2629] netsock2.c: Splitting '192.168.15.139' into... [2011-11-08 17:30:14] DEBUG[2629] netsock2.c: ...host '192.168.15.139' and port ''. [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.15.139:5060 [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 0 [ 38]: REGISTER sip:192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.139;branch=z9hG4bK3541af568014C1ED [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 2 [ 65]: From: "User T" ;tag=D9258B89-11310632 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 3 [ 27]: To: [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 4 [ 19]: CSeq: 5158 REGISTER [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: 5e4b484d-d1aa508b-9d6f5b94@192.168.15.139 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 6 [132]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 9 [179]: Authorization: Digest username="5237", realm="asterisk-phone.lusi.on.ca", nonce="6a4a2550", uri="sip:192.168.15.251:5060", response="7a6d4e53ed57bcaf6392fd7281df863e", algorithm=MD5 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 11 [ 12]: Expires: 300 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Header 13 [ 0]: [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Initializing initreq for method REGISTER - callid 5e4b484d-d1aa508b-9d6f5b94@192.168.15.139 [2011-11-08 17:30:14] DEBUG[2629] netsock2.c: Splitting '192.168.15.139' into... [2011-11-08 17:30:14] DEBUG[2629] netsock2.c: ...host '192.168.15.139' and port ''. [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Store REGISTER's src-IP:port for call routing. [2011-11-08 17:30:14] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.139:5060 [2011-11-08 17:30:14] DEBUG[14661] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/5237 PeerStatus: Registered Address: 192.168.15.139:5060 [2011-11-08 17:30:14] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5237 [2011-11-08 17:30:14] DEBUG[2499] chan_sip.c: Checking device state for peer 5237 [2011-11-08 17:30:14] DEBUG[2499] devicestate.c: Changing state for SIP/5237 - state 1 (Not in use) [2011-11-08 17:30:14] DEBUG[2499] devicestate.c: device 'SIP/5237' state '1' [2011-11-08 17:30:14] DEBUG[2756] app_queue.c: Device 'SIP/5237' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:14] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 0 [ 52]: INVITE sip:5221@192.168.15.251:5060;user=phone SIP/2.0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187;branch=z9hG4bK1d4a77dcA4D0C10F [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 2 [ 63]: From: "User1" ;tag=F94A270E-80FB35DB [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 3 [ 38]: To: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 5 [ 47]: Call-ID: 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 10 [ 26]: Supported: 100rel,replaces [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 11 [ 34]: Allow-Events: talk,hold,conference [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 14 [ 19]: Content-Length: 270 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 15 [ 0]: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 1 [ 45]: o=- 1320791152 1320791152 IN IP4 192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 5 [ 10]: a=sendrecv [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 6 [ 31]: m=audio 2240 RTP/AVP 0 8 18 101 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 10 [ 19]: a=fmtp:18 annexb=no [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:15] DEBUG[2629] acl.c: For destination '192.168.15.187', our source address is '192.168.15.251'. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.251:5060 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for 55659242-cbacca8-e5c8eb75@192.168.15.187 - INVITE (No RTP) [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-11-08 17:30:15] DEBUG[2629] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,replaces" [2011-11-08 17:30:15] DEBUG[2629] sip/reqresp_parser.c: Found SIP option: -100rel- [2011-11-08 17:30:15] DEBUG[2629] sip/reqresp_parser.c: Matched SIP option: 100rel [2011-11-08 17:30:15] DEBUG[2629] sip/reqresp_parser.c: Found SIP option: -replaces- [2011-11-08 17:30:15] DEBUG[2629] sip/reqresp_parser.c: Matched SIP option: replaces [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Initializing initreq for method INVITE - callid 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981100 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 0 [ 38]: ACK sip:5221@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187;branch=z9hG4bK1d4a77dcA4D0C10F [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 2 [ 63]: From: "User1" ;tag=F94A270E-80FB35DB [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 3 [ 53]: To: ;tag=as7c7d189e [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 5 [ 47]: Call-ID: 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 12 [ 0]: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981100 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Stopping retransmission on '55659242-cbacca8-e5c8eb75@192.168.15.187' of Response 1: Match Found [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 0 [ 52]: INVITE sip:5221@192.168.15.251:5060;user=phone SIP/2.0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187;branch=z9hG4bK2c67a041D113A174 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 2 [ 63]: From: "User1" ;tag=F94A270E-80FB35DB [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 3 [ 38]: To: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 4 [ 14]: CSeq: 2 INVITE [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 5 [ 47]: Call-ID: 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 10 [ 26]: Supported: 100rel,replaces [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 11 [ 34]: Allow-Events: talk,hold,conference [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 12 [195]: Authorization: Digest username="5321", realm="asterisk-phone.lusi.on.ca", nonce="596e2521", uri="sip:5221@192.168.15.251:5060;user=phone", response="20462517779cc727f18852460777de1b", algorithm=MD5 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 15 [ 19]: Content-Length: 270 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 16 [ 0]: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 1 [ 45]: o=- 1320791152 1320791152 IN IP4 192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 5 [ 10]: a=sendrecv [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 6 [ 31]: m=audio 2240 RTP/AVP 0 8 18 101 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 10 [ 19]: a=fmtp:18 annexb=no [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Initializing initreq for method INVITE - callid 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x20687428' [2011-11-08 17:30:15] DEBUG[2629] res_rtp_asterisk.c: Allocated port 19214 for RTP instance '0x20687428' [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: RTP instance '0x20687428' is setup and ready to go [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x20638958' [2011-11-08 17:30:15] DEBUG[2629] res_rtp_asterisk.c: Allocated port 11898 for RTP instance '0x20638958' [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: RTP instance '0x20638958' is setup and ready to go [2011-11-08 17:30:15] DEBUG[2629] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x20638958' [2011-11-08 17:30:15] DEBUG[2629] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x20687428' [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Setting NAT on RTP to On [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Setting NAT on VRTP to On [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing session-level SDP o=- 1320791152 1320791152 IN IP4 192.168.15.187... UNSUPPORTED. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.15.187... OK. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Setting payload 0 based on m type on 0x40299570 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Setting payload 8 based on m type on 0x40299570 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Setting payload 18 based on m type on 0x40299570 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Setting payload 101 based on m type on 0x40299570 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Incorporating payload 0 on 0x40299570 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Incorporating payload 8 on 0x40299570 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Incorporating payload 18 on 0x40299570 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Incorporating payload 101 on 0x40299570 [2011-11-08 17:30:15] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x20687428' [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Copying payload 0 from 0x40299570 to 0x206875f0 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Copying payload 8 from 0x40299570 to 0x206875f0 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Copying payload 18 from 0x40299570 to 0x206875f0 [2011-11-08 17:30:15] DEBUG[2629] rtp_engine.c: Copying payload 101 from 0x40299570 to 0x206875f0 [2011-11-08 17:30:15] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x20638958' [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Checking SIP call limits for device 5321 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Updating call counter for incoming call [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Call from peer '5321' is 2 out of 100 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: *** Our native formats are 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: *** Our capabilities are 0x380006 (gsm|ulaw|h263|h263p|h264) [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: This channel can handle video! HOLLYWOOD next! [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: build_route: Contact hop: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: SIP/5321-0001cd9b: New call is still down.... Trying... [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:15] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 2 (In use) [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: device 'SIP/5321' state '2' [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:15] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 2 (In use) [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: device 'SIP/5321' state '2' [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/5321-0001cd9b ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 5321 CallerIDName: User1 AccountCode: Exten: 5221 Context: DLPN_DialPlan1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: SIPURI Value: sip:5321@192.168.15.187 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: SIPDOMAIN Value: 192.168.15.251:5060 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: SIPCALLID Value: 55659242-cbacca8-e5c8eb75@192.168.15.187 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5321-0001cd9b Uniqueid: 1320791415.126037 Channeltype: SIP SIPcallid: 55659242-cbacca8-e5c8eb75@192.168.15.187 SIPfullcontact: sip:5321@192.168.15.187 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5321-0001cd9b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 5321 CallerIDName: User1 ConnectedLineNum: ConnectedLineName: Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 2 Paused: 0 [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Result of 'HINT' is 'SIP/5221' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Launching 'Macro' [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5321-0001cd9b Context: DLPN_DialPlan1 Extension: 5221 Priority: 1 Application: Macro AppData: stdexten,5221,SIP/5221 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: MACRO_EXTEN Value: 5221 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: MACRO_CONTEXT Value: DLPN_DialPlan1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: MACRO_PRIORITY Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: ARG1 Value: 5221 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: ARG2 Value: SIP/5221 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Result of 'FEATURES' is '' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Launching 'Set' [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5321-0001cd9b Context: macro-stdexten Extension: s Priority: 1 Application: Set AppData: __DYNAMIC_FEATURES= Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: __DYNAMIC_FEATURES Value: Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[24824] app_macro.c: Executed application: Set [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Result of 'ARG1' is '5221' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Launching 'Set' [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5321-0001cd9b Context: macro-stdexten Extension: s Priority: 2 Application: Set AppData: ORIG_ARG1=5221 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: ORIG_ARG1 Value: 5221 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[24824] app_macro.c: Executed application: Set [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Result of 'ARG1' is '5221' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Result of 'FOLLOWME_5221' is '0' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Expression result is '0' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Launching 'GotoIf' [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5321-0001cd9b Context: macro-stdexten Extension: s Priority: 3 Application: GotoIf AppData: 0?6:4 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[24824] app_macro.c: Executed application: GotoIf [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Result of 'ARG2' is 'SIP/5221' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Result of 'RINGTIME' is '30' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Result of 'DIALOPTIONS' is '' [2011-11-08 17:30:15] DEBUG[24824] pbx.c: Launching 'Dial' [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5321-0001cd9b Context: macro-stdexten Extension: s Priority: 4 Application: Dial AppData: SIP/5221,30, Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: DIALSTATUS Value: Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: DIALEDPEERNUMBER Value: Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: DIALEDPEERNAME Value: Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: ANSWEREDTIME Value: Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: DIALEDTIME Value: Uniqueid: 1320791415.126037 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Allocating new SIP dialog for 7ecc7c5f64993916599b79d60c17b952@192.168.7.11:0 - INVITE (No RTP) [2011-11-08 17:30:15] DEBUG[24824] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x206a1f08' [2011-11-08 17:30:15] DEBUG[24824] res_rtp_asterisk.c: Allocated port 17724 for RTP instance '0x206a1f08' [2011-11-08 17:30:15] DEBUG[24824] rtp_engine.c: RTP instance '0x206a1f08' is setup and ready to go [2011-11-08 17:30:15] DEBUG[24824] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x206a1f08' [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Setting NAT on RTP to On [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2011-11-08 17:30:15] DEBUG[24824] acl.c: For destination '192.168.7.2', our source address is '192.168.7.11'. [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Setting SIP_TRANSPORT_TCP with address 192.168.7.11:5060 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/5221-0001cd9c ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 5221 CallerIDName: User1 AccountCode: Exten: Context: DLPN_LUSI_Internet_Unrestricted Uniqueid: 1320791415.126038 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: *** Our native formats are 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: *** Our capabilities are 0x80e (gsm|ulaw|alaw|g726) [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: This channel will not be able to handle video. [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9c Variable: SIPCALLID Value: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 Uniqueid: 1320791415.126038 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5221-0001cd9c Uniqueid: 1320791415.126038 Channeltype: SIP SIPcallid: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 SIPfullcontact: sip:5221@192.168.7.2:57144;transport=tcp [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5221-0001cd9c Channeltype: SIP SIPcallid: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 SIPfullcontact: sip:5221@192.168.7.2:57144;transport=tcp Peername: 5221 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9c Variable: DIALEDPEERNUMBER Value: 5221 Uniqueid: 1320791415.126038 [2011-11-08 17:30:15] DEBUG[24824] rtp_engine.c: Seeded SDP of 'SIP/5221-0001cd9c' with that of 'SIP/5321-0001cd9b' [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable DIALEDTIME. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable ANSWEREDTIME. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable DIALEDPEERNAME. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable DIALEDPEERNUMBER. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable DIALSTATUS. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable MACRO_DEPTH. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable ORIG_ARG1. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Copying hard-transferable variable DYNAMIC_FEATURES. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable ARG2. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable ARG1. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable MACRO_PRIORITY. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable MACRO_CONTEXT. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable MACRO_EXTEN. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable SIPCALLID. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable SIPDOMAIN. [2011-11-08 17:30:15] DEBUG[24824] channel.c: Not copying variable SIPURI. [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Outgoing Call for 5221 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Call to peer '5221' is 1 out of 100 [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:15] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 6 (Ringing) [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: device 'SIP/5221' state '6' [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5221 Context: default Hint: SIP/5221 Status: 8 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: ** Our capability: 0x80e (gsm|ulaw|alaw|g726) Video flag: False Text flag: False [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: -- Done with adding codecs to SDP [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Done building SDP. Settling with this capability: 0x80e (gsm|ulaw|alaw|g726) [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Initializing initreq for method INVITE - callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 0 [ 55]: INVITE sip:5221@192.168.7.2:57144;transport=tcp SIP/2.0 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/TCP 192.168.7.11:5060;branch=z9hG4bK453897c6;rport [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 4 [ 46]: To: [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 5 [ 51]: Contact: [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 6 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 9 [ 35]: Date: Tue, 08 Nov 2011 22:30:15 GMT [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 12 [ 88]: Remote-Party-ID: "User1" ;party=calling;privacy=off;screen=no [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:15] DEBUG[24824] chan_sip.c: Trying to put 'INVITE sip:' onto TCP socket destined for 192.168.7.2:57144 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/5321-0001cd9b Destination: SIP/5221-0001cd9c CallerIDNum: 5321 CallerIDName: User1 ConnectedLineNum: ConnectedLineName: UniqueID: 1320791415.126037 DestUniqueID: 1320791415.126038 Dialstring: 5221 [2011-11-08 17:30:15] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '2' (In use) [2011-11-08 17:30:15] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '2' (In use) [2011-11-08 17:30:15] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 2 Paused: 0 [2011-11-08 17:30:15] DEBUG[2500] app_queue.c: Extension '5221@default' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 0 [ 38]: REGISTER sip:192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.146;branch=z9hG4bK2f64fa5a5448DD2F [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 2 [ 66]: From: "User S" ;tag=23FCFABA-591219A1 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 3 [ 27]: To: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 4 [ 19]: CSeq: 5157 REGISTER [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: e359715e-d52bdef4-3dad3ea3@192.168.15.146 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 6 [132]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 9 [179]: Authorization: Digest username="5235", realm="asterisk-phone.lusi.on.ca", nonce="3480233a", uri="sip:192.168.15.251:5060", response="9dbc14463ad36c9e320b1c008f75abde", algorithm=MD5 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 11 [ 12]: Expires: 300 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 13 [ 0]: [2011-11-08 17:30:15] DEBUG[2629] acl.c: For destination '192.168.15.146', our source address is '192.168.15.251'. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.251:5060 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for e359715e-d52bdef4-3dad3ea3@192.168.15.146 - REGISTER (No RTP) [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Initializing initreq for method REGISTER - callid e359715e-d52bdef4-3dad3ea3@192.168.15.146 [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: Splitting '192.168.15.146' into... [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: ...host '192.168.15.146' and port ''. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.15.146:5060 [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 0 [ 38]: REGISTER sip:192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.146;branch=z9hG4bKcc9fc6d0B333C78D [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 2 [ 66]: From: "User S" ;tag=23FCFABA-591219A1 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 3 [ 27]: To: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 4 [ 19]: CSeq: 5158 REGISTER [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: e359715e-d52bdef4-3dad3ea3@192.168.15.146 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 6 [132]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 9 [179]: Authorization: Digest username="5235", realm="asterisk-phone.lusi.on.ca", nonce="0dd5f8e3", uri="sip:192.168.15.251:5060", response="b4e4a83c7a3b819dd58fef31fe3d104e", algorithm=MD5 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 11 [ 12]: Expires: 300 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Header 13 [ 0]: [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Initializing initreq for method REGISTER - callid e359715e-d52bdef4-3dad3ea3@192.168.15.146 [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: Splitting '192.168.15.146' into... [2011-11-08 17:30:15] DEBUG[2629] netsock2.c: ...host '192.168.15.146' and port ''. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Store REGISTER's src-IP:port for call routing. [2011-11-08 17:30:15] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.146:5060 [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/5235 PeerStatus: Registered Address: 192.168.15.146:5060 [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5235 [2011-11-08 17:30:15] DEBUG[2499] chan_sip.c: Checking device state for peer 5235 [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: Changing state for SIP/5235 - state 1 (Not in use) [2011-11-08 17:30:15] DEBUG[2499] devicestate.c: device 'SIP/5235' state '1' [2011-11-08 17:30:15] DEBUG[2756] app_queue.c: Device 'SIP/5235' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: Got RTCP report of 76 bytes [2011-11-08 17:30:15] DEBUG[14661] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.15.184:2231 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 49916 SequenceNumberCycles: 0 IAJitter: 12 LastSR: 0.0000000000 DLSR: 0.0000(sec) [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:15] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.15.187:2237 [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: Got RTCP report of 76 bytes [2011-11-08 17:30:16] DEBUG[14661] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.15.187:2237 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 1 FractionLost: 1 PacketsLost: 0 HighestSequence: 4074 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:16] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 0 [ 37]: PUBLISH sip:5221@192.168.7.11 SIP/2.0 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 1 [ 95]: Via: SIP/2.0/TCP 192.168.7.2:57144;rport;branch=z9hG4bKPjjmPPFep5HBVRf6PHg28mc3qbUtJAXEGz;alias [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 3 [ 66]: From: ;tag=LmeUSJ39QZ49sqgHf9NxXtQe0LNLwmq9 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 4 [ 27]: To: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 5 [ 41]: Call-ID: 4LULHPBL2boOV9CUGIUDvVTrWEdJZS74 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 6 [ 18]: CSeq: 7211 PUBLISH [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 7 [ 15]: Event: presence [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 8 [ 29]: User-Agent: Bria iPhone 1.3.4 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 9 [ 34]: Content-Type: application/pidf+xml [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 564 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 0 [ 37]: PUBLISH sip:5221@192.168.7.11 SIP/2.0 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 1 [ 95]: Via: SIP/2.0/TCP 192.168.7.2:57144;rport;branch=z9hG4bKPjjmPPFep5HBVRf6PHg28mc3qbUtJAXEGz;alias [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 3 [ 66]: From: ;tag=LmeUSJ39QZ49sqgHf9NxXtQe0LNLwmq9 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 4 [ 27]: To: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 5 [ 41]: Call-ID: 4LULHPBL2boOV9CUGIUDvVTrWEdJZS74 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 6 [ 18]: CSeq: 7211 PUBLISH [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 7 [ 15]: Event: presence [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 8 [ 29]: User-Agent: Bria iPhone 1.3.4 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 9 [ 34]: Content-Type: application/pidf+xml [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 564 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 0 [ 38]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 1 [173]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 2 [ 48]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 3 [ 10]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 4 [ 22]: open [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 5 [ 11]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 6 [ 49]: 2011-11-08T17:30:16.012Z [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 7 [ 24]: Available [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 8 [ 9]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 9 [ 52]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 10 [ 19]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 11 [ 19]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 12 [ 20]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 13 [ 30]: Available [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 14 [ 13]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Body 15 [ 11]: [2011-11-08 17:30:17] DEBUG[28924] acl.c: For destination '192.168.7.2', our source address is '192.168.7.11'. [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Setting SIP_TRANSPORT_TCP with address 192.168.7.11:5060 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Allocating new SIP dialog for 4LULHPBL2boOV9CUGIUDvVTrWEdJZS74 - PUBLISH (No RTP) [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: **** Received PUBLISH (15) - Command in SIP PUBLISH [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Trying to put 'SIP/2.0 489' onto TCP socket destined for 192.168.7.2:57144 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: SIP message could not be handled, bad request: 4LULHPBL2boOV9CUGIUDvVTrWEdJZS74 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 4 [ 26]: To: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 7 [ 0]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 4 [ 26]: To: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 7 [ 0]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: SIP response 100 to standard invite [2011-11-08 17:30:17] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: ~HASH~SIP_CAUSE~SIP/5221-0001cd9c~ Value: SIP 100 Trying Uniqueid: 1320791415.126037 [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 6 [ 51]: Contact: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 8 [ 18]: Content-Length: 0 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 9 [ 0]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 6 [ 51]: Contact: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 8 [ 18]: Content-Length: 0 [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: Header 9 [ 0]: [2011-11-08 17:30:17] DEBUG[28924] chan_sip.c: SIP response 180 to standard invite [2011-11-08 17:30:17] DEBUG[24824] rtp_engine.c: Setting early bridge SDP of 'SIP/5321-0001cd9b' with that of 'SIP/5221-0001cd9c' [2011-11-08 17:30:17] DEBUG[24824] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:17] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:17] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:17] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 6 (Ringing) [2011-11-08 17:30:17] DEBUG[2499] devicestate.c: device 'SIP/5221' state '6' [2011-11-08 17:30:17] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5221-0001cd9c ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 5221 CallerIDName: User1 ConnectedLineNum: 5321 ConnectedLineName: User1 Uniqueid: 1320791415.126038 [2011-11-08 17:30:17] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: ~HASH~SIP_CAUSE~SIP/5221-0001cd9c~ Value: SIP 180 Ringing Uniqueid: 1320791415.126037 [2011-11-08 17:30:17] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981086 (5) REGISTER - 2 [2011-11-08 17:30:17] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #5981086)) [2011-11-08 17:30:17] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:17] DEBUG[2629] chan_sip.c: Destroying SIP dialog 4LULHPBL2boOV9CUGIUDvVTrWEdJZS74 [2011-11-08 17:30:17] DEBUG[2629] chan_sip.c: ---------- SIP HISTORY for '4LULHPBL2boOV9CUGIUDvVTrWEdJZS74' [2011-11-08 17:30:17] DEBUG[2629] chan_sip.c: * SIP Call [2011-11-08 17:30:17] DEBUG[2629] chan_sip.c: 001. NeedDestroy Setting needdestroy because unknown event package in publish [2011-11-08 17:30:17] DEBUG[2629] chan_sip.c: ---------- END SIP HISTORY for '4LULHPBL2boOV9CUGIUDvVTrWEdJZS74' [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:17] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[14661] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 192.168.15.184:2231 OurSSRC: 1722147589 SentNTP: 1320791418.2102013952 SentRTP: 40000 SentPackets: 250 SentOctets: 40000 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0001 TheirLastSR: 737542800 DLSR: 2.6160 (sec) [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:18] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: No remote address on RTP instance '0x20571658' so dropping frame [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 0 [ 40]: REFER sip:5261@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187;branch=z9hG4bK8beed50d1A54F640 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 2 [ 51]: From: ;tag=5455E8AA-EF355777 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 3 [ 48]: To: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 4 [ 13]: CSeq: 2 REFER [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 5 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 9 [152]: Refer-To: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 10 [ 36]: Referred-By: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 11 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 13 [ 0]: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 55659242-cbacca8-e5c8eb75@192.168.15.187 (No check of from/to tags) [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP attended transfer: Transferer channel SIP/5321-0001cd9a, transferee channel SIP/5261-0001cd99 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/5261-0001cd99' [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Looking for callid 55659242-cbacca8-e5c8eb75@192.168.15.187 (fromtag F94A270E-80FB35DB totag as3b2f78b5) [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Matched INCOMING call - their tag is F94A270E-80FB35DB Our tag is as3b2f78b5 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP attended transfer: Attempting transfer in ringing state [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP attended transfer: trying to bridge SIP/5321-0001cd9b and SIP/5261-0001cd99 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Sip transfer:-------------------- [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: -- Transferer to PBX channel: SIP/5321-0001cd9a State Up [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: -- Transferer to PBX second channel (target): SIP/5321-0001cd9b State Ring [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: -- Bridged call to transferee: SIP/5261-0001cd99 State Up [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: -- No target second channel --- [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: -- END Sip transfer:-------------------- [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP transfer: Four channels to handle [2011-11-08 17:30:19] DEBUG[2629] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP transfer: trying to masquerade SIP/5261-0001cd99 into SIP/5321-0001cd9b [2011-11-08 17:30:19] DEBUG[2629] channel.c: Planning to masquerade channel SIP/5261-0001cd99 into the structure of SIP/5321-0001cd9b [2011-11-08 17:30:19] DEBUG[2629] channel.c: Done planning to masquerade channel SIP/5261-0001cd99 into the structure of SIP/5321-0001cd9b [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Strict routing enforced for session 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:19] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:19] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981108 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:19] DEBUG[2629] channel.c: Actually Masquerading SIP/5261-0001cd99(6) into the structure of SIP/5321-0001cd9b(4) [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP Fixup: New owner for dialogue 55659242-cbacca8-e5c8eb75@192.168.15.187: SIP/5261-0001cd99 (Old parent: SIP/5261-0001cd99) [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Hangup call SIP/5261-0001cd99, SIP callid 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: update_call_counter(5321) - decrement call limit counter on hangup [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Updating call counter for incoming call [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Call from peer '5321' removed from call limit 100 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Hanging up channel in state Ring (not UP) [2011-11-08 17:30:19] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x20687428' [2011-11-08 17:30:19] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x20638958' [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: AST hangup cause 16 (no match found in SIP) [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981110 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 603' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:19] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 2 (In use) [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: device 'SIP/5321' state '2' [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: SIPREFERRINGCONTEXT Value: DLPN_DialPlan1 Uniqueid: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: SIPREFERREDBYHDR Value: Uniqueid: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Attended Channel: SIP/5321-0001cd9a Uniqueid: 1320791406.126036 SIP-Callid: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 TargetChannel: SIP/5321-0001cd9b TargetUniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/5261-0001cd99 UniqueID: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/5261-0001cd99 CloneState: Up Original: SIP/5321-0001cd9b OriginalState: Ring [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/5261-0001cd99 Newname: SIP/5261-0001cd99 Uniqueid: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/5321-0001cd9b Newname: SIP/5261-0001cd99 Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/5261-0001cd99 Newname: SIP/5321-0001cd9b Uniqueid: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/5261-0001cd99 CallerIDNum: 5261 CallerIDName: WT1 Uniqueid: 1320791415.126037 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 2 Paused: 0 [2011-11-08 17:30:19] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '2' (In use) [2011-11-08 17:30:19] DEBUG[2629] channel.c: Putting channel SIP/5261-0001cd99 in ulaw/ulaw formats [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP Fixup: New owner for dialogue bbc7e461-d5bca2bb-bed7a630@192.168.15.184: SIP/5261-0001cd99 (Old parent: SIP/5321-0001cd9b) [2011-11-08 17:30:19] DEBUG[2629] channel.c: Driver for channel 'SIP/5261-0001cd99' does not support indication 3, emulating it [2011-11-08 17:30:19] DEBUG[2629] channel.c: Set channel SIP/5261-0001cd99 to write format slin [2011-11-08 17:30:19] DEBUG[2629] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-11-08 17:30:19] DEBUG[2629] channel.c: Released clone lock on 'SIP/5321-0001cd9b' [2011-11-08 17:30:19] DEBUG[2629] channel.c: Done Masquerading SIP/5261-0001cd99 (6) [2011-11-08 17:30:19] DEBUG[2629] res_rtp_asterisk.c: Changing ssrc from 1722147589 to 605060448 due to a source change [2011-11-08 17:30:19] DEBUG[24823] rtp_engine.c: rtp-engine-local-bridge: Oooh, something is weird, backing out [2011-11-08 17:30:19] DEBUG[24823] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/5321-0001cd9b, c1=SIP/5321-0001cd9a, flags: Yes,Yes,No,No [2011-11-08 17:30:19] DEBUG[24823] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:19] DEBUG[24823] channel.c: Bridge stops bridging channels SIP/5321-0001cd9b and SIP/5321-0001cd9a [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is '2011-11-08 17:30:06' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is '"WT1" <5261>' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is 'DLPN_LUSI_Internet_Unrestricted' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is 'SIP/5261-0001cd99' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is 'SIP/5321-0001cd9a' [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/5321-0001cd9b Channel2: SIP/5321-0001cd9a Uniqueid1: 1320791406.126035 Uniqueid2: 1320791406.126036 CallerID1: 5321 CallerID2: 5321 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: DIALEDTIME Value: 4 Uniqueid: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is 'Dial' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is 'SIP/5321' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is '13' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is '8' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is 'ANSWERED' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is 'DOCUMENTATION' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is '(null)' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is '1320791406.126035' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is '(null)' [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Function result is '(null)' [2011-11-08 17:30:19] DEBUG[24823] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2011-11-08 17:30:06','"WT1" <5261>','DLPN_LUSI_Internet_Unrestricted','SIP/5261-0001cd99','SIP/5321-0001cd9a','Dial','SIP/5321','13','8','ANSWERED','DOCUMENTATION','','1320791406.126035','','') [2011-11-08 17:30:19] DEBUG[2629] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:19] DEBUG[2629] channel.c: Set channel SIP/5261-0001cd99 to write format ulaw [2011-11-08 17:30:19] DEBUG[2629] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-11-08 17:30:19] DEBUG[2629] channel.c: Driver for channel 'SIP/5261-0001cd99' does not support indication 3, emulating it [2011-11-08 17:30:19] DEBUG[2629] channel.c: Set channel SIP/5261-0001cd99 to write format slin [2011-11-08 17:30:19] DEBUG[2629] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Strict routing enforced for session bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:19] DEBUG[24824] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:19] DEBUG[24824] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: -- Done with adding codecs to SDP [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Initializing already initialized SIP dialog bbc7e461-d5bca2bb-bed7a630@192.168.15.184 (presumably reinvite) [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 0 [ 36]: INVITE sip:5261@192.168.15.184 SIP/2.0 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.15.251:5060;branch=z9hG4bK1625d29f;rport [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 3 [ 55]: From: ;tag=as4ae76676 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 4 [ 55]: To: "WT1" ;tag=E3B34B85-105452F2 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 5 [ 37]: Contact: [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 6 [ 48]: Call-ID: bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 11 [ 87]: Remote-Party-ID: "User1" ;party=called;privacy=off;screen=no [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981111 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Trying to put 'UPDATE sip:' onto TCP socket destined for 192.168.7.2:57144 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/5261-0001cd99 UniqueID: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[24824] res_rtp_asterisk.c: No remote address on RTP instance '0x206a1f08' so dropping frame [2011-11-08 17:30:19] DEBUG[24824] res_rtp_asterisk.c: No remote address on RTP instance '0x206a1f08' so dropping frame [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.15.251:5060;branch=z9hG4bK1625d29f;rport [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 2 [ 55]: From: ;tag=as4ae76676 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 3 [ 55]: To: "WT1" ;tag=E3B34B85-105452F2 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 11 [ 19]: Content-Length: 199 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 12 [ 0]: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 1 [ 45]: o=- 1320791388 1320791389 IN IP4 192.168.15.184 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.184 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 5 [ 26]: m=audio 2230 RTP/AVP 0 101 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 6 [ 10]: a=sendrecv [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Acked pending invite 102 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981111 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Stopping retransmission on 'bbc7e461-d5bca2bb-bed7a630@192.168.15.184' of Request 102: Match Found [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: SIP response 200 to RE-invite on outgoing call bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Processing session-level SDP o=- 1320791388 1320791389 IN IP4 192.168.15.184... UNSUPPORTED. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [2011-11-08 17:30:19] DEBUG[2629] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:19] DEBUG[2629] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.15.184... OK. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-11-08 17:30:19] DEBUG[2629] rtp_engine.c: Setting payload 0 based on m type on 0x40298c30 [2011-11-08 17:30:19] DEBUG[2629] rtp_engine.c: Setting payload 101 based on m type on 0x40298c30 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-11-08 17:30:19] DEBUG[2629] rtp_engine.c: Incorporating payload 0 on 0x40298c30 [2011-11-08 17:30:19] DEBUG[2629] rtp_engine.c: Incorporating payload 101 on 0x40298c30 [2011-11-08 17:30:19] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2aaacc025808' [2011-11-08 17:30:19] DEBUG[2629] rtp_engine.c: Copying payload 0 from 0x40298c30 to 0x2aaacc0259d0 [2011-11-08 17:30:19] DEBUG[2629] rtp_engine.c: Copying payload 101 from 0x40298c30 to 0x2aaacc0259d0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: We have an owner, now see if we need to change this call [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Updating call counter for incoming call [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Strict routing enforced for session bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:19] DEBUG[2629] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:19] DEBUG[2629] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Trying to put 'ACK sip:526' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:19] DEBUG[24824] res_rtp_asterisk.c: No remote address on RTP instance '0x206a1f08' so dropping frame [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:19] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 2 (In use) [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: device 'SIP/5261' state '2' [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: ~HASH~SIP_CAUSE~SIP/5261-0001cd99~ Value: SIP 200 OK Uniqueid: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.15.251:5060;branch=z9hG4bK400aebb2;rport [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 2 [ 50]: From: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 3 [ 49]: To: ;tag=5455E8AA-EF355777 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 4 [ 16]: CSeq: 103 NOTIFY [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 5 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 7 [ 17]: Event: refer;id=2 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981108 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Stopping retransmission on '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' of Request 103: Match Found [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Got 200 OK on NOTIFY for transfer [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: ~HASH~SIP_CAUSE~SIP/5321-0001cd9a~ Value: SIP 200 OK Uniqueid: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[24823] channel.c: Hanging up channel 'SIP/5321-0001cd9a' [2011-11-08 17:30:19] DEBUG[24823] chan_sip.c: update_call_counter(5321) - decrement call limit counter on hangup [2011-11-08 17:30:19] DEBUG[24823] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:19] DEBUG[24823] chan_sip.c: Call to peer '5321' removed from call limit 100 [2011-11-08 17:30:19] DEBUG[24823] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060. [2011-11-08 17:30:19] DEBUG[24823] app_dial.c: Exiting with DIALSTATUS=ANSWER. [2011-11-08 17:30:19] DEBUG[24823] pbx.c: Spawn extension (DLPN_LUSI_Internet_Unrestricted,5321,1) exited non-zero on 'SIP/5321-0001cd9b' [2011-11-08 17:30:19] DEBUG[24823] channel.c: Soft-Hanging up channel 'SIP/5321-0001cd9b' [2011-11-08 17:30:19] DEBUG[24823] channel.c: Hanging up zombie 'SIP/5321-0001cd9b' [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:19] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 1 (Not in use) [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: device 'SIP/5321' state '1' [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:19] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 1 (Not in use) [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: device 'SIP/5321' state '1' [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:19] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 1 (Not in use) [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: device 'SIP/5321' state '1' [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/5321-0001cd9a Uniqueid: 1320791406.126036 CallerIDNum: 5321 CallerIDName: User1 ConnectedLineNum: 5261 ConnectedLineName: WT1 Cause: 16 Cause-txt: Normal Clearing [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5321-0001cd9b Variable: DIALSTATUS Value: ANSWER Uniqueid: 1320791406.126035 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/5321-0001cd9b UniqueID: 1320791406.126035 DialStatus: ANSWER [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/5321-0001cd9b Uniqueid: 1320791406.126035 CallerIDNum: 5321 CallerIDName: User1 ConnectedLineNum: 5221 ConnectedLineName: User1 Cause: 16 Cause-txt: Normal Clearing [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5321 Context: default Hint: SIP/5321 Status: 0 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 1 Paused: 0 [2011-11-08 17:30:19] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '1' (Not in use) [2011-11-08 17:30:19] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '1' (Not in use) [2011-11-08 17:30:19] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '1' (Not in use) [2011-11-08 17:30:19] DEBUG[2500] app_queue.c: Extension '5321@default' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 1 Paused: 0 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 1 Paused: 0 [2011-11-08 17:30:19] DEBUG[24824] res_rtp_asterisk.c: No remote address on RTP instance '0x206a1f08' so dropping frame [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 0 [ 38]: BYE sip:5261@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187;branch=z9hG4bKac099f73F7CCD0A6 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 2 [ 51]: From: ;tag=5455E8AA-EF355777 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 3 [ 48]: To: "WT1" ;tag=as6d6faef3 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 4 [ 11]: CSeq: 3 BYE [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 5 [ 59]: Call-ID: 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Initializing initreq for method BYE - callid 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:19] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:19] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Setting SIP_ALREADYGONE on dialog 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:19] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x20571658' [2011-11-08 17:30:19] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x206a7f28' [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Received bye, no owner, selfdestruct soon. [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:19] DEBUG[24824] res_rtp_asterisk.c: No remote address on RTP instance '0x206a1f08' so dropping frame [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 0 [ 33]: SIP/2.0 500 Internal Server Error [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK0e99e6cf;alias [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 103 UPDATE [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 7 [ 0]: [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 0 [ 33]: SIP/2.0 500 Internal Server Error [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK0e99e6cf;alias [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 103 UPDATE [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Header 7 [ 0]: [2011-11-08 17:30:19] DEBUG[28924] chan_sip.c: Setting SIP_ALREADYGONE on dialog 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:19] DEBUG[24824] channel.c: Hanging up channel 'SIP/5221-0001cd9c' [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Hangup call SIP/5221-0001cd9c, SIP callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: update_call_counter(5221) - decrement call limit counter on hangup [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Call to peer '5221' removed from call limit 100 [2011-11-08 17:30:19] DEBUG[24824] chan_sip.c: Hanging up channel in state Ringing (not UP) [2011-11-08 17:30:19] DEBUG[24824] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x206a1f08' [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:19] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 1 (Not in use) [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: device 'SIP/5221' state '1' [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:19] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 1 (Not in use) [2011-11-08 17:30:19] DEBUG[2499] devicestate.c: device 'SIP/5221' state '1' [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: ~HASH~SIP_CAUSE~SIP/5221-0001cd9c~ Value: SIP 500 Internal Server Error Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:19] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/5221-0001cd9c Uniqueid: 1320791415.126038 CallerIDNum: 5221 CallerIDName: User1 ConnectedLineNum: 5261 ConnectedLineName: WT1 Cause: 38 Cause-txt: Network out of order [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: DIALSTATUS Value: CONGESTION Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5221 Context: default Hint: SIP/5221 Status: 0 [2011-11-08 17:30:19] DEBUG[24824] app_dial.c: Exiting with DIALSTATUS=CONGESTION. [2011-11-08 17:30:19] DEBUG[24824] app_macro.c: Executed application: Dial [2011-11-08 17:30:19] DEBUG[24824] pbx.c: Result of 'DIALSTATUS' is 'CONGESTION' [2011-11-08 17:30:19] DEBUG[24824] pbx.c: Launching 'Goto' [2011-11-08 17:30:19] DEBUG[24824] app_macro.c: Executed application: Goto [2011-11-08 17:30:19] DEBUG[24824] pbx.c: Launching 'Goto' [2011-11-08 17:30:19] DEBUG[24824] app_macro.c: Executed application: Goto [2011-11-08 17:30:19] DEBUG[24824] pbx.c: Result of 'ORIG_ARG1' is '5221' [2011-11-08 17:30:19] DEBUG[24824] pbx.c: Launching 'VoiceMail' [2011-11-08 17:30:19] DEBUG[24824] app_voicemail.c: Before find_user [2011-11-08 17:30:19] DEBUG[24824] channel.c: Set channel SIP/5261-0001cd99 to write format ulaw [2011-11-08 17:30:19] DEBUG[24824] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-11-08 17:30:19] DEBUG[24824] channel.c: Set channel SIP/5261-0001cd99 to write format slin [2011-11-08 17:30:19] DEBUG[2500] app_queue.c: Extension '5221@default' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:19] DEBUG[24824] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/5261-0001cd99 UniqueID: 1320791415.126037 DialStatus: CONGESTION [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd99 Context: macro-stdexten Extension: s Priority: 5 Application: Goto AppData: s-CONGESTION,1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd99 Context: macro-stdexten Extension: s-CONGESTION Priority: 1 Application: Goto AppData: s-NOANSWER,1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd99 Context: macro-stdexten Extension: s-NOANSWER Priority: 1 Application: VoiceMail AppData: 5221,u Uniqueid: 1320791415.126037 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 0 [ 38]: ACK sip:5221@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.187;branch=z9hG4bK2c67a041D113A174 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 2 [ 63]: From: "User1" ;tag=F94A270E-80FB35DB [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 3 [ 53]: To: ;tag=as3b2f78b5 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 4 [ 11]: CSeq: 2 ACK [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 5 [ 47]: Call-ID: 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Header 12 [ 0]: [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981110 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Stopping retransmission on '55659242-cbacca8-e5c8eb75@192.168.15.187' of Response 2: Match Found [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: Destroying SIP dialog 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: ---------- SIP HISTORY for '72b12deb77477be446e585d404781e14@192.168.7.11:5060' [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: * SIP Call [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: 001. ConnectedLine Calling party is now WT1 <5261> [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: 002. Cancel Cause Network out of order [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: 003. NeedDestroy Setting needdestroy because hangup [2011-11-08 17:30:19] DEBUG[2629] chan_sip.c: ---------- END SIP HISTORY for '72b12deb77477be446e585d404781e14@192.168.7.11:5060' [2011-11-08 17:30:19] DEBUG[2629] rtp_engine.c: Destroyed RTP instance '0x206a1f08' [2011-11-08 17:30:20] DEBUG[24824] res_rtp_asterisk.c: Got RTCP report of 76 bytes [2011-11-08 17:30:20] DEBUG[14661] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.15.184:2231 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 50165 SequenceNumberCycles: 0 IAJitter: 7 LastSR: 11258.198061858509291520 DLSR: 2.3800(sec) RTT: 3(sec) [2011-11-08 17:30:21] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981086 (6) REGISTER - 2 [2011-11-08 17:30:21] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #5981086)) [2011-11-08 17:30:21] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:21] DEBUG[2629] chan_sip.c: Header 0 [ 4]: ªªªª [2011-11-08 17:30:21] DEBUG[2629] chan_sip.c: Header 0 [ 4]: ªªªª [2011-11-08 17:30:22] DEBUG[2629] chan_sip.c: Header 0 [ 4]: ªªªª [2011-11-08 17:30:23] DEBUG[14661] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 192.168.15.184:2231 OurSSRC: 605060448 SentNTP: 1320791423.2102755328 SentRTP: 79840 SentPackets: 499 SentOctets: 79840 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0000 TheirLastSR: 737870480 DLSR: 2.6160 (sec) [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 1 [ 45]: o=- 3529780216 3529780217 IN IP4 10.20.60.148 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 2 [ 9]: s=cpc_med [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.20.60.148 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 5 [ 26]: m=audio 4000 RTP/AVP 0 101 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 7 [ 10]: a=sendrecv [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:24] DEBUG[28924] chan_sip.c: Invalid SIP message - rejected , no callid, len 772 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981086 (7) REGISTER - 2 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 8 to 4000 ms (t1 500 ms (Retrans id #5981086)) [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 1 [ 45]: o=- 3529780216 3529780217 IN IP4 10.20.60.148 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 2 [ 9]: s=cpc_med [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.20.60.148 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 5 [ 26]: m=audio 4000 RTP/AVP 0 101 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 7 [ 10]: a=sendrecv [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:25] DEBUG[28924] chan_sip.c: Invalid SIP message - rejected , no callid, len 772 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981086 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Stopping retransmission on '331802930ec37a493cdec7b51ccd9338@192.168.7.11' of Request 133: Match Found [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: SIP Registry 192.168.146.138: refcount now 2 [2011-11-08 17:30:25] DEBUG[2629] netsock2.c: Splitting '192.168.146.138' into... [2011-11-08 17:30:25] DEBUG[2629] netsock2.c: ...host '192.168.146.138' and port ''. [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for 331802930ec37a493cdec7b51ccd9338@192.168.7.11 - REGISTER (No RTP) [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: SIP Registry 192.168.146.138: refcount now 3 [2011-11-08 17:30:25] DEBUG[2629] acl.c: For destination '192.168.146.138', our source address is '192.168.7.11'. [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.7.11:5060 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: SIP Registry 192.168.146.138: refcount now 4 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Scheduled a registration timeout for 192.168.146.138 id #5981114 [2011-11-08 17:30:25] DEBUG[2629] netsock2.c: Splitting '192.168.146.138' into... [2011-11-08 17:30:25] DEBUG[2629] netsock2.c: ...host '192.168.146.138' and port ''. [2011-11-08 17:30:25] DEBUG[2629] netsock2.c: Splitting '192.168.146.138' into... [2011-11-08 17:30:25] DEBUG[2629] netsock2.c: ...host '192.168.146.138' and port ''. [2011-11-08 17:30:25] DEBUG[2629] netsock2.c: Splitting '192.168.146.138' into... [2011-11-08 17:30:25] DEBUG[2629] netsock2.c: ...host '192.168.146.138' and port ''. [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Initializing initreq for method REGISTER - callid 331802930ec37a493cdec7b51ccd9338@192.168.7.11 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 0 [ 36]: REGISTER sip:192.168.146.138 SIP/2.0 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.7.11:5060;branch=z9hG4bK6a62a3f8 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 3 [ 47]: From: ;tag=as53dc26a7 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 4 [ 30]: To: [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 5 [ 54]: Call-ID: 331802930ec37a493cdec7b51ccd9338@192.168.7.11 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 6 [ 18]: CSeq: 134 REGISTER [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 7 [ 24]: User-Agent: Asterisk PBX [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 8 [ 12]: Expires: 120 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Header 9 [ 34]: Contact: [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: REGISTER attempt 33 to 7193@192.168.146.138 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981115 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: SIP Registry 192.168.146.138: refcount now 3 [2011-11-08 17:30:25] DEBUG[14661] manager.c: Examining event: Event: Registry Privilege: system,all ChannelType: SIP Username: 7193 Domain: 192.168.146.138 Status: Request Sent [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: Destroying SIP dialog 331802930ec37a493cdec7b51ccd9338@192.168.7.11 [2011-11-08 17:30:25] DEBUG[2629] chan_sip.c: [2011-11-08 17:30:25] DEBUG[24824] res_rtp_asterisk.c: Got RTCP report of 76 bytes [2011-11-08 17:30:25] DEBUG[14661] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.15.184:2231 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 50415 SequenceNumberCycles: 0 IAJitter: 8 LastSR: 11263.198149822660739072 DLSR: 2.3800(sec) RTT: 3(sec) [2011-11-08 17:30:26] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981115 (1) REGISTER - 2 [2011-11-08 17:30:26] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #5981115)) [2011-11-08 17:30:26] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 1 [ 45]: o=- 3529780216 3529780217 IN IP4 10.20.60.148 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 2 [ 9]: s=cpc_med [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.20.60.148 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 5 [ 26]: m=audio 4000 RTP/AVP 0 101 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 7 [ 10]: a=sendrecv [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:26] DEBUG[28924] chan_sip.c: Invalid SIP message - rejected , no callid, len 772 [2011-11-08 17:30:27] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981115 (2) REGISTER - 2 [2011-11-08 17:30:27] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #5981115)) [2011-11-08 17:30:27] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:28] DEBUG[14661] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 192.168.15.184:2231 OurSSRC: 605060448 SentNTP: 1320791428.2102042624 SentRTP: 119840 SentPackets: 749 SentOctets: 119840 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0001 TheirLastSR: 738198160 DLSR: 2.6160 (sec) [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 1 [ 45]: o=- 3529780216 3529780217 IN IP4 10.20.60.148 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 2 [ 9]: s=cpc_med [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.20.60.148 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 5 [ 26]: m=audio 4000 RTP/AVP 0 101 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 7 [ 10]: a=sendrecv [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:28] DEBUG[28924] chan_sip.c: Invalid SIP message - rejected , no callid, len 772 [2011-11-08 17:30:29] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981115 (3) REGISTER - 2 [2011-11-08 17:30:29] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #5981115)) [2011-11-08 17:30:29] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 0 [ 38]: BYE sip:5321@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bKd5b9e586307D8269 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 2 [ 57]: From: "WT1" ;tag=E3B34B85-105452F2 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 3 [ 53]: To: ;tag=as4ae76676 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 4 [ 11]: CSeq: 2 BYE [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Initializing initreq for method BYE - callid bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:30] DEBUG[2629] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:30] DEBUG[2629] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Setting SIP_ALREADYGONE on dialog bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:30] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2aaacc025808' [2011-11-08 17:30:30] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: RTPAUDIOQOS Value: ssrc=605060448;themssrc=2075349923;lp=0;rxjitter=0.000169;rxcount=834;txjitter=0.000000;txcount=837;rlp=0;rtt=0.003000 Uniqueid: 1320791415.126037 [2011-11-08 17:30:30] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1320791415.126037 [2011-11-08 17:30:30] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1320791415.126037 [2011-11-08 17:30:30] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1320791415.126037 [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Received bye, issuing owner hangup [2011-11-08 17:30:30] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:30] DEBUG[24824] app_voicemail.c: Hang up during prefile playback [2011-11-08 17:30:30] DEBUG[24824] app_macro.c: Spawn extension (macro-stdexten,s-NOANSWER,1) exited non-zero on 'SIP/5261-0001cd99' in macro 'stdexten' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Spawn extension (DLPN_DialPlan1,5221,1) exited non-zero on 'SIP/5261-0001cd99' [2011-11-08 17:30:30] DEBUG[24824] channel.c: Soft-Hanging up channel 'SIP/5261-0001cd99' [2011-11-08 17:30:30] DEBUG[24824] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-11-08 17:30:30] DEBUG[24824] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-11-08 17:30:30] DEBUG[24824] channel.c: Hanging up channel 'SIP/5261-0001cd99' [2011-11-08 17:30:30] DEBUG[24824] chan_sip.c: Hangup call SIP/5261-0001cd99, SIP callid bbc7e461-d5bca2bb-bed7a630@192.168.15.184 [2011-11-08 17:30:30] DEBUG[24824] chan_sip.c: update_call_counter(5261) - decrement call limit counter on hangup [2011-11-08 17:30:30] DEBUG[24824] chan_sip.c: Updating call counter for incoming call [2011-11-08 17:30:30] DEBUG[24824] chan_sip.c: Call from peer '5261' removed from call limit 100 [2011-11-08 17:30:30] DEBUG[24824] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2aaacc025808' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '2011-11-08 17:30:19' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '"WT1" <5261>' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is 'DLPN_DialPlan1' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is 'SIP/5261-0001cd99' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is 'SIP/5321-0001cd9a' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is 'VoiceMail' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '5221,u' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '11' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '0' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is 'FAILED' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is 'DOCUMENTATION' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '(null)' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '1320791406.126035' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '(null)' [2011-11-08 17:30:30] DEBUG[24824] pbx.c: Function result is '(null)' [2011-11-08 17:30:30] DEBUG[24824] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2011-11-08 17:30:19','"WT1" <5261>','DLPN_DialPlan1','SIP/5261-0001cd99','SIP/5321-0001cd9a','VoiceMail','5221,u','11','0','FAILED','DOCUMENTATION','','1320791406.126035','','') [2011-11-08 17:30:30] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:30] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:30] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 1 (Not in use) [2011-11-08 17:30:30] DEBUG[2499] devicestate.c: device 'SIP/5261' state '1' [2011-11-08 17:30:30] DEBUG[2500] app_queue.c: Extension '5261@default' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:30] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:30] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: VMSTATUS Value: FAILED Uniqueid: 1320791415.126037 [2011-11-08 17:30:30] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd99 Variable: MACRO_DEPTH Value: 0 Uniqueid: 1320791415.126037 [2011-11-08 17:30:30] DEBUG[14661] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/5261-0001cd99 Uniqueid: 1320791415.126037 CallerIDNum: 5261 CallerIDName: WT1 ConnectedLineNum: 5221 ConnectedLineName: User1 Cause: 38 Cause-txt: Network out of order [2011-11-08 17:30:30] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5261 Context: default Hint: SIP/5261 Status: 0 [2011-11-08 17:30:30] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:30] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:30] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 1 (Not in use) [2011-11-08 17:30:30] DEBUG[2499] devicestate.c: device 'SIP/5261' state '1' [2011-11-08 17:30:30] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 0 [ 52]: BYE sip:5321@192.168.7.11:5060;transport=TCP SIP/2.0 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 1 [ 95]: Via: SIP/2.0/TCP 192.168.7.2:57144;rport;branch=z9hG4bKPjJaPZJKODhStqAoBDmSlfLpYG6xx0Ytf6;alias [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 3 [ 65]: From: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 4 [ 54]: To: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 5 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 6 [ 15]: CSeq: 17840 BYE [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 7 [ 29]: User-Agent: Bria iPhone 1.3.4 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 8 [ 18]: Content-Length: 0 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 9 [ 0]: [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 0 [ 52]: BYE sip:5321@192.168.7.11:5060;transport=TCP SIP/2.0 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 1 [ 95]: Via: SIP/2.0/TCP 192.168.7.2:57144;rport;branch=z9hG4bKPjJaPZJKODhStqAoBDmSlfLpYG6xx0Ytf6;alias [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 3 [ 65]: From: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 4 [ 54]: To: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 5 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 6 [ 15]: CSeq: 17840 BYE [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 7 [ 29]: User-Agent: Bria iPhone 1.3.4 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 8 [ 18]: Content-Length: 0 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Header 9 [ 0]: [2011-11-08 17:30:31] DEBUG[28924] acl.c: For destination '192.168.7.2', our source address is '192.168.7.11'. [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Setting SIP_TRANSPORT_TCP with address 192.168.7.11:5060 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Trying to put 'SIP/2.0 481' onto TCP socket destined for 192.168.7.2:57144 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: That's odd... Got a request in unknown dialog. Callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:31] DEBUG[28924] chan_sip.c: Invalid SIP message - rejected , no callid, len 423 [2011-11-08 17:30:33] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981115 (4) REGISTER - 2 [2011-11-08 17:30:33] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #5981115)) [2011-11-08 17:30:33] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 0 [ 52]: INVITE sip:5221@192.168.15.251:5060;user=phone SIP/2.0 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bKe2ad7983634A88DA [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 2 [ 57]: From: "WT1" ;tag=39B1AF71-80E690EE [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 3 [ 38]: To: [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 10 [ 26]: Supported: 100rel,replaces [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 11 [ 34]: Allow-Events: talk,hold,conference [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 14 [ 19]: Content-Length: 270 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Header 15 [ 0]: [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 1 [ 45]: o=- 1320791416 1320791416 IN IP4 192.168.15.184 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.184 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 5 [ 10]: a=sendrecv [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 6 [ 31]: m=audio 2232 RTP/AVP 0 8 18 101 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 10 [ 19]: a=fmtp:18 annexb=no [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:34] DEBUG[2629] acl.c: For destination '192.168.15.184', our source address is '192.168.15.251'. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.251:5060 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 - INVITE (No RTP) [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2011-11-08 17:30:34] DEBUG[2629] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,replaces" [2011-11-08 17:30:34] DEBUG[2629] sip/reqresp_parser.c: Found SIP option: -100rel- [2011-11-08 17:30:34] DEBUG[2629] sip/reqresp_parser.c: Matched SIP option: 100rel [2011-11-08 17:30:34] DEBUG[2629] sip/reqresp_parser.c: Found SIP option: -replaces- [2011-11-08 17:30:34] DEBUG[2629] sip/reqresp_parser.c: Matched SIP option: replaces [2011-11-08 17:30:34] DEBUG[2629] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:34] DEBUG[2629] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Initializing initreq for method INVITE - callid fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x206a1f08' [2011-11-08 17:30:34] DEBUG[2629] res_rtp_asterisk.c: Allocated port 15898 for RTP instance '0x206a1f08' [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: RTP instance '0x206a1f08' is setup and ready to go [2011-11-08 17:30:34] DEBUG[2629] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x206a1f08' [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Setting NAT on RTP to On [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing session-level SDP o=- 1320791416 1320791416 IN IP4 192.168.15.184... UNSUPPORTED. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [2011-11-08 17:30:34] DEBUG[2629] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:34] DEBUG[2629] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.15.184... OK. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Setting payload 0 based on m type on 0x40299570 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Setting payload 8 based on m type on 0x40299570 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Setting payload 18 based on m type on 0x40299570 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Setting payload 101 based on m type on 0x40299570 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Incorporating payload 0 on 0x40299570 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Incorporating payload 8 on 0x40299570 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Incorporating payload 18 on 0x40299570 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Incorporating payload 101 on 0x40299570 [2011-11-08 17:30:34] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x206a1f08' [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Copying payload 0 from 0x40299570 to 0x206a20d0 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Copying payload 8 from 0x40299570 to 0x206a20d0 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Copying payload 18 from 0x40299570 to 0x206a20d0 [2011-11-08 17:30:34] DEBUG[2629] rtp_engine.c: Copying payload 101 from 0x40299570 to 0x206a20d0 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Checking SIP call limits for device 5261 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Updating call counter for incoming call [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Call from peer '5261' is 1 out of 100 [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:34] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 2 (In use) [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: *** Our native formats are 0x4 (ulaw) [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: *** Our capabilities are 0x80e (gsm|ulaw|alaw|g726) [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: This channel will not be able to handle video. [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: device 'SIP/5261' state '2' [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/5261-0001cd9d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 5261 CallerIDName: WT1 AccountCode: Exten: 5221 Context: DLPN_LUSI_Internet_Unrestricted Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: SIPURI Value: sip:5261@192.168.15.184 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5261 Context: default Hint: SIP/5261 Status: 1 [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: build_route: Contact hop: [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: SIP/5261-0001cd9d: New call is still down.... Trying... [2011-11-08 17:30:34] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:34] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 2 (In use) [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: device 'SIP/5261' state '2' [2011-11-08 17:30:34] DEBUG[2500] app_queue.c: Extension '5261@default' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:34] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Result of 'HINT' is 'SIP/5221' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Launching 'Macro' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Result of 'FEATURES' is '' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Launching 'Set' [2011-11-08 17:30:34] DEBUG[24825] app_macro.c: Executed application: Set [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Result of 'ARG1' is '5221' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Launching 'Set' [2011-11-08 17:30:34] DEBUG[24825] app_macro.c: Executed application: Set [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Result of 'ARG1' is '5221' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Result of 'FOLLOWME_5221' is '0' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Expression result is '0' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Launching 'GotoIf' [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: SIPDOMAIN Value: 192.168.15.251:5060 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: SIPCALLID Value: fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5261-0001cd9d Uniqueid: 1320791434.126039 Channeltype: SIP SIPcallid: fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 SIPfullcontact: sip:5261@192.168.15.184 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5261-0001cd9d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 5261 CallerIDName: WT1 ConnectedLineNum: ConnectedLineName: Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd9d Context: DLPN_LUSI_Internet_Unrestricted Extension: 5221 Priority: 1 Application: Macro AppData: stdexten,5221,SIP/5221 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_EXTEN Value: 5221 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_CONTEXT Value: DLPN_LUSI_Internet_Unrestricted Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_PRIORITY Value: 1 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: ARG1 Value: 5221 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: ARG2 Value: SIP/5221 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd9d Context: macro-stdexten Extension: s Priority: 1 Application: Set AppData: __DYNAMIC_FEATURES= Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: __DYNAMIC_FEATURES Value: Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd9d Context: macro-stdexten Extension: s Priority: 2 Application: Set AppData: ORIG_ARG1=5221 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: ORIG_ARG1 Value: 5221 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd9d Context: macro-stdexten Extension: s Priority: 3 Application: GotoIf AppData: 0?6:4 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[24825] app_macro.c: Executed application: GotoIf [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Result of 'ARG2' is 'SIP/5221' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Result of 'RINGTIME' is '30' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Result of 'DIALOPTIONS' is '' [2011-11-08 17:30:34] DEBUG[24825] pbx.c: Launching 'Dial' [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Allocating new SIP dialog for 619879486aaf06e1100920a4026a0922@192.168.7.11:0 - INVITE (No RTP) [2011-11-08 17:30:34] DEBUG[24825] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2aaacc0020f8' [2011-11-08 17:30:34] DEBUG[24825] res_rtp_asterisk.c: Allocated port 10266 for RTP instance '0x2aaacc0020f8' [2011-11-08 17:30:34] DEBUG[24825] rtp_engine.c: RTP instance '0x2aaacc0020f8' is setup and ready to go [2011-11-08 17:30:34] DEBUG[24825] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x2aaacc0020f8' [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Setting NAT on RTP to On [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2011-11-08 17:30:34] DEBUG[24825] acl.c: For destination '192.168.7.2', our source address is '192.168.7.11'. [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Setting SIP_TRANSPORT_TCP with address 192.168.7.11:5060 [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: *** Our native formats are 0x4 (ulaw) [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: *** Our capabilities are 0x80e (gsm|ulaw|alaw|g726) [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: This channel will not be able to handle video. [2011-11-08 17:30:34] DEBUG[24825] rtp_engine.c: Seeded SDP of 'SIP/5221-0001cd9e' with that of 'SIP/5261-0001cd9d' [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable DIALEDTIME. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable ANSWEREDTIME. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable DIALEDPEERNAME. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable DIALEDPEERNUMBER. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable DIALSTATUS. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable MACRO_DEPTH. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable ORIG_ARG1. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Copying hard-transferable variable DYNAMIC_FEATURES. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable ARG2. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable ARG1. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable MACRO_PRIORITY. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable MACRO_CONTEXT. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable MACRO_EXTEN. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable SIPCALLID. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable SIPDOMAIN. [2011-11-08 17:30:34] DEBUG[24825] channel.c: Not copying variable SIPURI. [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Outgoing Call for 5221 [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Call to peer '5221' is 1 out of 100 [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:34] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 6 (Ringing) [2011-11-08 17:30:34] DEBUG[2499] devicestate.c: device 'SIP/5221' state '6' [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: ** Our capability: 0x80e (gsm|ulaw|alaw|g726) Video flag: False Text flag: False [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: -- Done with adding codecs to SDP [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Done building SDP. Settling with this capability: 0x80e (gsm|ulaw|alaw|g726) [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Initializing initreq for method INVITE - callid 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 0 [ 55]: INVITE sip:5221@192.168.7.2:57144;transport=tcp SIP/2.0 [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/TCP 192.168.7.11:5060;branch=z9hG4bK0c0e5a71;rport [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 4 [ 46]: To: [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 5 [ 51]: Contact: [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 6 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 9 [ 35]: Date: Tue, 08 Nov 2011 22:30:34 GMT [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 12 [ 82]: Remote-Party-ID: "WT1" ;party=calling;privacy=off;screen=no [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:34] DEBUG[24825] chan_sip.c: Trying to put 'INVITE sip:' onto TCP socket destined for 192.168.7.2:57144 [2011-11-08 17:30:34] DEBUG[2500] app_queue.c: Extension '5221@default' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [2011-11-08 17:30:34] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_DEPTH Value: 1 Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/5261-0001cd9d Context: macro-stdexten Extension: s Priority: 4 Application: Dial AppData: SIP/5221,30, Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALSTATUS Value: Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALEDPEERNUMBER Value: Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALEDPEERNAME Value: Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: ANSWEREDTIME Value: Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALEDTIME Value: Uniqueid: 1320791434.126039 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/5221-0001cd9e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 5221 CallerIDName: User1 AccountCode: Exten: Context: DLPN_LUSI_Internet_Unrestricted Uniqueid: 1320791434.126040 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: SIPCALLID Value: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 Uniqueid: 1320791434.126040 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5221-0001cd9e Uniqueid: 1320791434.126040 Channeltype: SIP SIPcallid: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 SIPfullcontact: sip:5221@192.168.7.2:57144;transport=tcp [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5221-0001cd9e Channeltype: SIP SIPcallid: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 SIPfullcontact: sip:5221@192.168.7.2:57144;transport=tcp Peername: 5221 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: DIALEDPEERNUMBER Value: 5221 Uniqueid: 1320791434.126040 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/5261-0001cd9d Destination: SIP/5221-0001cd9e CallerIDNum: 5261 CallerIDName: WT1 ConnectedLineNum: ConnectedLineName: UniqueID: 1320791434.126039 DestUniqueID: 1320791434.126040 Dialstring: 5221 [2011-11-08 17:30:34] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5221 Context: default Hint: SIP/5221 Status: 8 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK0c0e5a71;alias [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 4 [ 26]: To: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 7 [ 0]: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK0c0e5a71;alias [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 4 [ 26]: To: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 7 [ 0]: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: SIP response 100 to standard invite [2011-11-08 17:30:36] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: ~HASH~SIP_CAUSE~SIP/5221-0001cd9e~ Value: SIP 100 Trying Uniqueid: 1320791434.126039 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 1 [ 45]: o=- 3529780216 3529780217 IN IP4 10.20.60.148 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 2 [ 9]: s=cpc_med [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.20.60.148 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 5 [ 26]: m=audio 4000 RTP/AVP 0 101 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 7 [ 10]: a=sendrecv [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Invalid SIP message - rejected , no callid, len 772 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK0c0e5a71;alias [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=oieJUR3CXljRoUw2cSpTRyEP..lRuw6Y [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 6 [ 51]: Contact: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 8 [ 18]: Content-Length: 0 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 9 [ 0]: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK0c0e5a71;alias [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=oieJUR3CXljRoUw2cSpTRyEP..lRuw6Y [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 6 [ 51]: Contact: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 8 [ 18]: Content-Length: 0 [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: Header 9 [ 0]: [2011-11-08 17:30:36] DEBUG[28924] chan_sip.c: SIP response 180 to standard invite [2011-11-08 17:30:36] DEBUG[24825] rtp_engine.c: Setting early bridge SDP of 'SIP/5261-0001cd9d' with that of 'SIP/5221-0001cd9e' [2011-11-08 17:30:36] DEBUG[24825] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:36] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:36] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:36] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 6 (Ringing) [2011-11-08 17:30:36] DEBUG[2499] devicestate.c: device 'SIP/5221' state '6' [2011-11-08 17:30:36] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [2011-11-08 17:30:36] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5221-0001cd9e ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 5221 CallerIDName: User1 ConnectedLineNum: 5261 ConnectedLineName: WT1 Uniqueid: 1320791434.126040 [2011-11-08 17:30:36] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: ~HASH~SIP_CAUSE~SIP/5221-0001cd9e~ Value: SIP 180 Ringing Uniqueid: 1320791434.126039 [2011-11-08 17:30:36] DEBUG[2629] chan_sip.c: Re-scheduled destruction of SIP call 74f62b6c7555dd580f7b442e1053ebaa@192.168.15.251:5060 [2011-11-08 17:30:36] DEBUG[2629] chan_sip.c: Re-scheduled destruction of SIP call 5780df407c59a24a5205e9d8279165c2@192.168.15.251:5060 [2011-11-08 17:30:37] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981115 (5) REGISTER - 2 [2011-11-08 17:30:37] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #5981115)) [2011-11-08 17:30:37] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.7.2:38300 [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: Got RTCP report of 32 bytes [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2aaacc0020f8' [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: RTP NAT: Got audio from other end. Now sending to address 192.168.7.2:54677 [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.7.2:38300 [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: Got RTCP report of 32 bytes [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.7.2:38300 PT: 201(Receiver Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 0 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK0c0e5a71;alias [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=oieJUR3CXljRoUw2cSpTRyEP..lRuw6Y [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK0c0e5a71;alias [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=oieJUR3CXljRoUw2cSpTRyEP..lRuw6Y [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 1 [ 45]: o=- 3529780235 3529780236 IN IP4 10.20.60.148 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 2 [ 9]: s=cpc_med [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.20.60.148 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 5 [ 26]: m=audio 4002 RTP/AVP 0 101 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 7 [ 10]: a=sendrecv [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: SIP response 200 to standard invite [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing session-level SDP o=- 3529780235 3529780236 IN IP4 10.20.60.148... UNSUPPORTED. [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing session-level SDP s=cpc_med... UNSUPPORTED. [2011-11-08 17:30:39] DEBUG[28924] netsock2.c: Splitting '10.20.60.148' into... [2011-11-08 17:30:39] DEBUG[28924] netsock2.c: ...host '10.20.60.148' and port ''. [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing session-level SDP c=IN IP4 10.20.60.148... OK. [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-11-08 17:30:39] DEBUG[28924] rtp_engine.c: Setting payload 0 based on m type on 0x432c3ca0 [2011-11-08 17:30:39] DEBUG[28924] rtp_engine.c: Setting payload 101 based on m type on 0x432c3ca0 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [2011-11-08 17:30:39] DEBUG[28924] rtp_engine.c: Incorporating payload 0 on 0x432c3ca0 [2011-11-08 17:30:39] DEBUG[28924] rtp_engine.c: Incorporating payload 101 on 0x432c3ca0 [2011-11-08 17:30:39] DEBUG[28924] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2aaacc0020f8' [2011-11-08 17:30:39] DEBUG[28924] rtp_engine.c: Copying payload 0 from 0x432c3ca0 to 0x2aaacc0022c0 [2011-11-08 17:30:39] DEBUG[28924] rtp_engine.c: Copying payload 101 from 0x432c3ca0 to 0x2aaacc0022c0 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: We have an owner, now see if we need to change this call [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: build_route: Contact hop: [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:39] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 2 (In use) [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: device 'SIP/5221' state '2' [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: ChannelUpdate Privilege: system,all Channel: SIP/5221-0001cd9e Channeltype: SIP Uniqueid: 1320791434.126040 SIPcallid: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 SIPfullcontact: sip:5221@192.168.7.2:57144;transport=tcp Peername: 5221 [2011-11-08 17:30:39] DEBUG[2500] app_queue.c: Extension '5221@default' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:39] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5221 Context: default Hint: SIP/5221 Status: 1 [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Strict routing enforced for session 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:39] DEBUG[28924] netsock2.c: Splitting '192.168.7.2:57144' into... [2011-11-08 17:30:39] DEBUG[28924] netsock2.c: ...host '192.168.7.2' and port '57144'. [2011-11-08 17:30:39] DEBUG[28924] chan_sip.c: Trying to put 'ACK sip:522' onto TCP socket destined for 192.168.7.2:57144 [2011-11-08 17:30:39] DEBUG[24825] rtp_engine.c: Setting early bridge SDP of 'SIP/5261-0001cd9d' with that of 'SIP/5221-0001cd9e' [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:39] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 2 (In use) [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: device 'SIP/5221' state '2' [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:39] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 2 (In use) [2011-11-08 17:30:39] DEBUG[2499] devicestate.c: device 'SIP/5261' state '2' [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5221-0001cd9e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 5221 CallerIDName: User1 ConnectedLineNum: 5261 ConnectedLineName: WT1 Uniqueid: 1320791434.126040 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALSTATUS Value: ANSWER Uniqueid: 1320791434.126039 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALEDPEERNAME Value: SIP/5221-0001cd9e Uniqueid: 1320791434.126039 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALEDPEERNUMBER Value: 5221 Uniqueid: 1320791434.126039 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: BRIDGEPEER Value: SIP/5221-0001cd9e Uniqueid: 1320791434.126039 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: BRIDGEPEER Value: SIP/5261-0001cd9d Uniqueid: 1320791434.126040 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/5261-0001cd9d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 5261 CallerIDName: WT1 ConnectedLineNum: 5221 ConnectedLineName: User1 Uniqueid: 1320791434.126039 [2011-11-08 17:30:39] DEBUG[24825] chan_sip.c: SIP answering channel: SIP/5261-0001cd9d [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:39] DEBUG[24825] chan_sip.c: Setting framing from config on incoming call [2011-11-08 17:30:39] DEBUG[24825] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [2011-11-08 17:30:39] DEBUG[24825] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2011-11-08 17:30:39] DEBUG[24825] chan_sip.c: -- Done with adding codecs to SDP [2011-11-08 17:30:39] DEBUG[24825] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [2011-11-08 17:30:39] DEBUG[24825] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981121 [2011-11-08 17:30:39] DEBUG[24825] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:39] DEBUG[24825] features.c: bridge answer set, chan answer set [2011-11-08 17:30:39] DEBUG[24825] features.c: Removing dialed interfaces datastore on SIP/5221-0001cd9e since we're bridging [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:39] DEBUG[24825] res_rtp_asterisk.c: Setting the marker bit due to a source update [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: ~HASH~SIP_CAUSE~SIP/5221-0001cd9e~ Value: SIP 200 OK Uniqueid: 1320791434.126039 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/5221-0001cd9e Uniqueid: 1320791434.126040 AccountCode: OldAccountCode: [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/5261-0001cd9d Channel2: SIP/5221-0001cd9e Uniqueid1: 1320791434.126039 Uniqueid2: 1320791434.126040 CallerID1: 5261 CallerID2: 5221 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: BRIDGEPEER Value: SIP/5221-0001cd9e Uniqueid: 1320791434.126039 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: BRIDGEPVTCALLID Value: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 Uniqueid: 1320791434.126039 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: BRIDGEPEER Value: SIP/5261-0001cd9d Uniqueid: 1320791434.126040 [2011-11-08 17:30:39] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: BRIDGEPVTCALLID Value: fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 Uniqueid: 1320791434.126040 [2011-11-08 17:30:39] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:39] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 0 [ 38]: ACK sip:5221@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bKef196580637D474B [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 2 [ 57]: From: "WT1" ;tag=39B1AF71-80E690EE [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 3 [ 53]: To: ;tag=as1f0e8f3a [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134 [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 9 [ 19]: Accept-Language: en [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Header 12 [ 0]: [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981121 [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Stopping retransmission on 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' of Response 1: Match Found [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' Method: INVITE [2011-11-08 17:30:39] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' Method: ACK [2011-11-08 17:30:40] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' Method: INVITE [2011-11-08 17:30:40] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' Method: ACK [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 1 [ 45]: o=- 3529780216 3529780217 IN IP4 10.20.60.148 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 2 [ 9]: s=cpc_med [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.20.60.148 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 5 [ 26]: m=audio 4000 RTP/AVP 0 101 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 7 [ 10]: a=sendrecv [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:40] DEBUG[28924] chan_sip.c: Invalid SIP message - rejected , no callid, len 772 [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981115 (6) REGISTER - 2 [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #5981115)) [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' Method: INVITE [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' Method: ACK [2011-11-08 17:30:41] DEBUG[24825] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2aaacc0020f8' [2011-11-08 17:30:41] DEBUG[24825] res_rtp_asterisk.c: RTP NAT: Got audio from other end. Now sending to address 192.168.7.2:54677 [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: Auto destroying SIP dialog '98cce35e-c38be244-dc5be10f@192.168.15.149' [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: Destroying SIP dialog 98cce35e-c38be244-dc5be10f@192.168.15.149 [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: ---------- SIP HISTORY for '98cce35e-c38be244-dc5be10f@192.168.15.149' [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: * SIP Call [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: 001. AuthChal Auth challenge sent for - nc 0 [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: 002. RegRequest Succeeded : Account [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: 003. AuthOK Auth challenge successful for 5223 [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: 004. CancelDestroy [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: 005. RegRequest Succeeded : Account [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: 006. AutoDestroy 98cce35e-c38be244-dc5be10f@192.168.15.149 [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: ---------- END SIP HISTORY for '98cce35e-c38be244-dc5be10f@192.168.15.149' [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' Method: INVITE [2011-11-08 17:30:41] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' Method: ACK [2011-11-08 17:30:41] DEBUG[24825] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.7.2:38300 [2011-11-08 17:30:41] DEBUG[24825] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2011-11-08 17:30:41] DEBUG[14661] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.7.2:38300 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 27302 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [2011-11-08 17:30:42] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' Method: INVITE [2011-11-08 17:30:42] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' Method: ACK [2011-11-08 17:30:43] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' Method: INVITE [2011-11-08 17:30:43] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' Method: ACK [2011-11-08 17:30:43] DEBUG[24825] res_rtp_asterisk.c: Got RTCP report of 76 bytes [2011-11-08 17:30:43] DEBUG[14661] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.15.184:2233 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 19965 SequenceNumberCycles: 0 IAJitter: 24 LastSR: 0.0000000000 DLSR: 0.0000(sec) [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' Method: INVITE [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' Method: ACK [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK453897c6;alias [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 3 [ 56]: From: "User1" ;tag=as6dc324d4 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=Yn6.U-TiwEOIP9bOOaAm81QBixFBQMdz [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 7 [ 51]: Contact: [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 10 [ 21]: Content-Length: 207 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 0 [ 3]: v=0 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 1 [ 45]: o=- 3529780216 3529780217 IN IP4 10.20.60.148 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 2 [ 9]: s=cpc_med [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.20.60.148 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 4 [ 5]: t=0 0 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 5 [ 26]: m=audio 4000 RTP/AVP 0 101 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 7 [ 10]: a=sendrecv [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: That's odd... Got a response on a call we don't know about. Callid 72b12deb77477be446e585d404781e14@192.168.7.11:5060 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Invalid SIP message - rejected , no callid, len 772 [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 0 [ 38]: BYE sip:5221@192.168.15.251:5060 SIP/2.0 [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.15.184;branch=z9hG4bK88311c2111AAD95 [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 2 [ 57]: From: "WT1" ;tag=39B1AF71-80E690EE [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 3 [ 53]: To: ;tag=as1f0e8f3a [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 4 [ 11]: CSeq: 2 BYE [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 5 [ 48]: Call-ID: fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 6 [ 32]: Contact: [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.7.0134 [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 8 [ 19]: Accept-Language: en [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Header 11 [ 0]: [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Initializing initreq for method BYE - callid fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 [2011-11-08 17:30:44] DEBUG[2629] netsock2.c: Splitting '192.168.15.184' into... [2011-11-08 17:30:44] DEBUG[2629] netsock2.c: ...host '192.168.15.184' and port ''. [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Setting SIP_ALREADYGONE on dialog fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 [2011-11-08 17:30:44] DEBUG[2629] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x206a1f08' [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Received bye, issuing owner hangup [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.184:5060 [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' Method: INVITE [2011-11-08 17:30:44] DEBUG[2629] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'fa9e6d0d-cfbdb787-2c40978c@192.168.15.184' Method: BYE [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: RTPAUDIOQOS Value: ssrc=689119580;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSBRIDGED Value: ssrc=689119580;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOS Value: ssrc=884217990;themssrc=871442030;lp=0;rxjitter=0.000000;rxcount=1;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: RTPAUDIOQOSBRIDGED Value: ssrc=884217990;themssrc=871442030;lp=0;rxjitter=0.000000;rxcount=1;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[24825] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a hangup [2011-11-08 17:30:44] DEBUG[24825] channel.c: Returning from native bridge, channels: SIP/5261-0001cd9d, SIP/5221-0001cd9e [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is '2011-11-08 17:30:34' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is '"WT1" <5261>' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is 'DLPN_LUSI_Internet_Unrestricted' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is 'SIP/5261-0001cd9d' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is 'SIP/5221-0001cd9e' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is 'Dial' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is 'SIP/5221,30,' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is '10' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is '5' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is 'ANSWERED' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is 'DOCUMENTATION' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is '(null)' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is '1320791434.126039' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is '(null)' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Function result is '(null)' [2011-11-08 17:30:44] DEBUG[24825] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2011-11-08 17:30:34','"WT1" <5261>','DLPN_LUSI_Internet_Unrestricted','SIP/5261-0001cd9d','SIP/5221-0001cd9e','Dial','SIP/5221,30,','10','5','ANSWERED','DOCUMENTATION','','1320791434.126039','','') [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/5261-0001cd9d Channel2: SIP/5221-0001cd9e Uniqueid1: 1320791434.126039 Uniqueid2: 1320791434.126040 CallerID1: 5261 CallerID2: 5221 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: ANSWEREDTIME Value: 5 Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALEDTIME Value: 10 Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[24825] channel.c: Hanging up channel 'SIP/5221-0001cd9e' [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: Hangup call SIP/5221-0001cd9e, SIP callid 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: update_call_counter(5221) - decrement call limit counter on hangup [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: Call to peer '5221' removed from call limit 100 [2011-11-08 17:30:44] DEBUG[24825] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2aaacc0020f8' [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:44] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 1 (Not in use) [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: device 'SIP/5221' state '1' [2011-11-08 17:30:44] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOS Value: ssrc=884217990;themssrc=871442030;lp=0;rxjitter=0.000000;rxcount=1;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5221 Context: default Hint: SIP/5221 Status: 0 [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: Strict routing enforced for session 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:44] DEBUG[24825] netsock2.c: Splitting '192.168.7.2:57144' into... [2011-11-08 17:30:44] DEBUG[24825] netsock2.c: ...host '192.168.7.2' and port '57144'. [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: Trying to put 'BYE sip:522' onto TCP socket destined for 192.168.7.2:57144 [2011-11-08 17:30:44] DEBUG[24825] app_dial.c: Exiting with DIALSTATUS=ANSWER. [2011-11-08 17:30:44] DEBUG[24825] app_macro.c: Spawn extension (macro-stdexten,s,4) exited non-zero on 'SIP/5261-0001cd9d' in macro 'stdexten' [2011-11-08 17:30:44] DEBUG[24825] pbx.c: Spawn extension (DLPN_LUSI_Internet_Unrestricted,5221,1) exited non-zero on 'SIP/5261-0001cd9d' [2011-11-08 17:30:44] DEBUG[24825] channel.c: Soft-Hanging up channel 'SIP/5261-0001cd9d' [2011-11-08 17:30:44] DEBUG[24825] channel.c: Hanging up channel 'SIP/5261-0001cd9d' [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: Hangup call SIP/5261-0001cd9d, SIP callid fa9e6d0d-cfbdb787-2c40978c@192.168.15.184 [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: update_call_counter(5261) - decrement call limit counter on hangup [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: Updating call counter for incoming call [2011-11-08 17:30:44] DEBUG[24825] chan_sip.c: Call from peer '5261' removed from call limit 100 [2011-11-08 17:30:44] DEBUG[24825] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x206a1f08' [2011-11-08 17:30:44] DEBUG[2500] app_queue.c: Extension '5221@default' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5221 [2011-11-08 17:30:44] DEBUG[2499] chan_sip.c: Checking device state for peer 5221 [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: Changing state for SIP/5221 - state 1 (Not in use) [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: device 'SIP/5221' state '1' [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:44] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 1 (Not in use) [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: device 'SIP/5261' state '1' [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5261 [2011-11-08 17:30:44] DEBUG[2499] chan_sip.c: Checking device state for peer 5261 [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: Changing state for SIP/5261 - state 1 (Not in use) [2011-11-08 17:30:44] DEBUG[2499] devicestate.c: device 'SIP/5261' state '1' [2011-11-08 17:30:44] DEBUG[2500] app_queue.c: Extension '5261@default' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:44] DEBUG[2756] app_queue.c: Device 'SIP/5221' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:44] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:44] DEBUG[2756] app_queue.c: Device 'SIP/5261' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5221-0001cd9e Variable: RTPAUDIOQOS Value: ssrc=884217990;themssrc=871442030;lp=0;rxjitter=0.000000;rxcount=1;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 Uniqueid: 1320791434.126040 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/5221-0001cd9e Uniqueid: 1320791434.126040 CallerIDNum: 5221 CallerIDName: User1 ConnectedLineNum: 5261 ConnectedLineName: WT1 Cause: 16 Cause-txt: Normal Clearing [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: DIALSTATUS Value: ANSWER Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/5261-0001cd9d UniqueID: 1320791434.126039 DialStatus: ANSWER [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/5261-0001cd9d Variable: MACRO_DEPTH Value: 0 Uniqueid: 1320791434.126039 [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/5261-0001cd9d Uniqueid: 1320791434.126039 CallerIDNum: 5261 CallerIDName: WT1 ConnectedLineNum: 5221 ConnectedLineName: User1 Cause: 16 Cause-txt: Normal Clearing [2011-11-08 17:30:44] DEBUG[14661] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 5261 Context: default Hint: SIP/5261 Status: 0 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK14506ad5;alias [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=oieJUR3CXljRoUw2cSpTRyEP..lRuw6Y [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 7 [ 0]: [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TCP 192.168.7.11:5060;rport=5060;received=192.168.7.11;branch=z9hG4bK14506ad5;alias [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 2 [ 59]: Call-ID: 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 3 [ 50]: From: "WT1" ;tag=as31980058 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 4 [ 63]: To: ;tag=oieJUR3CXljRoUw2cSpTRyEP..lRuw6Y [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [2011-11-08 17:30:44] DEBUG[28924] chan_sip.c: Header 7 [ 0]: [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: SIP TIMER: Rescheduling retransmission #5981115 (7) REGISTER - 2 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: ** SIP timers: Rescheduling retransmission 8 to 4000 ms (t1 500 ms (Retrans id #5981115)) [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Destroying SIP dialog 637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: ---------- SIP HISTORY for '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: * SIP Call [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: 001. Hangup Cause Normal Clearing [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: 002. NeedDestroy Setting needdestroy because received 200 response [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: ---------- END SIP HISTORY for '637f3761029135cc44a23bb207a2e1ed@192.168.7.11:5060' [2011-11-08 17:30:45] DEBUG[2629] rtp_engine.c: Destroyed RTP instance '0x2aaacc0020f8' [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5981115 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Stopping retransmission on '331802930ec37a493cdec7b51ccd9338@192.168.7.11' of Request 134: Match Found [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: SIP Registry 192.168.146.138: refcount now 2 [2011-11-08 17:30:45] DEBUG[2629] netsock2.c: Splitting '192.168.146.138' into... [2011-11-08 17:30:45] DEBUG[2629] netsock2.c: ...host '192.168.146.138' and port ''. [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for 331802930ec37a493cdec7b51ccd9338@192.168.7.11 - REGISTER (No RTP) [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: SIP Registry 192.168.146.138: refcount now 3 [2011-11-08 17:30:45] DEBUG[2629] acl.c: For destination '192.168.146.138', our source address is '192.168.7.11'. [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.7.11:5060 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: SIP Registry 192.168.146.138: refcount now 4 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Scheduled a registration timeout for 192.168.146.138 id #5981124 [2011-11-08 17:30:45] DEBUG[2629] netsock2.c: Splitting '192.168.146.138' into... [2011-11-08 17:30:45] DEBUG[2629] netsock2.c: ...host '192.168.146.138' and port ''. [2011-11-08 17:30:45] DEBUG[2629] netsock2.c: Splitting '192.168.146.138' into... [2011-11-08 17:30:45] DEBUG[2629] netsock2.c: ...host '192.168.146.138' and port ''. [2011-11-08 17:30:45] DEBUG[2629] netsock2.c: Splitting '192.168.146.138' into... [2011-11-08 17:30:45] DEBUG[2629] netsock2.c: ...host '192.168.146.138' and port ''. [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Initializing initreq for method REGISTER - callid 331802930ec37a493cdec7b51ccd9338@192.168.7.11 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 0 [ 36]: REGISTER sip:192.168.146.138 SIP/2.0 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.7.11:5060;branch=z9hG4bK3817564a [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 3 [ 47]: From: ;tag=as29ee9456 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 4 [ 30]: To: [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 5 [ 54]: Call-ID: 331802930ec37a493cdec7b51ccd9338@192.168.7.11 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 6 [ 18]: CSeq: 135 REGISTER [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 7 [ 24]: User-Agent: Asterisk PBX [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 8 [ 12]: Expires: 120 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Header 9 [ 34]: Contact: [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: REGISTER attempt 34 to 7193@192.168.146.138 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5981125 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 192.168.146.138:5060 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: SIP Registry 192.168.146.138: refcount now 3 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: Destroying SIP dialog 331802930ec37a493cdec7b51ccd9338@192.168.7.11 [2011-11-08 17:30:45] DEBUG[2629] chan_sip.c: [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: Auto destroying SIP dialog '55659242-cbacca8-e5c8eb75@192.168.15.187' [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: Destroying SIP dialog 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: ---------- SIP HISTORY for '55659242-cbacca8-e5c8eb75@192.168.15.187' [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: * SIP Call [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 001. AuthChal Auth challenge sent for - nc 0 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 002. CancelDestroy [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 003. Invite New call: 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 004. AuthOK Auth challenge successful for 5321 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 005. ConnectedLine Called party is now User1 <5221> [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 006. Masq Old channel: SIP/5261-0001cd99 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 007. Masq (cont) ...new owner: SIP/5261-0001cd99 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 008. Cancel Cause Normal Clearing [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 009. AutoDestroy 55659242-cbacca8-e5c8eb75@192.168.15.187 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: ---------- END SIP HISTORY for '55659242-cbacca8-e5c8eb75@192.168.15.187' [2011-11-08 17:30:51] DEBUG[2629] rtp_engine.c: Destroyed RTP instance '0x20687428' [2011-11-08 17:30:51] DEBUG[2629] rtp_engine.c: Destroyed RTP instance '0x20638958' [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: Auto destroying SIP dialog '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: Destroying SIP dialog 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: Updating call counter for outgoing call [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: This call did not properly clean up call limits. Call ID 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: ---------- SIP HISTORY for '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: * SIP Call [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 001. Hold INVITE [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 002. Xfer Refer accepted [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 003. Xfer Refer succeeded [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 004. CancelDestroy [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: 005. AutoDestroy 73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060 [2011-11-08 17:30:51] DEBUG[2629] chan_sip.c: ---------- END SIP HISTORY for '73f104f60d8440102b3941b6297d38c7@192.168.15.251:5060' [2011-11-08 17:30:51] DEBUG[2629] rtp_engine.c: Destroyed RTP instance '0x20571658' [2011-11-08 17:30:51] DEBUG[2629] rtp_engine.c: Destroyed RTP instance '0x206a7f28' [2011-11-08 17:30:51] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:51] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:51] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 1 (Not in use) [2011-11-08 17:30:51] DEBUG[2499] devicestate.c: device 'SIP/5321' state '1' [2011-11-08 17:30:51] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '1' (Not in use) [2011-11-08 17:30:51] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 1 Paused: 0 [2011-11-08 17:30:52] DEBUG[2629] acl.c: For destination '192.168.15.187', our source address is '192.168.15.251'. [2011-11-08 17:30:52] DEBUG[2629] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.251:5060 [2011-11-08 17:30:52] DEBUG[2629] chan_sip.c: Allocating new SIP dialog for 744462e1-d878ed47-de376414@192.168.15.187 - REGISTER (No RTP) [2011-11-08 17:30:52] DEBUG[2629] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2011-11-08 17:30:52] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:52] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:52] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:53] DEBUG[2629] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2011-11-08 17:30:53] DEBUG[2629] netsock2.c: Splitting '192.168.15.187' into... [2011-11-08 17:30:53] DEBUG[2629] netsock2.c: ...host '192.168.15.187' and port ''. [2011-11-08 17:30:53] DEBUG[2629] chan_sip.c: Store REGISTER's src-IP:port for call routing. [2011-11-08 17:30:53] DEBUG[14661] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/5321 PeerStatus: Registered Address: 192.168.15.187:5060 [2011-11-08 17:30:53] DEBUG[2629] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.187:5060 [2011-11-08 17:30:53] DEBUG[2499] devicestate.c: No provider found, checking channel drivers for SIP - 5321 [2011-11-08 17:30:53] DEBUG[2499] chan_sip.c: Checking device state for peer 5321 [2011-11-08 17:30:53] DEBUG[2499] devicestate.c: Changing state for SIP/5321 - state 1 (Not in use) [2011-11-08 17:30:53] DEBUG[2499] devicestate.c: device 'SIP/5321' state '1' [2011-11-08 17:30:53] DEBUG[2756] app_queue.c: Device 'SIP/5321' changed to state '1' (Not in use) [2011-11-08 17:30:53] DEBUG[14661] manager.c: Examining event: Event: QueueMemberStatus Privilege: agent,all Queue: 5550 Location: SIP/5321 MemberName: SIP/5321 Membership: dynamic Penalty: 0 CallsTaken: 23 LastCall: 1317337470 Status: 1 Paused: 0