[Jun 20 13:10:59] Asterisk 11.22.0 built by root @ asterisk on a x86_64 running Linux on 2016-06-13 09:00:20 UTC [Jun 20 13:10:59] DEBUG[19951] config.c: Parsing /opt/asterisk-ibercom/etc/asterisk/logger.conf [Jun 20 13:10:59] VERBOSE[19951] config.c: == Parsing '/opt/asterisk-ibercom/etc/asterisk/logger.conf': Found [Jun 20 13:10:59] VERBOSE[19951] logger.c: Asterisk Queue Logger restarted [Jun 20 13:11:08] DEBUG[13372] chan_sip.c: Auto destroying SIP dialog '730817124ebf576c57702b612ba6d0dc@10.0.0.9:5060' [Jun 20 13:11:08] DEBUG[13372] chan_sip.c: Destroying SIP dialog 730817124ebf576c57702b612ba6d0dc@10.0.0.9:5060 [Jun 20 13:11:08] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '730817124ebf576c57702b612ba6d0dc@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> INVITE sip:23830@asterisk.mydomain.com:5100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK481fcb04 Max-Forwards: 70 From: "37233" ;tag=as3b6a65c5 To: Contact: Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com CSeq: 102 INVITE User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:11:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "37233" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 255 v=0 o=root 462808907 462808907 IN IP4 10.0.0.9 s=Digium Gateway c=IN IP4 10.0.0.9 t=0 0 m=audio 10130 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 0 [ 46]: INVITE sip:23830@asterisk.mydomain.com:5100 SIP/2.0 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK481fcb04 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 3 [ 57]: From: "37233" ;tag=as3b6a65c5 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 4 [ 37]: To: [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 5 [ 38]: Contact: [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 6 [ 58]: Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:11:35 GMT [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 12 [ 90]: Remote-Party-ID: "37233" ;party=calling;privacy=off;screen=yes [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 14 [ 19]: Content-Length: 255 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 15 [ 0]: [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 0 [ 3]: v=0 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 1 [ 46]: o=root 462808907 462808907 IN IP4 10.0.0.9 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 2 [ 16]: s=Digium Gateway [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.0.0.9 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 5 [ 27]: m=audio 10130 RTP/AVP 8 101 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Body 11 [ 10]: a=sendrecv [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: --- (15 headers 12 lines) --- [Jun 20 13:11:35] DEBUG[13372] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:11:35] DEBUG[13372] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:11:35] DEBUG[13372] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Allocating new SIP dialog for 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com - INVITE (No RTP) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 20 13:11:35] DEBUG[13372][C-0000000d] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer" [Jun 20 13:11:35] DEBUG[13372][C-0000000d] sip/reqresp_parser.c: Found SIP option: -replaces- [Jun 20 13:11:35] DEBUG[13372][C-0000000d] sip/reqresp_parser.c: Matched SIP option: replaces [Jun 20 13:11:35] DEBUG[13372][C-0000000d] sip/reqresp_parser.c: Found SIP option: -timer- [Jun 20 13:11:35] DEBUG[13372][C-0000000d] sip/reqresp_parser.c: Matched SIP option: timer [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Initializing initreq for method INVITE - callid 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Using INVITE request as basis request - 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Found peer 'ISP_SIPrunk' for '37233' from 10.0.0.9:5060 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f2e44007f68' [Jun 20 13:11:35] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Allocated port 42232 for RTP instance '0x7f2e44007f68' [Jun 20 13:11:35] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:42232 (42232) for RTP instance '0x7f2e44007f68' [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.31' into... [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.31' and port ''. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] rtp_engine.c: RTP instance '0x7f2e44007f68' is setup and ready to go [Jun 20 13:11:35] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f2e44007f68' [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] netsock2.c: == Using SIP RTP TOS bits 184 [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Setting NAT on RTP to On [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP o=root 462808907 462808907 IN IP4 10.0.0.9... OK. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP s=Digium Gateway... UNSUPPORTED OR FAILED. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.9... OK. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Found RTP audio format 8 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] rtp_engine.c: Setting payload 8 based on m type on 0x7f2e32785600 [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Found RTP audio format 101 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] rtp_engine.c: Setting payload 101 based on m type on 0x7f2e32785600 [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e44007f68' [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Peer audio RTP is at port 10.0.0.9:10130 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] rtp_engine.c: Copying payload 8 from 0x7f2e32785600 to 0x7f2e44008130 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] rtp_engine.c: Copying payload 101 from 0x7f2e32785600 to 0x7f2e44008130 [Jun 20 13:11:35] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f2e44007f68' [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: We're settling with these formats: (alaw) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Checking SIP call limits for device [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Updating call counter for incoming call [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: Splitting 'asterisk.mydomain.com:5100' into... [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:11:35] DEBUG[13372][C-0000000d] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: Looking for 23830 in incoming (domain asterisk.mydomain.com) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Incoming INVITE with 'timer' option supported [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: *** Our native formats are (alaw) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: *** Joint capabilities are (alaw) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: *** Our capabilities are (alaw|g729) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: This channel will not be able to handle video. [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Contact hop: [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: SIP/ISP_SIPrunk-00000025: New call is still down.... Trying... [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.9:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK481fcb04;received=10.0.0.9;rport=5060 From: "37233" ;tag=as3b6a65c5 To: Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com CSeq: 102 INVITE Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:11:35] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - ISP_SIPrunk [Jun 20 13:11:35] DEBUG[13365] chan_sip.c: Checking device state for peer ISP_SIPrunk [Jun 20 13:11:35] DEBUG[13365] devicestate.c: Changing state for SIP/ISP_SIPrunk - state 1 (Not in use) [Jun 20 13:11:35] DEBUG[13365] devicestate.c: device 'SIP/ISP_SIPrunk' state '1' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'EXTEN' is '23830' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Function CALLERID(num) result is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Macro' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [23830@incoming:1] Macro("SIP/ISP_SIPrunk-00000025", "llamadaEntranteUC,23830,37233") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'CALLINGPRES' is '1' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Function CALLERID(all) result is '"37233" <37233>' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'NoOp' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:1] NoOp("SIP/ISP_SIPrunk-00000025", "CallingPress --1-- caller id --"37233" <37233>--") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: NoOp [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:2] Set("SIP/ISP_SIPrunk-00000025", "TRANSFER_CONTEXT=station_transferUC") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:3] Set("SIP/ISP_SIPrunk-00000025", "_TRANSFER_CONTEXT=station_transferUC") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:4] Set("SIP/ISP_SIPrunk-00000025", "__TRANSFER_CONTEXT=station_transferUC") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG1' is '23830' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:5] Set("SIP/ISP_SIPrunk-00000025", "NUMCALLED=23830") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG2' is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:6] Set("SIP/ISP_SIPrunk-00000025", "NUMCALLER=37233") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Expression result is '0' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'GotoIf' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:7] GotoIf("SIP/ISP_SIPrunk-00000025", "0?desconegut:conegut") in new stack [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Goto (macro-llamadaEntranteUC,s,10) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: GotoIf [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Expression result is '1' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Function LEN(37233) result is '5' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Expression result is '0' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Expression result is '0' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'GotoIf' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:10] GotoIf("SIP/ISP_SIPrunk-00000025", "0?addZeroPrefix") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Not taking any branch [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: GotoIf [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Goto' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:11] Goto("SIP/ISP_SIPrunk-00000025", "s,nextZeroPrefix") in new stack [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Goto (macro-llamadaEntranteUC,s,13) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Goto [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMCALLED' is '23830' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:13] Set("SIP/ISP_SIPrunk-00000025", "_UCUSER=23830") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:14] Set("SIP/ISP_SIPrunk-00000025", "CDR(PRIVACY-ID)="Privacy: off"") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Function CALLERID(name) result is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:15] Set("SIP/ISP_SIPrunk-00000025", "CALLERID(all)=37233 <37233>") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Function CALLERID(name) result is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'DOMINI' is 'mydomain.com' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'SIPAddHeader' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:16] SIPAddHeader("SIP/ISP_SIPrunk-00000025", "P-Asserted-Identity: 37233 ") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: SIP Header added "P-Asserted-Identity: 37233 " as __SIPADDHEADER01 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: SIPAddHeader [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'CALLINGPRES' is '1' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Function MATH(1>30) result is 'FALSE' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Expression result is '0' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'GotoIf' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:17] GotoIf("SIP/ISP_SIPrunk-00000025", "0?privacy:privacyoff") in new stack [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Goto (macro-llamadaEntranteUC,s,22) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: GotoIf [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Function CALLERID(name) result is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'DOMINI' is 'mydomain.com' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'SIPAddHeader' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:22] SIPAddHeader("SIP/ISP_SIPrunk-00000025", "Remote-Party-ID: 37233 ") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: SIP Header added "Remote-Party-ID: 37233 " as __SIPADDHEADER02 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] app_macro.c: Executed application: SIPAddHeader [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMCALLED' is '23830' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] pbx.c: Launching 'Dial' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-llamadaEntranteUC:23] Dial("SIP/ISP_SIPrunk-00000025", "SIP/PROXY-out/23830,,t") in new stack [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Allocating new SIP dialog for 6a49ca461cceac9e0c1544c730f1816e@mydomain.com - INVITE (No RTP) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f2e34037868' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Allocated port 44752 for RTP instance '0x7f2e34037868' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:44752 (44752) for RTP instance '0x7f2e34037868' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] netsock2.c: Splitting '10.0.0.31' into... [Jun 20 13:11:35] DEBUG[19952][C-0000000d] netsock2.c: ...host '10.0.0.31' and port ''. [Jun 20 13:11:35] DEBUG[19952][C-0000000d] rtp_engine.c: RTP instance '0x7f2e34037868' is setup and ready to go [Jun 20 13:11:35] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f2e34037868' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] netsock2.c: == Using SIP RTP TOS bits 184 [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Setting NAT on RTP to On [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:11:35] DEBUG[19952][C-0000000d] acl.c: For destination '10.0.0.37', our source address is '10.0.0.31'. [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Setting NAT on RTP to On [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: SIP call-id changed from '6a49ca461cceac9e0c1544c730f1816e@mydomain.com' to '627d96dc204c343679fb0b126b8ad013@mydomain.com' [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: *** Our native formats are (alaw) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: *** Joint capabilities are (alaw) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: *** Our capabilities are (alaw|g729) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: This channel will not be able to handle video. [Jun 20 13:11:35] DEBUG[19952][C-0000000d] channel_internal_api.c: Channel Call ID changing from [C-0000000d] to [C-0000000d] [Jun 20 13:11:35] DEBUG[19952][C-0000000d] channel.c: Inheriting variable SIPADDHEADER02 from SIP/ISP_SIPrunk-00000025 to SIP/PROXY-out-00000026. [Jun 20 13:11:35] DEBUG[19952][C-0000000d] channel.c: Inheriting variable SIPADDHEADER01 from SIP/ISP_SIPrunk-00000025 to SIP/PROXY-out-00000026. [Jun 20 13:11:35] DEBUG[19952][C-0000000d] channel.c: Inheriting variable UCUSER from SIP/ISP_SIPrunk-00000025 to SIP/PROXY-out-00000026. [Jun 20 13:11:35] DEBUG[19952][C-0000000d] channel.c: Inheriting variable TRANSFER_CONTEXT from SIP/ISP_SIPrunk-00000025 to SIP/PROXY-out-00000026. [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Outgoing Call for 23830 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Adding SIP Header "Remote-Party-ID" with content :37233 : [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Adding SIP Header "P-Asserted-Identity" with content :37233 : [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: ** Our capability: (alaw|g729) Video flag: False Text flag: False [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: ** Our prefcodec: (alaw) [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] chan_sip.c: Audio is at 44752 [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] chan_sip.c: Adding codec 100004 (alaw) to SDP [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] chan_sip.c: Adding codec 100008 (g729) to SDP [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: -- Done with adding codecs to SDP [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Done building SDP. Settling with this capability: (alaw|g729) [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Initializing initreq for method INVITE - callid 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 0 [ 48]: INVITE sip:23830@mydomain.com:5070 SIP/2.0 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK4f3c94e9;rport [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 3 [ 64]: From: "37233" ;tag=as368c8ffe [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 4 [ 39]: To: [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 5 [ 39]: Contact: [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 6 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 11.22.0 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:11:35 GMT [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 12 [ 53]: Remote-Party-ID: 37233 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 13 [ 57]: P-Asserted-Identity: 37233 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] chan_sip.c: Reliably Transmitting (NAT) to 10.0.0.37:5070: INVITE sip:23830@mydomain.com:5070 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK4f3c94e9;rport Max-Forwards: 70 From: "37233" ;tag=as368c8ffe To: Contact: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 102 INVITE User-Agent: Asterisk PBX 11.22.0 Date: Mon, 20 Jun 2016 11:11:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: 37233 P-Asserted-Identity: 37233 Content-Type: application/sdp Content-Length: 285 v=0 o=root 2026361913 2026361913 IN IP4 10.0.0.31 s=Asterisk PBX 11.22.0 c=IN IP4 10.0.0.31 t=0 0 m=audio 44752 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] app_dial.c: -- Called SIP/PROXY-out/23830 [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> SIP/2.0 100 Runing Incoming CPL script... Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK4f3c94e9;rport=5100 From: "37233" ;tag=as368c8ffe To: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 102 INVITE Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 <-------------> [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 0 [ 41]: SIP/2.0 100 Runing Incoming CPL script... [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK4f3c94e9;rport=5100 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as368c8ffe [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 3 [ 39]: To: [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 6 [ 43]: Server: OpenSIPS (1.6.2-notls (i386/linux)) [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 8 [ 0]: [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: --- (8 headers 0 lines) --- [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: *** SIP TIMER: Cancelling retransmission #16 - INVITE (got response) [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '627d96dc204c343679fb0b126b8ad013@mydomain.com' Request 102: Found [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: SIP response 100 to standard invite [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK4f3c94e9;rport=5100 From: "37233" ;tag=as368c8ffe To: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 102 INVITE Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 <-------------> [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK4f3c94e9;rport=5100 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as368c8ffe [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 3 [ 39]: To: [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 6 [ 43]: Server: OpenSIPS (1.6.2-notls (i386/linux)) [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 8 [ 0]: [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: --- (8 headers 0 lines) --- [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '627d96dc204c343679fb0b126b8ad013@mydomain.com' Request 102: Found [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: SIP response 100 to standard invite [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK4f3c94e9;rport=5100 From: "37233" ;tag=as368c8ffe To: "userB" ;tag=1E252C18-715B31C3 CSeq: 102 INVITE Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com Contact: Record-Route: , , User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 Allow-Events: talk,hold,conference Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 Content-Length: 0 <-------------> [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK4f3c94e9;rport=5100 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as368c8ffe [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 3 [ 84]: To: "userB" ;tag=1E252C18-715B31C3 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 5 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 6 [ 33]: Contact: [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 7 [157]: Record-Route: , , [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 9 [ 34]: Allow-Events: talk,hold,conference [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 10 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jun 20 13:11:35] DEBUG[13372] chan_sip.c: Header 12 [ 0]: [Jun 20 13:11:35] VERBOSE[13372] chan_sip.c: --- (12 headers 0 lines) --- [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '627d96dc204c343679fb0b126b8ad013@mydomain.com' Request 102: Found [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: SIP response 180 to standard invite [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:35] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:35] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] app_dial.c: -- SIP/PROXY-out-00000026 is ringing [Jun 20 13:11:35] DEBUG[19952][C-0000000d] rtp_engine.c: Setting early bridge SDP of 'SIP/ISP_SIPrunk-00000025' with that of 'SIP/PROXY-out-00000026' [Jun 20 13:11:35] VERBOSE[19952][C-0000000d] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.9:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK481fcb04;received=10.0.0.9;rport=5060 From: "37233" ;tag=as3b6a65c5 To: ;tag=as5d0bd156 Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com CSeq: 102 INVITE Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Jun 20 13:11:35] DEBUG[19952][C-0000000d] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:11:35] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:11:35] DEBUG[13365] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:11:35] DEBUG[13365] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:11:35] DEBUG[13365] devicestate.c: device 'SIP/PROXY-out' state '1' [Jun 20 13:11:36] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> OPTIONS sip:asterisk.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK7663ae96 Max-Forwards: 70 From: "asterisk" ;tag=as326ee10e To: Contact: Call-ID: 3c2d1b3878b2b7a768073e3d1cf04691@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:11:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK7663ae96 [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as326ee10e [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 6 [ 59]: Call-ID: 3c2d1b3878b2b7a768073e3d1cf04691@10.0.0.9:5060 [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:11:36 GMT [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Header 13 [ 0]: [Jun 20 13:11:36] VERBOSE[13372] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:11:36] DEBUG[13372] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:11:36] DEBUG[13372] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:11:36] DEBUG[13372] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:11:36] VERBOSE[13372] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Allocating new SIP dialog for 3c2d1b3878b2b7a768073e3d1cf04691@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:11:36] DEBUG[13372] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:11:36] DEBUG[13372] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:11:36] DEBUG[13372] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:11:36] DEBUG[13372] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:11:36] VERBOSE[13372] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:11:36] VERBOSE[13372] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.9:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK7663ae96;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as326ee10e To: ;tag=as3c56c559 Call-ID: 3c2d1b3878b2b7a768073e3d1cf04691@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:11:36] DEBUG[13372] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:11:36] VERBOSE[13372] chan_sip.c: Scheduling destruction of SIP dialog '3c2d1b3878b2b7a768073e3d1cf04691@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:11:38] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK4f3c94e9;rport=5100 From: "37233" ;tag=as368c8ffe To: "userB" ;tag=1E252C18-715B31C3 CSeq: 102 INVITE Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com Contact: Record-Route: , , Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 Content-Type: application/sdp Content-Length: 233 P-RTP-Proxy: Yes v=0 o=- 1466365615 1466365615 IN IP4 10.11.198.230 s=Polycom IP Phone c=IN IP4 10.0.0.174 t=0 0 a=sendrecv m=audio 25106 RTP/AVP 8 127 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=nortpproxy:yes <-------------> [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK4f3c94e9;rport=5100 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as368c8ffe [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 3 [ 84]: To: "userB" ;tag=1E252C18-715B31C3 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 5 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 6 [ 33]: Contact: [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 7 [157]: Record-Route: , , [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 8 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 9 [ 26]: Supported: 100rel,replaces [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 10 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 11 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 13 [ 19]: Content-Length: 233 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 14 [ 16]: P-RTP-Proxy: Yes [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 15 [ 0]: [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 0 [ 3]: v=0 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 1 [ 46]: o=- 1466365615 1466365615 IN IP4 10.11.198.230 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 3 [ 23]: c=IN IP4 10.0.0.174 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 5 [ 10]: a=sendrecv [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 6 [ 27]: m=audio 25106 RTP/AVP 8 127 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 7 [ 10]: a=sendrecv [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 9 [ 33]: a=rtpmap:127 telephone-event/8000 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Body 10 [ 16]: a=nortpproxy:yes [Jun 20 13:11:38] VERBOSE[13372] chan_sip.c: --- (15 headers 11 lines) --- [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Acked pending invite 102 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Stopping retransmission on '627d96dc204c343679fb0b126b8ad013@mydomain.com' of Request 102: Match Found [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: SIP response 200 to standard invite [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP o=- 1466365615 1466365615 IN IP4 10.11.198.230... OK. [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Jun 20 13:11:38] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.174' into... [Jun 20 13:11:38] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.174' and port ''. [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.174... OK. [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: Found RTP audio format 8 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] rtp_engine.c: Setting payload 8 based on m type on 0x7f2e32784a00 [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: Found RTP audio format 127 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] rtp_engine.c: Setting payload 127 based on m type on 0x7f2e32784a00 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: Found audio description format telephone-event for ID 127 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:127 telephone-event/8000... OK. [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=nortpproxy:yes... UNSUPPORTED OR FAILED. [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 20 13:11:38] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Set role to CONTROLLING (0x7f2e34037868) [Jun 20 13:11:38] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Set role failed; no ice instance (0x7f2e34037868) [Jun 20 13:11:38] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e34037868' [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: Peer audio RTP is at port 10.0.0.174:25106 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] rtp_engine.c: Copying payload 8 from 0x7f2e32784a00 to 0x7f2e34037a30 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] rtp_engine.c: Copying payload 127 from 0x7f2e32784a00 to 0x7f2e34037a30 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f2e34037868' [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: We're settling with these formats: (alaw) [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: We have an owner, now see if we need to change this call [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 20 13:11:38] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.37:5070' into... [Jun 20 13:11:38] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.37' and port '5070'. [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: set_destination: set destination to 10.0.0.37:5070 [Jun 20 13:11:38] VERBOSE[13372][C-0000000d] chan_sip.c: Transmitting (NAT) to 10.0.0.37:5070: ACK sip:u3-0@10.0.0.37 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK31e96492;rport Route: ,, Max-Forwards: 70 From: "37233" ;tag=as368c8ffe To: ;tag=1E252C18-715B31C3 Contact: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 102 ACK User-Agent: Asterisk PBX 11.22.0 Content-Length: 0 --- [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'ACK sip:u3-' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:11:38] VERBOSE[19952][C-0000000d] app_dial.c: -- SIP/PROXY-out-00000026 answered SIP/ISP_SIPrunk-00000025 [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: SIP answering channel: SIP/ISP_SIPrunk-00000025 [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: Setting framing from config on incoming call [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: ** Our prefcodec: (nothing) [Jun 20 13:11:38] VERBOSE[19952][C-0000000d] chan_sip.c: Audio is at 42232 [Jun 20 13:11:38] VERBOSE[19952][C-0000000d] chan_sip.c: Adding codec 100004 (alaw) to SDP [Jun 20 13:11:38] VERBOSE[19952][C-0000000d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: -- Done with adding codecs to SDP [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Jun 20 13:11:38] VERBOSE[19952][C-0000000d] chan_sip.c: <--- Reliably Transmitting (NAT) to 10.0.0.9:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK481fcb04;received=10.0.0.9;rport=5060 From: "37233" ;tag=as3b6a65c5 To: ;tag=as5d0bd156 Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com CSeq: 102 INVITE Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 238 v=0 o=root 1148723751 1148723751 IN IP4 10.0.0.31 s=Asterisk PBX 11.22.0 c=IN IP4 10.0.0.31 t=0 0 m=audio 42232 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:11:38] DEBUG[19952][C-0000000d] chan_sip.c: Session timer started: 8 - 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com 900000ms [Jun 20 13:11:38] DEBUG[19952][C-0000000d] features.c: bridge answer set, chan answer set [Jun 20 13:11:38] DEBUG[19952][C-0000000d] features.c: Removing dialed interfaces datastore on SIP/PROXY-out-00000026 since we're bridging [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:38] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:11:38] DEBUG[13365] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:11:38] DEBUG[13365] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:11:38] DEBUG[13365] devicestate.c: device 'SIP/PROXY-out' state '1' [Jun 20 13:11:38] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - ISP_SIPrunk [Jun 20 13:11:38] DEBUG[13365] chan_sip.c: Checking device state for peer ISP_SIPrunk [Jun 20 13:11:38] DEBUG[13365] devicestate.c: Changing state for SIP/ISP_SIPrunk - state 1 (Not in use) [Jun 20 13:11:38] DEBUG[13365] devicestate.c: device 'SIP/ISP_SIPrunk' state '1' [Jun 20 13:11:38] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> ACK sip:23830@10.0.0.31:5100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK00891ef8 Max-Forwards: 70 From: "37233" ;tag=as3b6a65c5 To: ;tag=as5d0bd156 Contact: Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com CSeq: 102 ACK User-Agent: Digium Gateway Content-Length: 0 <-------------> [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 0 [ 40]: ACK sip:23830@10.0.0.31:5100 SIP/2.0 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK00891ef8 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 3 [ 57]: From: "37233" ;tag=as3b6a65c5 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 4 [ 52]: To: ;tag=as5d0bd156 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 5 [ 38]: Contact: [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 6 [ 58]: Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 20 13:11:38] DEBUG[13372] chan_sip.c: Header 10 [ 0]: [Jun 20 13:11:38] VERBOSE[13372] chan_sip.c: --- (10 headers 0 lines) --- [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Jun 20 13:11:38] DEBUG[13372][C-0000000d] chan_sip.c: Stopping retransmission on '111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com' of Response 102: Match Found [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: 0x7f2e44011810 -- Probation learning mode pass with source address 10.0.0.9:10130 [Jun 20 13:11:38] VERBOSE[19952][C-0000000d] res_rtp_asterisk.c: > 0x7f2e44011810 -- Probation passed - setting RTP source address to 10.0.0.9:10130 [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f2e34037868' [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: 0x7f2e34005600 -- Probation learning mode pass with source address 10.0.0.174:25106 [Jun 20 13:11:38] VERBOSE[19952][C-0000000d] res_rtp_asterisk.c: > 0x7f2e34005600 -- Probation passed - setting RTP source address to 10.0.0.174:25106 [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Jun 20 13:11:38] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Jun 20 13:11:40] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> INVITE sip:37233@10.0.0.31:5100 SIP/2.0 Record-Route: Record-Route: Record-Route: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.38515c32.0 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.28515c32.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKa6af.633eb6a1.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKf65a9716DF06F6D1 From: "userB" ;tag=1E252C18-715B31C3 To: "37233" ;tag=as368c8ffe CSeq: 1 INVITE Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com Contact: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 Allow-Events: talk,hold,conference Max-Forwards: 11 Content-Type: application/sdp Content-Length: 233 P-RTP-Proxy: Yes v=0 o=- 1466365615 1466365616 IN IP4 10.11.198.230 s=Polycom IP Phone c=IN IP4 10.0.0.174 t=0 0 a=sendonly m=audio 25106 RTP/AVP 8 101 a=sendonly a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=nortpproxy:yes <-------------> [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 0 [ 43]: INVITE sip:37233@10.0.0.31:5100 SIP/2.0 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 1 [ 67]: Record-Route: [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 2 [ 67]: Record-Route: [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 3 [ 68]: Record-Route: [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 4 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.38515c32.0 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 5 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.28515c32.0 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 6 [ 60]: Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKa6af.633eb6a1.0 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 7 [ 72]: Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKf65a9716DF06F6D1 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 8 [ 86]: From: "userB" ;tag=1E252C18-715B31C3 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 9 [ 62]: To: "37233" ;tag=as368c8ffe [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 10 [ 14]: CSeq: 1 INVITE [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 11 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 12 [ 33]: Contact: [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 13 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 14 [ 34]: Allow-Events: talk,hold,conference [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 15 [ 16]: Max-Forwards: 11 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 17 [ 19]: Content-Length: 233 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 18 [ 16]: P-RTP-Proxy: Yes [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 19 [ 0]: [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 0 [ 3]: v=0 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 1 [ 46]: o=- 1466365615 1466365616 IN IP4 10.11.198.230 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 3 [ 23]: c=IN IP4 10.0.0.174 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 5 [ 10]: a=sendonly [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 6 [ 27]: m=audio 25106 RTP/AVP 8 101 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 7 [ 10]: a=sendonly [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Body 10 [ 16]: a=nortpproxy:yes [Jun 20 13:11:40] VERBOSE[13372] chan_sip.c: --- (19 headers 11 lines) --- [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 20 13:11:40] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.37:5070' into... [Jun 20 13:11:40] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.37' and port '5070'. [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Sending to 10.0.0.37:5070 (NAT) [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Initializing initreq for method INVITE - callid 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP o=- 1466365615 1466365616 IN IP4 10.11.198.230... OK. [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Jun 20 13:11:40] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.174' into... [Jun 20 13:11:40] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.174' and port ''. [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.174... OK. [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP a=sendonly... OK. [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Found RTP audio format 8 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] rtp_engine.c: Setting payload 8 based on m type on 0x7f2e32785600 [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Found RTP audio format 101 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] rtp_engine.c: Setting payload 101 based on m type on 0x7f2e32785600 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=nortpproxy:yes... UNSUPPORTED OR FAILED. [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 20 13:11:40] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e34037868' [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Peer audio RTP is at port 10.0.0.174:25106 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] rtp_engine.c: Copying payload 8 from 0x7f2e32785600 to 0x7f2e34037a30 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] rtp_engine.c: Copying payload 101 from 0x7f2e32785600 to 0x7f2e34037a30 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f2e34037868' [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: We're settling with these formats: (alaw) [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: We have an owner, now see if we need to change this call [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Jun 20 13:11:40] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e34037868' [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Got a SIP re-invite for call 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: SIP/PROXY-out-00000026: This call is UP.... [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.37:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.38515c32.0;received=10.0.0.37;rport=5070 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.28515c32.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKa6af.633eb6a1.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKf65a9716DF06F6D1 Record-Route: Record-Route: Record-Route: From: "userB" ;tag=1E252C18-715B31C3 To: "37233" ;tag=as368c8ffe Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 1 INVITE Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Setting framing from config on incoming call [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: ** Our prefcodec: (alaw) [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Audio is at 44752 [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Adding codec 100004 (alaw) to SDP [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: -- Done with adding codecs to SDP [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Jun 20 13:11:40] VERBOSE[13372][C-0000000d] chan_sip.c: <--- Reliably Transmitting (NAT) to 10.0.0.37:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.38515c32.0;received=10.0.0.37;rport=5070 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.28515c32.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKa6af.633eb6a1.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKf65a9716DF06F6D1 Record-Route: Record-Route: Record-Route: From: "userB" ;tag=1E252C18-715B31C3 To: "37233" ;tag=as368c8ffe Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 1 INVITE Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 2026361913 2026361914 IN IP4 10.0.0.31 s=Asterisk PBX 11.22.0 c=IN IP4 10.0.0.31 t=0 0 m=audio 44752 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly <------------> [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:40] VERBOSE[19952][C-0000000d] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/ISP_SIPrunk-00000025 [Jun 20 13:11:40] DEBUG[19952][C-0000000d] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:40] DEBUG[19952][C-0000000d] channel.c: Got a FRAME_CONTROL (32) frame on channel SIP/PROXY-out-00000026 [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:40] DEBUG[19952][C-0000000d] channel.c: Bridge stops bridging channels SIP/ISP_SIPrunk-00000025 and SIP/PROXY-out-00000026 [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] channel.c: Set channel SIP/ISP_SIPrunk-00000025 to write format slin [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_musiconhold.c: SIP/ISP_SIPrunk-00000025 Opened file 0 '/opt/asterisk-ibercom/var/lib/asterisk/moh/reno_project-system' [Jun 20 13:11:40] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> ACK sip:37233@10.0.0.31:5100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.28515c32.2 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.28515c32.2 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKa6af.633eb6a1.2 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKa9da55b4E57D27F From: "userB" ;tag=1E252C18-715B31C3 To: "37233" ;tag=as368c8ffe CSeq: 1 ACK Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com Contact: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 Max-Forwards: 11 Content-Length: 0 <-------------> [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 0 [ 40]: ACK sip:37233@10.0.0.31:5100 SIP/2.0 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.28515c32.2 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 2 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKa6af.28515c32.2 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 3 [ 60]: Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKa6af.633eb6a1.2 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 4 [ 71]: Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKa9da55b4E57D27F [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 5 [ 86]: From: "userB" ;tag=1E252C18-715B31C3 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 6 [ 62]: To: "37233" ;tag=as368c8ffe [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 7 [ 11]: CSeq: 1 ACK [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 8 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 9 [ 33]: Contact: [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 10 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 11 [ 16]: Max-Forwards: 11 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:11:40] DEBUG[13372] chan_sip.c: Header 13 [ 0]: [Jun 20 13:11:40] VERBOSE[13372] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Jun 20 13:11:40] DEBUG[13372][C-0000000d] chan_sip.c: Stopping retransmission on '627d96dc204c343679fb0b126b8ad013@mydomain.com' of Response 1: Match Found [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:40] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:41] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:42] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Got RTCP report of 80 bytes [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:43] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:44] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:45] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:46] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:47] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Got RTCP report of 60 bytes [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:48] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e34037868' so dropping frame [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> REFER sip:37233@10.0.0.31:5100 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK76af.287d9624.0 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK76af.187d9624.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK76af.4c5b9121.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK48227bbd902BA740 From: "userB" ;tag=1E252C18-715B31C3 To: "37233" ;tag=as368c8ffe CSeq: 2 REFER Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com Contact: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 Refer-To: Referred-By: Max-Forwards: 11 Content-Length: 0 <-------------> [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 0 [ 42]: REFER sip:37233@10.0.0.31:5100 SIP/2.0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 1 [ 68]: Record-Route: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 2 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK76af.287d9624.0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 3 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK76af.187d9624.0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK76af.4c5b9121.0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 5 [ 72]: Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK48227bbd902BA740 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 6 [ 86]: From: "userB" ;tag=1E252C18-715B31C3 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 7 [ 62]: To: "37233" ;tag=as368c8ffe [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 8 [ 13]: CSeq: 2 REFER [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 9 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 10 [ 33]: Contact: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 11 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 12 [137]: Refer-To: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 13 [ 58]: Referred-By: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 14 [ 16]: Max-Forwards: 11 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 16 [ 0]: [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: --- (16 headers 0 lines) --- [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Call 627d96dc204c343679fb0b126b8ad013@mydomain.com got a SIP call transfer from caller: (REFER)! [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 8a52df2a-8e60a45-4ff57988@10.11.198.230 (No check of from/to tags) [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: SIP transfer to extension u6-2@station_transferUC by userB@mydomain.com [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: This SIP transfer is to a remote SIP extension (remote domain 10.0.0.37?Replaces=8a52df2a-8e60a45-4ff57988%4010.11.198.230%3Bto-tag%3D963635824%3Bfrom-tag%3DF680C2CD-244325D0) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: SIP attended transfer: Transferer channel SIP/PROXY-out-00000026, transferee channel SIP/ISP_SIPrunk-00000025 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.37:5070 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK76af.287d9624.0;received=10.0.0.37;rport=5070 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK76af.187d9624.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK76af.4c5b9121.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK48227bbd902BA740 Record-Route: From: "userB" ;tag=1E252C18-715B31C3 To: "37233" ;tag=as368c8ffe Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 2 REFER Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Looking for callid 8a52df2a-8e60a45-4ff57988@10.11.198.230 (fromtag F680C2CD-244325D0 totag 963635824) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: SIP attended transfer: Not our call - generating INVITE with replaces [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: SIP attended transfer: Still not our call - generating INVITE with replaces [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: chan1->name: SIP/PROXY-out-00000026 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] res_musiconhold.c: -- Stopped music on hold on SIP/ISP_SIPrunk-00000025 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] channel.c: Set channel SIP/ISP_SIPrunk-00000025 to write format alaw [Jun 20 13:11:49] DEBUG[13372][C-0000000d] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 20 13:11:49] DEBUG[13372][C-0000000d] channel.c: Soft-Hanging up channel 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Attended transfer succeeded. Telling transferer. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 20 13:11:49] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.37:5070' into... [Jun 20 13:11:49] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.37' and port '5070'. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: set_destination: set destination to 10.0.0.37:5070 [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Reliably Transmitting (NAT) to 10.0.0.37:5070: NOTIFY sip:u3-0@10.0.0.37 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK60bdbcb1;rport Route: ,, Max-Forwards: 70 From: "37233" ;tag=as368c8ffe To: "userB" ;tag=1E252C18-715B31C3 Contact: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 103 NOTIFY User-Agent: Asterisk PBX 11.22.0 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 16 SIP/2.0 200 Ok --- [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Didn't get a frame from channel: SIP/ISP_SIPrunk-00000025 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Bridge stops bridging channels SIP/ISP_SIPrunk-00000025 and SIP/PROXY-out-00000026 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(clid)' (from 'CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 9) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(clid) result is '"37233" <37233>' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 8) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(src) result is '37233' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 8) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(dst) result is '23830' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(dcontext) result is 'incoming' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 12) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(channel) result is 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 15) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(dstchannel) result is 'SIP/PROXY-out-00000026' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 12) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(lastapp) result is 'Dial' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(lastdata) result is 'SIP/PROXY-out/23830,,t' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 10) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(start) result is '2016-06-20 13:11:35' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 11) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(answer) result is '2016-06-20 13:11:38' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 8) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(end) result is '2016-06-20 13:11:49' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(duration)' (from 'CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(duration) result is '14' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(billsec)' (from 'CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 12) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(billsec) result is '11' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 16) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 16) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(accountcode) result is '(null)' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(uniqueid) result is 'asterisk-ibercom-1466421095.37' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)}","${CDR(privacy-id)}" ' len 14) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(userfield) result is '(null)' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(privacy-id)' (from 'CDR(privacy-id)}" ' len 15) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CDR(privacy-id) result is '"Privacy: off"' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Hanging up channel 'SIP/PROXY-out-00000026' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: update_call_counter(23830) - decrement call limit counter on hangup [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Call to peer 'PROXY-out' removed from call limit 0 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 627d96dc204c343679fb0b126b8ad013@mydomain.com. [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] chan_sip.c: Scheduling destruction of SIP dialog '627d96dc204c343679fb0b126b8ad013@mydomain.com' in 32000 ms (Method: REFER) [Jun 20 13:11:49] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:11:49] DEBUG[13365] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:11:49] DEBUG[13365] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:11:49] DEBUG[13365] devicestate.c: device 'SIP/PROXY-out' state '1' [Jun 20 13:11:49] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:11:49] DEBUG[13365] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:11:49] DEBUG[13365] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:11:49] DEBUG[13365] devicestate.c: device 'SIP/PROXY-out' state '1' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] app_macro.c: Spawn extension (station_transferUC,u6-2,1) exited non-zero on 'SIP/ISP_SIPrunk-00000025' in macro 'llamadaEntranteUC' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] app_macro.c: == Spawn extension (station_transferUC, u6-2, 1) exited non-zero on 'SIP/ISP_SIPrunk-00000025' in macro 'llamadaEntranteUC' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Spawn extension (station_transferUC,u6-2,1) exited non-zero on 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] pbx.c: == Spawn extension (station_transferUC, u6-2, 1) exited non-zero on 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'EXTEN' is 'u6-2' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Function CALLERID(num) result is '37233' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'UCLID' is NULL [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Launching 'Macro' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [u6-2@station_transferUC:1] Macro("SIP/ISP_SIPrunk-00000025", "transferUCAAi,u6-2,37233,") in new stack [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG2' is '37233' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG1' is 'u6-2' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG3' is '' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG4' is NULL [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'UCUSER' is '23830' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Launching 'NoOp' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-transferUCAAi:1] NoOp("SIP/ISP_SIPrunk-00000025", "Transfer call 37233 to u6-2. Prefix . GWFailover . --UCUser: 23830--") in new stack [Jun 20 13:11:49] DEBUG[19952][C-0000000d] app_macro.c: Executed application: NoOp [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG1' is 'u6-2' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-transferUCAAi:2] Set("SIP/ISP_SIPrunk-00000025", "NUMDESTI=u6-2") in new stack [Jun 20 13:11:49] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG2' is '37233' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-transferUCAAi:3] Set("SIP/ISP_SIPrunk-00000025", "NUMORIGEN=37233") in new stack [Jun 20 13:11:49] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG3' is '' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-transferUCAAi:4] Set("SIP/ISP_SIPrunk-00000025", "CALLEDPREFIX=") in new stack [Jun 20 13:11:49] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'ARG4' is NULL [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Launching 'Set' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-transferUCAAi:5] Set("SIP/ISP_SIPrunk-00000025", "GWFAILOVER=") in new stack [Jun 20 13:11:49] DEBUG[19952][C-0000000d] app_macro.c: Executed application: Set [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Result of 'NUMDESTI' is 'u6-2' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] pbx.c: Launching 'Dial' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] pbx.c: -- Executing [s@macro-transferUCAAi:6] Dial("SIP/ISP_SIPrunk-00000025", "SIP/NB-out/u6-2,,t") in new stack [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Allocating new SIP dialog for 28cb4270469bc1393f4e69934c3e2334@mydomain.com - INVITE (No RTP) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f2e340166f8' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Allocated port 43718 for RTP instance '0x7f2e340166f8' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:43718 (43718) for RTP instance '0x7f2e340166f8' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] netsock2.c: Splitting '10.0.0.31' into... [Jun 20 13:11:49] DEBUG[19952][C-0000000d] netsock2.c: ...host '10.0.0.31' and port ''. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] rtp_engine.c: RTP instance '0x7f2e340166f8' is setup and ready to go [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f2e340166f8' [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] netsock2.c: == Using SIP RTP TOS bits 184 [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Setting NAT on RTP to On [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:11:49] DEBUG[19952][C-0000000d] acl.c: For destination '10.0.0.37', our source address is '10.0.0.31'. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Setting NAT on RTP to On [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: SIP call-id changed from '28cb4270469bc1393f4e69934c3e2334@mydomain.com' to '19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: *** Our native formats are (alaw) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: *** Joint capabilities are (alaw) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: *** Our capabilities are (alaw|g729) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: This channel will not be able to handle video. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel_internal_api.c: Channel Call ID changing from [C-0000000d] to [C-0000000d] [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Inheriting variable SIPTRANSFER_REPLACES from SIP/ISP_SIPrunk-00000025 to SIP/NB-out-00000027. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Inheriting variable SIPTRANSFER_REFERER from SIP/ISP_SIPrunk-00000025 to SIP/NB-out-00000027. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Inheriting variable SIPTRANSFER from SIP/ISP_SIPrunk-00000025 to SIP/NB-out-00000027. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Inheriting variable SIPADDHEADER02 from SIP/ISP_SIPrunk-00000025 to SIP/NB-out-00000027. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Inheriting variable SIPADDHEADER01 from SIP/ISP_SIPrunk-00000025 to SIP/NB-out-00000027. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Inheriting variable UCUSER from SIP/ISP_SIPrunk-00000025 to SIP/NB-out-00000027. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] channel.c: Inheriting variable TRANSFER_CONTEXT from SIP/ISP_SIPrunk-00000025 to SIP/NB-out-00000027. [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Call for u6-2 transferred by [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Outgoing Call for u6-2 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Adding SIP Header "Remote-Party-ID" with content :37233 : [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Adding SIP Header "P-Asserted-Identity" with content :37233 : [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: ** Our capability: (alaw|g729) Video flag: False Text flag: False [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: ** Our prefcodec: (alaw) [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] chan_sip.c: Audio is at 43718 [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] chan_sip.c: Adding codec 100004 (alaw) to SDP [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] chan_sip.c: Adding codec 100008 (g729) to SDP [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: -- Done with adding codecs to SDP [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Done building SDP. Settling with this capability: (alaw|g729) [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Initializing initreq for method INVITE - callid 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 0 [ 42]: INVITE sip:u6-2@mydomain.com SIP/2.0 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1ee09173;rport [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 3 [ 64]: From: "37233" ;tag=as606c66b7 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 4 [ 33]: To: [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 5 [ 39]: Contact: [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 6 [ 60]: Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 11.22.0 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:11:49 GMT [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 10 [ 93]: Replaces: 8a52df2a-8e60a45-4ff57988@10.11.198.230;to-tag=963635824;from-tag=F680C2CD-244325D0 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 11 [ 17]: Require: replaces [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 12 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 14 [ 53]: Remote-Party-ID: 37233 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 15 [ 57]: P-Asserted-Identity: 37233 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] chan_sip.c: Reliably Transmitting (NAT) to 10.0.0.37:5060: INVITE sip:u6-2@mydomain.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1ee09173;rport Max-Forwards: 70 From: "37233" ;tag=as606c66b7 To: Contact: Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com CSeq: 102 INVITE User-Agent: Asterisk PBX 11.22.0 Date: Mon, 20 Jun 2016 11:11:49 GMT Replaces: 8a52df2a-8e60a45-4ff57988@10.11.198.230;to-tag=963635824;from-tag=F680C2CD-244325D0 Require: replaces Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: 37233 P-Asserted-Identity: 37233 Content-Type: application/sdp Content-Length: 283 v=0 o=root 101083964 101083964 IN IP4 10.0.0.31 s=Asterisk PBX 11.22.0 c=IN IP4 10.0.0.31 t=0 0 m=audio 43718 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.0.37:5060 [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] app_dial.c: -- Called SIP/NB-out/u6-2 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e340166f8' so dropping frame [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1ee09173;rport=5100 From: "37233" ;tag=as606c66b7 To: Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com CSeq: 102 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 <-------------> [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1ee09173;rport=5100 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as606c66b7 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 3 [ 33]: To: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 6 [ 42]: Server: OpenSER (1.2.2-notls (i386/linux)) [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 8 [ 0]: [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: --- (8 headers 0 lines) --- [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com' Request 102: Found [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: SIP response 100 to standard invite [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK60bdbcb1;rport=5100 From: "37233" ;tag=as368c8ffe To: "userB" ;tag=1E252C18-715B31C3 CSeq: 103 NOTIFY Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com Contact: Record-Route: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 Content-Length: 0 <-------------> [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK60bdbcb1;rport=5100 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as368c8ffe [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 3 [ 84]: To: "userB" ;tag=1E252C18-715B31C3 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 4 [ 16]: CSeq: 103 NOTIFY [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 5 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 6 [ 33]: Contact: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 7 [ 61]: Record-Route: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 8 [ 17]: Event: refer;id=2 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 10 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 12 [ 0]: [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: --- (12 headers 0 lines) --- [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Stopping retransmission on '627d96dc204c343679fb0b126b8ad013@mydomain.com' of Request 103: Match Found [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e340166f8' so dropping frame [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5070 ---> BYE sip:37233@10.0.0.31:5100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK86af.d10a8861.0 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK86af.c10a8861.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK86af.bee4aec7.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK11ccebab1D8CD9BE From: "userB" ;tag=1E252C18-715B31C3 To: "37233" ;tag=as368c8ffe CSeq: 3 BYE Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com Contact: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 Max-Forwards: 11 Content-Length: 0 <-------------> [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 0 [ 40]: BYE sip:37233@10.0.0.31:5100 SIP/2.0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK86af.d10a8861.0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 2 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK86af.c10a8861.0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 3 [ 60]: Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK86af.bee4aec7.0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 4 [ 72]: Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK11ccebab1D8CD9BE [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 5 [ 86]: From: "userB" ;tag=1E252C18-715B31C3 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 6 [ 62]: To: "37233" ;tag=as368c8ffe [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 7 [ 11]: CSeq: 3 BYE [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 8 [ 60]: Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 9 [ 33]: Contact: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 10 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 11 [ 16]: Max-Forwards: 11 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 13 [ 0]: [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Initializing initreq for method BYE - callid 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:49] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.37:5070' into... [Jun 20 13:11:49] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.37' and port '5070'. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Sending to 10.0.0.37:5070 (NAT) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Setting SIP_ALREADYGONE on dialog 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:11:49] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e34037868' [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Scheduling destruction of SIP dialog '627d96dc204c343679fb0b126b8ad013@mydomain.com' in 32000 ms (Method: BYE) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Received bye, no owner, selfdestruct soon. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.37:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK86af.d10a8861.0;received=10.0.0.37;rport=5070 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK86af.c10a8861.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK86af.bee4aec7.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK11ccebab1D8CD9BE From: "userB" ;tag=1E252C18-715B31C3 To: "37233" ;tag=as368c8ffe Call-ID: 627d96dc204c343679fb0b126b8ad013@mydomain.com CSeq: 3 BYE Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: No remote address on RTP instance '0x7f2e340166f8' so dropping frame [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1ee09173;rport=5100 Record-Route: Record-Route: Record-Route: Record-Route: Record-Route: From: "37233" ;tag=as606c66b7 To: ;tag=1953874510 Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2130 1.0.5.18 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 242 P-RTP-Proxy: Yes P-Asserted-Identity: "userC" v=0 o=userC 8002 8000 IN IP4 10.11.199.248 s=SIP Call c=IN IP4 10.0.0.174 t=0 0 m=audio 35102 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes <-------------> [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1ee09173;rport=5100 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 2 [ 61]: Record-Route: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 3 [ 60]: Record-Route: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Record-Route: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 5 [ 60]: Record-Route: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 6 [ 61]: Record-Route: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 7 [ 64]: From: "37233" ;tag=as606c66b7 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 8 [ 48]: To: ;tag=1953874510 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 9 [ 60]: Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 10 [ 16]: CSeq: 102 INVITE [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 11 [ 33]: Contact: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 12 [ 32]: Supported: replaces, path, timer [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 13 [ 40]: User-Agent: Grandstream GXP2130 1.0.5.18 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 14 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 16 [ 19]: Content-Length: 242 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 17 [ 16]: P-RTP-Proxy: Yes [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 18 [ 67]: P-Asserted-Identity: "userC" [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Header 19 [ 0]: [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 0 [ 3]: v=0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 1 [ 46]: o=userC 8002 8000 IN IP4 10.11.199.248 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 2 [ 10]: s=SIP Call [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 3 [ 23]: c=IN IP4 10.0.0.174 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 5 [ 27]: m=audio 35102 RTP/AVP 8 101 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 6 [ 10]: a=sendrecv [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Jun 20 13:11:49] DEBUG[13372] chan_sip.c: Body 11 [ 16]: a=nortpproxy:yes [Jun 20 13:11:49] VERBOSE[13372] chan_sip.c: --- (19 headers 12 lines) --- [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Acked pending invite 102 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Stopping retransmission on '19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com' of Request 102: Match Found [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: SIP response 200 to standard invite [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP o=userC 8002 8000 IN IP4 10.11.199.248... OK. [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Jun 20 13:11:49] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.174' into... [Jun 20 13:11:49] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.174' and port ''. [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.174... OK. [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Found RTP audio format 8 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] rtp_engine.c: Setting payload 8 based on m type on 0x7f2e32784a00 [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Found RTP audio format 101 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] rtp_engine.c: Setting payload 101 based on m type on 0x7f2e32784a00 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Processing media-level (audio) SDP a=nortpproxy:yes... UNSUPPORTED OR FAILED. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Set role to CONTROLLING (0x7f2e340166f8) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Set role failed; no ice instance (0x7f2e340166f8) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e340166f8' [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Peer audio RTP is at port 10.0.0.174:35102 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] rtp_engine.c: Copying payload 8 from 0x7f2e32784a00 to 0x7f2e340168c0 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] rtp_engine.c: Copying payload 101 from 0x7f2e32784a00 to 0x7f2e340168c0 [Jun 20 13:11:49] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f2e340166f8' [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: We're settling with these formats: (alaw) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: We have an owner, now see if we need to change this call [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: build_route: Record-Route hop: [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: list_route: hop: [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 20 13:11:49] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.37' into... [Jun 20 13:11:49] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.37' and port ''. [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: set_destination: set destination to 10.0.0.37:5060 [Jun 20 13:11:49] VERBOSE[13372][C-0000000d] chan_sip.c: Transmitting (NAT) to 10.0.0.37:5060: ACK sip:u6-2@10.0.0.37 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK64304881;rport Route: ,,,, Max-Forwards: 70 From: "37233" ;tag=as606c66b7 To: ;tag=1953874510 Contact: Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com CSeq: 102 ACK User-Agent: Asterisk PBX 11.22.0 Content-Length: 0 --- [Jun 20 13:11:49] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'ACK sip:u6-' onto UDP socket destined for 10.0.0.37:5060 [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] app_dial.c: -- SIP/NB-out-00000027 answered SIP/ISP_SIPrunk-00000025 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] features.c: bridge answer set, chan answer set [Jun 20 13:11:49] DEBUG[19952][C-0000000d] features.c: Removing dialed interfaces datastore on SIP/NB-out-00000027 since we're bridging [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:11:49] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - NB-out [Jun 20 13:11:49] DEBUG[13365] chan_sip.c: Checking device state for peer NB-out [Jun 20 13:11:49] DEBUG[13365] devicestate.c: Changing state for SIP/NB-out - state 1 (Not in use) [Jun 20 13:11:49] DEBUG[13365] devicestate.c: device 'SIP/NB-out' state '1' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f2e340166f8' [Jun 20 13:11:49] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: 0x7f2e3403ca30 -- Probation learning mode pass with source address 10.0.0.174:35102 [Jun 20 13:11:49] VERBOSE[19952][C-0000000d] res_rtp_asterisk.c: > 0x7f2e3403ca30 -- Probation passed - setting RTP source address to 10.0.0.174:35102 [Jun 20 13:11:50] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:11:55] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Allocating new SIP dialog for 645c63c14bbe67825e4e4a92002d158a@mydomain.com - OPTIONS (No RTP) [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:11:58] DEBUG[13372] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: SIP call-id changed from '645c63c14bbe67825e4e4a92002d158a@mydomain.com' to '13207cc453a7315643839bef18307dc1@mydomain.com' [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Initializing initreq for method OPTIONS - callid 13207cc453a7315643839bef18307dc1@mydomain.com [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK602a2a4f;rport [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as449b7440 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 6 [ 60]: Call-ID: 13207cc453a7315643839bef18307dc1@mydomain.com [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 11.22.0 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:11:58 GMT [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:11:58] VERBOSE[13372] chan_sip.c: Reliably Transmitting (NAT) to 10.0.0.9:5060: OPTIONS sip:10.0.0.9 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK602a2a4f;rport Max-Forwards: 70 From: "asterisk" ;tag=as449b7440 To: Contact: Call-ID: 13207cc453a7315643839bef18307dc1@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.22.0 Date: Mon, 20 Jun 2016 11:11:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:11:58] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK602a2a4f;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as449b7440 To: ;tag=as58fbfaf8 Call-ID: 13207cc453a7315643839bef18307dc1@mydomain.com CSeq: 102 OPTIONS Server: Digium Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK602a2a4f;received=10.0.0.31;rport=5100 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as449b7440 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 3 [ 37]: To: ;tag=as58fbfaf8 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Call-ID: 13207cc453a7315643839bef18307dc1@mydomain.com [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Header 11 [ 0]: [Jun 20 13:11:58] VERBOSE[13372] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Stopping retransmission on '13207cc453a7315643839bef18307dc1@mydomain.com' of Request 102: Match Found [Jun 20 13:11:58] DEBUG[13372] chan_sip.c: Destroying SIP dialog 13207cc453a7315643839bef18307dc1@mydomain.com [Jun 20 13:11:58] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '13207cc453a7315643839bef18307dc1@mydomain.com' Method: OPTIONS [Jun 20 13:12:00] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:12:01] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> BYE sip:23830@10.0.0.31:5100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK4b3169a9 Max-Forwards: 70 From: "37233" ;tag=as3b6a65c5 To: ;tag=as5d0bd156 Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com CSeq: 103 BYE User-Agent: Digium Gateway X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 0 [ 40]: BYE sip:23830@10.0.0.31:5100 SIP/2.0 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK4b3169a9 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 3 [ 57]: From: "37233" ;tag=as3b6a65c5 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 4 [ 52]: To: ;tag=as5d0bd156 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 5 [ 58]: Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 6 [ 13]: CSeq: 103 BYE [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 7 [ 26]: User-Agent: Digium Gateway [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 11 [ 0]: [Jun 20 13:12:01] VERBOSE[13372] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:12:01] DEBUG[13372][C-0000000d] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jun 20 13:12:01] DEBUG[13372][C-0000000d] chan_sip.c: Initializing initreq for method BYE - callid 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:12:01] DEBUG[13372][C-0000000d] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:12:01] DEBUG[13372][C-0000000d] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:12:01] VERBOSE[13372][C-0000000d] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:12:01] DEBUG[13372][C-0000000d] chan_sip.c: Setting SIP_ALREADYGONE on dialog 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:12:01] DEBUG[13372][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e44007f68' [Jun 20 13:12:01] DEBUG[13372][C-0000000d] chan_sip.c: Session timer stopped: 8 - 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:12:01] VERBOSE[13372][C-0000000d] chan_sip.c: Scheduling destruction of SIP dialog '111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com' in 6400 ms (Method: BYE) [Jun 20 13:12:01] DEBUG[13372][C-0000000d] chan_sip.c: Received bye, issuing owner hangup [Jun 20 13:12:01] VERBOSE[13372][C-0000000d] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.9:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK4b3169a9;received=10.0.0.9;rport=5060 From: "37233" ;tag=as3b6a65c5 To: ;tag=as5d0bd156 Call-ID: 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com CSeq: 103 BYE Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jun 20 13:12:01] DEBUG[13372][C-0000000d] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:12:01] DEBUG[19952][C-0000000d] channel.c: Didn't get a frame from channel: SIP/ISP_SIPrunk-00000025 [Jun 20 13:12:01] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:12:01] DEBUG[19952][C-0000000d] channel.c: Bridge stops bridging channels SIP/ISP_SIPrunk-00000025 and SIP/NB-out-00000027 [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(clid)' (from 'CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 9) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(clid) result is '"37233" <37233>' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 8) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(src) result is '37233' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 8) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(dst) result is 'u6-2' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(dcontext) result is 'station_transferUC' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 12) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(channel) result is 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 15) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(dstchannel) result is 'SIP/NB-out-00000027' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 12) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(lastapp) result is 'Dial' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(lastdata) result is 'SIP/NB-out/u6-2,,t' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 10) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(start) result is '2016-06-20 13:11:49' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 11) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(answer) result is '2016-06-20 13:11:49' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 8) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(end) result is '2016-06-20 13:12:01' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(duration)' (from 'CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(duration) result is '12' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(billsec)' (from 'CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 12) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(billsec) result is '12' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 16) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 16) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(accountcode) result is '(null)' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)}","${CDR(userfield)}","${CDR(privacy-id)}" ' len 13) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(uniqueid) result is 'asterisk-ibercom-1466421095.37' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)}","${CDR(privacy-id)}" ' len 14) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(userfield) result is '(null)' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Evaluating 'CDR(privacy-id)' (from 'CDR(privacy-id)}" ' len 15) [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Function CDR(privacy-id) result is '"Privacy: off"' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] channel.c: Hanging up channel 'SIP/NB-out-00000027' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] chan_sip.c: Hangup call SIP/NB-out-00000027, SIP callid 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com [Jun 20 13:12:01] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e340166f8' [Jun 20 13:12:01] VERBOSE[19952][C-0000000d] chan_sip.c: Scheduling destruction of SIP dialog '19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com' in 32000 ms (Method: INVITE) [Jun 20 13:12:01] VERBOSE[19952][C-0000000d] chan_sip.c: set_destination: Parsing for address/port to send to [Jun 20 13:12:01] DEBUG[19952][C-0000000d] netsock2.c: Splitting '10.0.0.37' into... [Jun 20 13:12:01] DEBUG[19952][C-0000000d] netsock2.c: ...host '10.0.0.37' and port ''. [Jun 20 13:12:01] VERBOSE[19952][C-0000000d] chan_sip.c: set_destination: set destination to 10.0.0.37:5060 [Jun 20 13:12:01] VERBOSE[19952][C-0000000d] chan_sip.c: Reliably Transmitting (NAT) to 10.0.0.37:5060: BYE sip:u6-2@10.0.0.37 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK5b31ea7e;rport Route: ,,,, Max-Forwards: 70 From: "37233" ;tag=as606c66b7 To: ;tag=1953874510 Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com CSeq: 103 BYE User-Agent: Asterisk PBX 11.22.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jun 20 13:12:01] DEBUG[19952][C-0000000d] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Jun 20 13:12:01] DEBUG[19952][C-0000000d] chan_sip.c: Trying to put 'BYE sip:u6-' onto UDP socket destined for 10.0.0.37:5060 [Jun 20 13:12:01] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - NB-out [Jun 20 13:12:01] DEBUG[13365] chan_sip.c: Checking device state for peer NB-out [Jun 20 13:12:01] DEBUG[13365] devicestate.c: Changing state for SIP/NB-out - state 1 (Not in use) [Jun 20 13:12:01] DEBUG[13365] devicestate.c: device 'SIP/NB-out' state '1' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jun 20 13:12:01] DEBUG[19952][C-0000000d] app_macro.c: Spawn extension (macro-transferUCAAi,s,6) exited non-zero on 'SIP/ISP_SIPrunk-00000025' in macro 'transferUCAAi' [Jun 20 13:12:01] VERBOSE[19952][C-0000000d] app_macro.c: == Spawn extension (macro-transferUCAAi, s, 6) exited non-zero on 'SIP/ISP_SIPrunk-00000025' in macro 'transferUCAAi' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] pbx.c: Spawn extension (station_transferUC,u6-2,1) exited non-zero on 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:12:01] VERBOSE[19952][C-0000000d] pbx.c: == Spawn extension (station_transferUC, u6-2, 1) exited non-zero on 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] channel.c: Soft-Hanging up channel 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] channel.c: Hanging up channel 'SIP/ISP_SIPrunk-00000025' [Jun 20 13:12:01] DEBUG[19952][C-0000000d] chan_sip.c: Hangup call SIP/ISP_SIPrunk-00000025, SIP callid 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:12:01] DEBUG[19952][C-0000000d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2e44007f68' [Jun 20 13:12:01] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - ISP_SIPrunk [Jun 20 13:12:01] DEBUG[13365] chan_sip.c: Checking device state for peer ISP_SIPrunk [Jun 20 13:12:01] DEBUG[13365] devicestate.c: Changing state for SIP/ISP_SIPrunk - state 1 (Not in use) [Jun 20 13:12:01] DEBUG[13365] devicestate.c: device 'SIP/ISP_SIPrunk' state '1' [Jun 20 13:12:01] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK5b31ea7e;rport=5100 From: "37233" ;tag=as606c66b7 To: ;tag=1953874510 Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com CSeq: 103 BYE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2130 1.0.5.18 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK5b31ea7e;rport=5100 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as606c66b7 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 3 [ 48]: To: ;tag=1953874510 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Call-ID: 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 6 [ 33]: Contact: [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 7 [ 32]: Supported: replaces, path, timer [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 8 [ 40]: User-Agent: Grandstream GXP2130 1.0.5.18 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 9 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Header 11 [ 0]: [Jun 20 13:12:01] VERBOSE[13372] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:12:01] DEBUG[13372][C-0000000d] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Jun 20 13:12:01] DEBUG[13372][C-0000000d] chan_sip.c: Stopping retransmission on '19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com' of Request 103: Match Found [Jun 20 13:12:01] DEBUG[13372] chan_sip.c: Destroying SIP dialog 19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com [Jun 20 13:12:01] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '19bc20cf1e09da3b0ba4d0081bc29ac2@mydomain.com' Method: INVITE [Jun 20 13:12:01] DEBUG[13372] rtp_engine.c: Destroyed RTP instance '0x7f2e340166f8' [Jun 20 13:12:08] DEBUG[13372] chan_sip.c: Auto destroying SIP dialog '111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com' [Jun 20 13:12:08] DEBUG[13372] chan_sip.c: Destroying SIP dialog 111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com [Jun 20 13:12:08] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '111cf79c06635fd9201e410e05a7d34f@asterisk.mydomain.com' Method: BYE [Jun 20 13:12:08] DEBUG[13372] rtp_engine.c: Destroyed RTP instance '0x7f2e44007f68' [Jun 20 13:12:08] DEBUG[13372] chan_sip.c: Auto destroying SIP dialog '3c2d1b3878b2b7a768073e3d1cf04691@10.0.0.9:5060' [Jun 20 13:12:08] DEBUG[13372] chan_sip.c: Destroying SIP dialog 3c2d1b3878b2b7a768073e3d1cf04691@10.0.0.9:5060 [Jun 20 13:12:08] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '3c2d1b3878b2b7a768073e3d1cf04691@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:12:10] VERBOSE[19951] asterisk.c: -- Remote UNIX connection disconnected [Jun 20 13:12:21] DEBUG[13372] chan_sip.c: Auto destroying SIP dialog '627d96dc204c343679fb0b126b8ad013@mydomain.com' [Jun 20 13:12:21] DEBUG[13372] chan_sip.c: Destroying SIP dialog 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:12:21] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '627d96dc204c343679fb0b126b8ad013@mydomain.com' Method: BYE [Jun 20 13:12:21] DEBUG[13372] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:12:21] DEBUG[13372] chan_sip.c: Call to peer 'PROXY-out' removed from call limit 0 [Jun 20 13:12:21] DEBUG[13372] chan_sip.c: This call did not properly clean up call limits. Call ID 627d96dc204c343679fb0b126b8ad013@mydomain.com [Jun 20 13:12:21] DEBUG[13372] rtp_engine.c: Destroyed RTP instance '0x7f2e34037868' [Jun 20 13:12:21] DEBUG[13365] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:12:21] DEBUG[13365] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:12:21] DEBUG[13365] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:12:21] DEBUG[13365] devicestate.c: device 'SIP/PROXY-out' state '1' [Jun 20 13:12:36] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> OPTIONS sip:asterisk.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK17638145 Max-Forwards: 70 From: "asterisk" ;tag=as5813afc4 To: Contact: Call-ID: 1336f8c16c32f3b937eb6b0b44a6e4e2@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:12:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK17638145 [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5813afc4 [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 6 [ 59]: Call-ID: 1336f8c16c32f3b937eb6b0b44a6e4e2@10.0.0.9:5060 [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:12:36 GMT [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Header 13 [ 0]: [Jun 20 13:12:36] VERBOSE[13372] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:12:36] DEBUG[13372] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:12:36] DEBUG[13372] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:12:36] DEBUG[13372] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:12:36] VERBOSE[13372] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Allocating new SIP dialog for 1336f8c16c32f3b937eb6b0b44a6e4e2@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:12:36] DEBUG[13372] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:12:36] DEBUG[13372] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:12:36] DEBUG[13372] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:12:36] DEBUG[13372] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:12:36] VERBOSE[13372] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:12:36] VERBOSE[13372] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.9:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK17638145;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as5813afc4 To: ;tag=as19b862a2 Call-ID: 1336f8c16c32f3b937eb6b0b44a6e4e2@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:12:36] DEBUG[13372] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:12:36] VERBOSE[13372] chan_sip.c: Scheduling destruction of SIP dialog '1336f8c16c32f3b937eb6b0b44a6e4e2@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Allocating new SIP dialog for 22d1b7435834870439e2b5095f25a3cb@mydomain.com - OPTIONS (No RTP) [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:12:58] DEBUG[13372] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: SIP call-id changed from '22d1b7435834870439e2b5095f25a3cb@mydomain.com' to '7f8845320f0e9cee20247f754967929f@mydomain.com' [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Initializing initreq for method OPTIONS - callid 7f8845320f0e9cee20247f754967929f@mydomain.com [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK6fe8e07d;rport [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as6d7ce33a [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 6 [ 60]: Call-ID: 7f8845320f0e9cee20247f754967929f@mydomain.com [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 11.22.0 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:12:58 GMT [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:12:58] VERBOSE[13372] chan_sip.c: Reliably Transmitting (NAT) to 10.0.0.9:5060: OPTIONS sip:10.0.0.9 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK6fe8e07d;rport Max-Forwards: 70 From: "asterisk" ;tag=as6d7ce33a To: Contact: Call-ID: 7f8845320f0e9cee20247f754967929f@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.22.0 Date: Mon, 20 Jun 2016 11:12:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:12:58] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK6fe8e07d;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as6d7ce33a To: ;tag=as01585571 Call-ID: 7f8845320f0e9cee20247f754967929f@mydomain.com CSeq: 102 OPTIONS Server: Digium Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK6fe8e07d;received=10.0.0.31;rport=5100 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as6d7ce33a [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 3 [ 37]: To: ;tag=as01585571 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Call-ID: 7f8845320f0e9cee20247f754967929f@mydomain.com [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Header 11 [ 0]: [Jun 20 13:12:58] VERBOSE[13372] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Stopping retransmission on '7f8845320f0e9cee20247f754967929f@mydomain.com' of Request 102: Match Found [Jun 20 13:12:58] DEBUG[13372] chan_sip.c: Destroying SIP dialog 7f8845320f0e9cee20247f754967929f@mydomain.com [Jun 20 13:12:58] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '7f8845320f0e9cee20247f754967929f@mydomain.com' Method: OPTIONS [Jun 20 13:13:08] DEBUG[13372] chan_sip.c: Auto destroying SIP dialog '1336f8c16c32f3b937eb6b0b44a6e4e2@10.0.0.9:5060' [Jun 20 13:13:08] DEBUG[13372] chan_sip.c: Destroying SIP dialog 1336f8c16c32f3b937eb6b0b44a6e4e2@10.0.0.9:5060 [Jun 20 13:13:08] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '1336f8c16c32f3b937eb6b0b44a6e4e2@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:13:36] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> OPTIONS sip:asterisk.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK2cdbddf2 Max-Forwards: 70 From: "asterisk" ;tag=as5c0b9d63 To: Contact: Call-ID: 7dc766621a749ae61bccf0572118393a@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:13:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK2cdbddf2 [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5c0b9d63 [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 6 [ 59]: Call-ID: 7dc766621a749ae61bccf0572118393a@10.0.0.9:5060 [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:13:36 GMT [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Header 13 [ 0]: [Jun 20 13:13:36] VERBOSE[13372] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:13:36] DEBUG[13372] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:13:36] DEBUG[13372] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:13:36] DEBUG[13372] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:13:36] VERBOSE[13372] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Allocating new SIP dialog for 7dc766621a749ae61bccf0572118393a@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:13:36] DEBUG[13372] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:13:36] DEBUG[13372] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:13:36] DEBUG[13372] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:13:36] DEBUG[13372] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:13:36] VERBOSE[13372] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:13:36] VERBOSE[13372] chan_sip.c: <--- Transmitting (NAT) to 10.0.0.9:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK2cdbddf2;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as5c0b9d63 To: ;tag=as6e758f5e Call-ID: 7dc766621a749ae61bccf0572118393a@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 11.22.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:13:36] DEBUG[13372] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:13:36] VERBOSE[13372] chan_sip.c: Scheduling destruction of SIP dialog '7dc766621a749ae61bccf0572118393a@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Allocating new SIP dialog for 2724fcb20ab685312384d5425cdfc385@mydomain.com - OPTIONS (No RTP) [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:13:58] DEBUG[13372] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: SIP call-id changed from '2724fcb20ab685312384d5425cdfc385@mydomain.com' to '228c3612633b80657777b4447f2c63d0@mydomain.com' [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Initializing initreq for method OPTIONS - callid 228c3612633b80657777b4447f2c63d0@mydomain.com [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK12f20df8;rport [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as0706c6c3 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 6 [ 60]: Call-ID: 228c3612633b80657777b4447f2c63d0@mydomain.com [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 11.22.0 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:13:58 GMT [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:13:58] VERBOSE[13372] chan_sip.c: Reliably Transmitting (NAT) to 10.0.0.9:5060: OPTIONS sip:10.0.0.9 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK12f20df8;rport Max-Forwards: 70 From: "asterisk" ;tag=as0706c6c3 To: Contact: Call-ID: 228c3612633b80657777b4447f2c63d0@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.22.0 Date: Mon, 20 Jun 2016 11:13:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:13:58] VERBOSE[13372] chan_sip.c: <--- SIP read from UDP:10.0.0.9:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK12f20df8;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as0706c6c3 To: ;tag=as7ff26a27 Call-ID: 228c3612633b80657777b4447f2c63d0@mydomain.com CSeq: 102 OPTIONS Server: Digium Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK12f20df8;received=10.0.0.31;rport=5100 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as0706c6c3 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 3 [ 37]: To: ;tag=as7ff26a27 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 4 [ 60]: Call-ID: 228c3612633b80657777b4447f2c63d0@mydomain.com [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Header 11 [ 0]: [Jun 20 13:13:58] VERBOSE[13372] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Stopping retransmission on '228c3612633b80657777b4447f2c63d0@mydomain.com' of Request 102: Match Found [Jun 20 13:13:58] DEBUG[13372] chan_sip.c: Destroying SIP dialog 228c3612633b80657777b4447f2c63d0@mydomain.com [Jun 20 13:13:58] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '228c3612633b80657777b4447f2c63d0@mydomain.com' Method: OPTIONS [Jun 20 13:14:08] DEBUG[13372] chan_sip.c: Auto destroying SIP dialog '7dc766621a749ae61bccf0572118393a@10.0.0.9:5060' [Jun 20 13:14:08] DEBUG[13372] chan_sip.c: Destroying SIP dialog 7dc766621a749ae61bccf0572118393a@10.0.0.9:5060 [Jun 20 13:14:08] VERBOSE[13372] chan_sip.c: Really destroying SIP dialog '7dc766621a749ae61bccf0572118393a@10.0.0.9:5060' Method: OPTIONS