[Jun 20 13:23:46] Asterisk 13.9.1 built by root @ asterisk on a x86_64 running Linux on 2016-05-18 10:03:05 UTC [Jun 20 13:23:46] DEBUG[21341] config.c: Parsing /opt/asterisk-ibercom/etc/asterisk/logger.conf [Jun 20 13:23:46] VERBOSE[21341] config.c: Parsing '/opt/asterisk-ibercom/etc/asterisk/logger.conf': Found [Jun 20 13:23:46] VERBOSE[21341] logger.c: Asterisk Queue Logger restarted [Jun 20 13:23:49] VERBOSE[21156] 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=z9hG4bK0045c8f8 Max-Forwards: 70 From: "asterisk" ;tag=as3b371ffb To: Contact: Call-ID: 3f97ac085f1663545e55d67c658e3c19@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:23:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK0045c8f8 [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as3b371ffb [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 3f97ac085f1663545e55d67c658e3c19@10.0.0.9:5060 [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:23:49 GMT [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:23:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:23:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:23:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:23:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:23:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 3f97ac085f1663545e55d67c658e3c19@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:23:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:23:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:23:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:23:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:23:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:23:49] VERBOSE[21156] 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=z9hG4bK0045c8f8;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as3b371ffb To: ;tag=as1009554b Call-ID: 3f97ac085f1663545e55d67c658e3c19@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:23:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:23:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '3f97ac085f1663545e55d67c658e3c19@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:23:53] VERBOSE[21156] 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=z9hG4bK5f6d5be8 Max-Forwards: 70 From: "37233" ;tag=as2a775170 To: Contact: Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com CSeq: 102 INVITE User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:23:53 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 378319983 378319983 IN IP4 10.0.0.9 s=Digium Gateway c=IN IP4 10.0.0.9 t=0 0 m=audio 10142 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 13:23:53] DEBUG[21156] chan_sip.c: Header 0 [ 46]: INVITE sip:23830@asterisk.mydomain.com:5100 SIP/2.0 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK5f6d5be8 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 3 [ 57]: From: "37233" ;tag=as2a775170 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 4 [ 37]: To: [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 5 [ 38]: Contact: [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 6 [ 58]: Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:23:53 GMT [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 12 [ 90]: Remote-Party-ID: "37233" ;party=calling;privacy=off;screen=yes [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 14 [ 19]: Content-Length: 255 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 15 [ 0]: [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 0 [ 3]: v=0 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 1 [ 46]: o=root 378319983 378319983 IN IP4 10.0.0.9 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 2 [ 16]: s=Digium Gateway [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.0.0.9 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 5 [ 27]: m=audio 10142 RTP/AVP 8 101 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Body 11 [ 10]: a=sendrecv [Jun 20 13:23:53] VERBOSE[21156] chan_sip.c: --- (15 headers 12 lines) --- [Jun 20 13:23:53] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:23:53] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:23:53] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:23:53] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com - INVITE (No RTP) [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 20 13:23:53] DEBUG[21156][C-00000002] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer" [Jun 20 13:23:53] DEBUG[21156][C-00000002] sip/reqresp_parser.c: Found SIP option: -replaces- [Jun 20 13:23:53] DEBUG[21156][C-00000002] sip/reqresp_parser.c: Matched SIP option: replaces [Jun 20 13:23:53] DEBUG[21156][C-00000002] sip/reqresp_parser.c: Found SIP option: -timer- [Jun 20 13:23:53] DEBUG[21156][C-00000002] sip/reqresp_parser.c: Matched SIP option: timer [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Using INVITE request as basis request - 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Found peer 'ISP_SIPtrunk' for '37233' from 10.0.0.9:5060 [Jun 20 13:23:53] DEBUG[21156][C-00000002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f2ec401c358' [Jun 20 13:23:53] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Allocated port 49560 for RTP instance '0x7f2ec401c358' [Jun 20 13:23:53] DEBUG[21156][C-00000002] rtp_engine.c: RTP instance '0x7f2ec401c358' is setup and ready to go [Jun 20 13:23:53] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f2ec401c358' [Jun 20 13:23:53] VERBOSE[21156][C-00000002] netsock2.c: Using SIP RTP TOS bits 184 [Jun 20 13:23:53] VERBOSE[21156][C-00000002] netsock2.c: Using SIP RTP CoS mark 5 [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Setting NAT on RTP to On [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP o=root 378319983 378319983 IN IP4 10.0.0.9... OK. [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP s=Digium Gateway... UNSUPPORTED OR FAILED. [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.9... OK. [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Found RTP audio format 8 [Jun 20 13:23:53] DEBUG[21156][C-00000002] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f2ed8180fb0 [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Found RTP audio format 101 [Jun 20 13:23:53] DEBUG[21156][C-00000002] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f2ed8180fb0 [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED. [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 20 13:23:53] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2ec401c358' [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Peer audio RTP is at port 10.0.0.9:10142 [Jun 20 13:23:53] DEBUG[21156][C-00000002] rtp_engine.c: Copying payload 8 (0x7f2ec4026ba0) from 0x7f2ed8180fb0 to 0x7f2ec401c520 [Jun 20 13:23:53] DEBUG[21156][C-00000002] rtp_engine.c: Copying payload 101 (0x7f2ec4018f80) from 0x7f2ed8180fb0 to 0x7f2ec401c520 [Jun 20 13:23:53] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f2ec401c358' [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: We're settling with these formats: (alaw) [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Checking SIP call limits for device [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Updating call counter for incoming call [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: Splitting 'asterisk.mydomain.com:5100' into... [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:23:53] DEBUG[21156][C-00000002] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:23:53] VERBOSE[21156][C-00000002] chan_sip.c: Looking for 23830 in incoming (domain asterisk.mydomain.com) [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Incoming INVITE with 'timer' option supported [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: *** Our native formats are (alaw) [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: *** Joint capabilities are (alaw) [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: *** Our capabilities are (alaw|g729) [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: This channel will not be able to handle video. [Jun 20 13:23:53] VERBOSE[21156][C-00000002] sip/route.c: sip_route_dump: route/path hop: [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: SIP/ISP_SIPtrunk-00000004: New call is still down.... Trying... [Jun 20 13:23:53] VERBOSE[21156][C-00000002] 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=z9hG4bK5f6d5be8;received=10.0.0.9;rport=5060 From: "37233" ;tag=as2a775170 To: Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com CSeq: 102 INVITE Server: Asterisk PBX 13.9.1 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 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:23:53] DEBUG[21133] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jun 20 13:23:53] DEBUG[21145] devicestate.c: No provider found, checking channel drivers for SIP - ISP_SIPtrunk [Jun 20 13:23:53] DEBUG[21145] chan_sip.c: Checking device state for peer ISP_SIPtrunk [Jun 20 13:23:53] DEBUG[21145] devicestate.c: Changing state for SIP/ISP_SIPtrunk - state 1 (Not in use) [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'EXTEN' is '23830' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Function CALLERID(num) result is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Macro' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [23830@incoming:1] Macro("SIP/ISP_SIPtrunk-00000004", "llamadaEntranteUC,23830,37233") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'CALLINGPRES' is '1' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Function CALLERID(all) result is '"37233" <37233>' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'NoOp' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:1] NoOp("SIP/ISP_SIPtrunk-00000004", "CallingPress --1-- caller id --"37233" <37233>--") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: NoOp [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Set' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:2] Set("SIP/ISP_SIPtrunk-00000004", "TRANSFER_CONTEXT=station_transferUC") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Set [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Set' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:3] Set("SIP/ISP_SIPtrunk-00000004", "_TRANSFER_CONTEXT=station_transferUC") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Set [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Set' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:4] Set("SIP/ISP_SIPtrunk-00000004", "__TRANSFER_CONTEXT=station_transferUC") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Set [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'ARG1' is '23830' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Set' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:5] Set("SIP/ISP_SIPtrunk-00000004", "NUMCALLED=23830") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Set [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'ARG2' is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Set' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:6] Set("SIP/ISP_SIPtrunk-00000004", "NUMCALLER=37233") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Set [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Expression result is '0' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'GotoIf' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:7] GotoIf("SIP/ISP_SIPtrunk-00000004", "0?desconegut:conegut") in new stack [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx_builtins.c: Goto (macro-llamadaEntranteUC,s,10) [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: GotoIf [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Expression result is '1' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Function LEN(37233) result is '5' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Expression result is '0' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Expression result is '0' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'GotoIf' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:10] GotoIf("SIP/ISP_SIPtrunk-00000004", "0?addZeroPrefix") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_builtins.c: Not taking any branch [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: GotoIf [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Goto' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:11] Goto("SIP/ISP_SIPtrunk-00000004", "s,nextZeroPrefix") in new stack [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx_builtins.c: Goto (macro-llamadaEntranteUC,s,13) [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Goto [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'NUMCALLED' is '23830' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Set' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:13] Set("SIP/ISP_SIPtrunk-00000004", "_UCUSER=23830") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Set [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Set' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:14] Set("SIP/ISP_SIPtrunk-00000004", "CDR(PRIVACY-ID)="Privacy: off"") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Set [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Function CALLERID(name) result is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Set' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:15] Set("SIP/ISP_SIPtrunk-00000004", "CALLERID(all)=37233 <37233>") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: Set [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Function CALLERID(name) result is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'DOMINI' is 'mydomain.com' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'SIPAddHeader' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:16] SIPAddHeader("SIP/ISP_SIPtrunk-00000004", "P-Asserted-Identity: 37233 ") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: SIP Header added "P-Asserted-Identity: 37233 " as __SIPADDHEADER01 [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: SIPAddHeader [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'CALLINGPRES' is '1' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Function MATH(1>30) result is 'FALSE' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Expression result is '0' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'GotoIf' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:17] GotoIf("SIP/ISP_SIPtrunk-00000004", "0?privacy:privacyoff") in new stack [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx_builtins.c: Goto (macro-llamadaEntranteUC,s,22) [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: GotoIf [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Function CALLERID(name) result is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'NUMCALLER' is '37233' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'DOMINI' is 'mydomain.com' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'SIPAddHeader' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:22] SIPAddHeader("SIP/ISP_SIPtrunk-00000004", "Remote-Party-ID: 37233 ") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: SIP Header added "Remote-Party-ID: 37233 " as __SIPADDHEADER02 [Jun 20 13:23:53] DEBUG[21342][C-00000002] app_macro.c: Executed application: SIPAddHeader [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx_variables.c: Result of 'NUMCALLED' is '23830' [Jun 20 13:23:53] DEBUG[21342][C-00000002] pbx.c: Launching 'Dial' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] pbx.c: Executing [s@macro-llamadaEntranteUC:23] Dial("SIP/ISP_SIPtrunk-00000004", "SIP/PROXY-out/23830,,t") in new stack [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Allocating new SIP dialog for 3f8652bb1275397370c233b151a0c76d@mydomain.com - INVITE (No RTP) [Jun 20 13:23:53] DEBUG[21342][C-00000002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1fbd428' [Jun 20 13:23:53] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Allocated port 43472 for RTP instance '0x1fbd428' [Jun 20 13:23:53] DEBUG[21342][C-00000002] rtp_engine.c: RTP instance '0x1fbd428' is setup and ready to go [Jun 20 13:23:53] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x1fbd428' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] netsock2.c: Using SIP RTP TOS bits 184 [Jun 20 13:23:53] VERBOSE[21342][C-00000002] netsock2.c: Using SIP RTP CoS mark 5 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Setting NAT on RTP to On [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:23:53] DEBUG[21342][C-00000002] acl.c: For destination '10.0.0.37', our source address is '10.0.0.31'. [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Setting NAT on RTP to On [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: SIP call-id changed from '3f8652bb1275397370c233b151a0c76d@mydomain.com' to '7086c41e68689783760a32b471464e08@mydomain.com' [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: *** Our native formats are (alaw) [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: *** Joint capabilities are (alaw) [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: *** Our capabilities are (alaw|g729) [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: This channel will not be able to handle video. [Jun 20 13:23:53] DEBUG[21342][C-00000002] channel_internal_api.c: Channel Call ID changing from [C-00000002] to [C-00000002] [Jun 20 13:23:53] DEBUG[21342][C-00000002] channel.c: Inheriting variable __SIPADDHEADER02 from SIP/ISP_SIPtrunk-00000004 to SIP/PROXY-out-00000005. [Jun 20 13:23:53] DEBUG[21342][C-00000002] channel.c: Inheriting variable __SIPADDHEADER01 from SIP/ISP_SIPtrunk-00000004 to SIP/PROXY-out-00000005. [Jun 20 13:23:53] DEBUG[21342][C-00000002] channel.c: Inheriting variable UCUSER from SIP/ISP_SIPtrunk-00000004 to SIP/PROXY-out-00000005. [Jun 20 13:23:53] DEBUG[21342][C-00000002] channel.c: Inheriting variable __TRANSFER_CONTEXT from SIP/ISP_SIPtrunk-00000004 to SIP/PROXY-out-00000005. [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Outgoing Call for 23830 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Adding SIP Header "Remote-Party-ID" with content :37233 : [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Adding SIP Header "P-Asserted-Identity" with content :37233 : [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: ** Our capability: (alaw|g729) Video flag: False Text flag: False [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: ** Our prefcodec: (alaw) [Jun 20 13:23:53] VERBOSE[21342][C-00000002] chan_sip.c: Audio is at 43472 [Jun 20 13:23:53] VERBOSE[21342][C-00000002] chan_sip.c: Adding codec alaw to SDP [Jun 20 13:23:53] VERBOSE[21342][C-00000002] chan_sip.c: Adding codec g729 to SDP [Jun 20 13:23:53] VERBOSE[21342][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (alaw|g729) [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 0 [ 48]: INVITE sip:23830@mydomain.com:5070 SIP/2.0 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK219a38f4;rport [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 3 [ 64]: From: "37233" ;tag=as29fc0405 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 4 [ 39]: To: [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 5 [ 39]: Contact: [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 6 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:23:53 GMT [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 12 [ 53]: Remote-Party-ID: 37233 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 13 [ 57]: P-Asserted-Identity: 37233 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jun 20 13:23:53] VERBOSE[21342][C-00000002] 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=z9hG4bK219a38f4;rport Max-Forwards: 70 From: "37233" ;tag=as29fc0405 To: Contact: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 102 INVITE User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:23:53 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: 300 v=0 o=root 1200515678 1200515678 IN IP4 10.0.0.31 s=Asterisk PBX 13.9.1 c=IN IP4 10.0.0.31 t=0 0 m=audio 43472 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=maxptime:150 a=sendrecv --- [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Jun 20 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:23:53] VERBOSE[21342][C-00000002] app_dial.c: Called SIP/PROXY-out/23830 [Jun 20 13:23:53] DEBUG[21133] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jun 20 13:23:53] VERBOSE[21156] 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=z9hG4bK219a38f4;rport=5100 From: "37233" ;tag=as29fc0405 To: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 102 INVITE Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 <-------------> [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 0 [ 41]: SIP/2.0 100 Runing Incoming CPL script... [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK219a38f4;rport=5100 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as29fc0405 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 3 [ 39]: To: [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 6 [ 43]: Server: OpenSIPS (1.6.2-notls (i386/linux)) [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 8 [ 0]: [Jun 20 13:23:53] VERBOSE[21156] chan_sip.c: --- (8 headers 0 lines) --- [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: *** SIP TIMER: Cancelling retransmission #13 - INVITE (got response) [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7086c41e68689783760a32b471464e08@mydomain.com' Request 102: Found [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: SIP response 100 to standard invite [Jun 20 13:23:53] VERBOSE[21156] 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=z9hG4bK219a38f4;rport=5100 From: "37233" ;tag=as29fc0405 To: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 102 INVITE Server: OpenSIPS (1.6.2-notls (i386/linux)) Content-Length: 0 <-------------> [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK219a38f4;rport=5100 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as29fc0405 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 3 [ 39]: To: [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 6 [ 43]: Server: OpenSIPS (1.6.2-notls (i386/linux)) [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 8 [ 0]: [Jun 20 13:23:53] VERBOSE[21156] chan_sip.c: --- (8 headers 0 lines) --- [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7086c41e68689783760a32b471464e08@mydomain.com' Request 102: Found [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: SIP response 100 to standard invite [Jun 20 13:23:53] VERBOSE[21156] 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=z9hG4bK219a38f4;rport=5100 From: "37233" ;tag=as29fc0405 To: "userB" ;tag=C4769972-BA58EA4D CSeq: 102 INVITE Call-ID: 7086c41e68689783760a32b471464e08@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 13:23:53] DEBUG[21156] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK219a38f4;rport=5100 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as29fc0405 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 3 [ 84]: To: "userB" ;tag=C4769972-BA58EA4D [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 5 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 6 [ 33]: Contact: [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 7 [157]: Record-Route: , , [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 9 [ 34]: Allow-Events: talk,hold,conference [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 10 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jun 20 13:23:53] DEBUG[21156] chan_sip.c: Header 12 [ 0]: [Jun 20 13:23:53] VERBOSE[21156] chan_sip.c: --- (12 headers 0 lines) --- [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7086c41e68689783760a32b471464e08@mydomain.com' Request 102: Found [Jun 20 13:23:53] DEBUG[21156][C-00000002] chan_sip.c: SIP response 180 to standard invite [Jun 20 13:23:53] DEBUG[21156][C-00000002] sip/route.c: sip_route_process_header: [Jun 20 13:23:53] DEBUG[21156][C-00000002] sip/route.c: sip_route_process_header: [Jun 20 13:23:53] DEBUG[21156][C-00000002] sip/route.c: sip_route_process_header: [Jun 20 13:23:53] VERBOSE[21156][C-00000002] sip/route.c: sip_route_dump: route/path hop: [Jun 20 13:23:53] VERBOSE[21156][C-00000002] sip/route.c: sip_route_dump: route/path hop: [Jun 20 13:23:53] VERBOSE[21156][C-00000002] sip/route.c: sip_route_dump: route/path hop: [Jun 20 13:23:53] VERBOSE[21342][C-00000002] app_dial.c: SIP/PROXY-out-00000005 is ringing [Jun 20 13:23:53] DEBUG[21342][C-00000002] rtp_engine.c: Setting early bridge SDP of 'SIP/ISP_SIPtrunk-00000004' with that of 'SIP/PROXY-out-00000005' [Jun 20 13:23:53] VERBOSE[21342][C-00000002] 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=z9hG4bK5f6d5be8;received=10.0.0.9;rport=5060 From: "37233" ;tag=as2a775170 To: ;tag=as27b2cbfb Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com CSeq: 102 INVITE Server: Asterisk PBX 13.9.1 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 13:23:53] DEBUG[21342][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:23:53] DEBUG[21145] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:23:53] DEBUG[21145] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:23:53] DEBUG[21145] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:23:55] VERBOSE[21156] 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=z9hG4bK219a38f4;rport=5100 From: "37233" ;tag=as29fc0405 To: "userB" ;tag=C4769972-BA58EA4D CSeq: 102 INVITE Call-ID: 7086c41e68689783760a32b471464e08@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=- 1466366352 1466366352 IN IP4 10.11.198.230 s=Polycom IP Phone c=IN IP4 10.0.0.174 t=0 0 a=sendrecv m=audio 35116 RTP/AVP 8 127 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=nortpproxy:yes <-------------> [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK219a38f4;rport=5100 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as29fc0405 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 3 [ 84]: To: "userB" ;tag=C4769972-BA58EA4D [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 5 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 6 [ 33]: Contact: [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 7 [157]: Record-Route: , , [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 8 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 9 [ 26]: Supported: 100rel,replaces [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 10 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 11 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 13 [ 19]: Content-Length: 233 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 14 [ 16]: P-RTP-Proxy: Yes [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 15 [ 0]: [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 0 [ 3]: v=0 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 1 [ 46]: o=- 1466366352 1466366352 IN IP4 10.11.198.230 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 3 [ 23]: c=IN IP4 10.0.0.174 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 5 [ 10]: a=sendrecv [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 6 [ 27]: m=audio 35116 RTP/AVP 8 127 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 7 [ 10]: a=sendrecv [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 9 [ 33]: a=rtpmap:127 telephone-event/8000 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Body 10 [ 16]: a=nortpproxy:yes [Jun 20 13:23:55] VERBOSE[21156] chan_sip.c: --- (15 headers 11 lines) --- [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Acked pending invite 102 [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Stopping retransmission on '7086c41e68689783760a32b471464e08@mydomain.com' of Request 102: Match Found [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: SIP response 200 to standard invite [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP o=- 1466366352 1466366352 IN IP4 10.11.198.230... OK. [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Jun 20 13:23:55] DEBUG[21156][C-00000002] netsock2.c: Splitting '10.0.0.174' into... [Jun 20 13:23:55] DEBUG[21156][C-00000002] netsock2.c: ...host '10.0.0.174' and port ''. [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.174... OK. [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [Jun 20 13:23:55] VERBOSE[21156][C-00000002] chan_sip.c: Found RTP audio format 8 [Jun 20 13:23:55] DEBUG[21156][C-00000002] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f2ed81803e0 [Jun 20 13:23:55] VERBOSE[21156][C-00000002] chan_sip.c: Found RTP audio format 127 [Jun 20 13:23:55] DEBUG[21156][C-00000002] rtp_engine.c: Don't have a default tx payload type 127 format for m type on 0x7f2ed81803e0 [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 20 13:23:55] VERBOSE[21156][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 20 13:23:55] VERBOSE[21156][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 127 [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:127 telephone-event/8000... OK. [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=nortpproxy:yes... UNSUPPORTED OR FAILED. [Jun 20 13:23:55] VERBOSE[21156][C-00000002] chan_sip.c: Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jun 20 13:23:55] VERBOSE[21156][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 20 13:23:55] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1fbd428' [Jun 20 13:23:55] VERBOSE[21156][C-00000002] chan_sip.c: Peer audio RTP is at port 10.0.0.174:35116 [Jun 20 13:23:55] DEBUG[21156][C-00000002] rtp_engine.c: Copying payload 8 (0x7f2ec403c370) from 0x7f2ed81803e0 to 0x1fbd5f0 [Jun 20 13:23:55] DEBUG[21156][C-00000002] rtp_engine.c: Copying payload 127 (0x7f2ec403c470) from 0x7f2ed81803e0 to 0x1fbd5f0 [Jun 20 13:23:55] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1fbd428' [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: We're settling with these formats: (alaw) [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:23:55] DEBUG[21156][C-00000002] sip/route.c: sip_route_process_header: [Jun 20 13:23:55] DEBUG[21156][C-00000002] sip/route.c: sip_route_process_header: [Jun 20 13:23:55] DEBUG[21156][C-00000002] sip/route.c: sip_route_process_header: [Jun 20 13:23:55] VERBOSE[21156][C-00000002] sip/route.c: sip_route_dump: route/path hop: [Jun 20 13:23:55] VERBOSE[21156][C-00000002] sip/route.c: sip_route_dump: route/path hop: [Jun 20 13:23:55] VERBOSE[21156][C-00000002] sip/route.c: sip_route_dump: route/path hop: [Jun 20 13:23:55] VERBOSE[21156][C-00000002] 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=z9hG4bK1afbd9d2;rport Route: ,, Max-Forwards: 70 From: "37233" ;tag=as29fc0405 To: ;tag=C4769972-BA58EA4D Contact: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 102 ACK User-Agent: Asterisk PBX 13.9.1 Content-Length: 0 --- [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Trying to put 'ACK sip:u3-' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:23:55] VERBOSE[21342][C-00000002] app_dial.c: SIP/PROXY-out-00000005 answered SIP/ISP_SIPtrunk-00000004 [Jun 20 13:23:55] DEBUG[21342][C-00000002] chan_sip.c: SIP answering channel: SIP/ISP_SIPtrunk-00000004 [Jun 20 13:23:55] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:23:55] DEBUG[21342][C-00000002] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Jun 20 13:23:55] DEBUG[21342][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [Jun 20 13:23:55] VERBOSE[21342][C-00000002] chan_sip.c: Audio is at 49560 [Jun 20 13:23:55] VERBOSE[21342][C-00000002] chan_sip.c: Adding codec alaw to SDP [Jun 20 13:23:55] VERBOSE[21342][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 20 13:23:55] DEBUG[21342][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Jun 20 13:23:55] DEBUG[21342][C-00000002] chan_sip.c: Setting framing on incoming call: 20 [Jun 20 13:23:55] DEBUG[21342][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Jun 20 13:23:55] VERBOSE[21342][C-00000002] 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=z9hG4bK5f6d5be8;received=10.0.0.9;rport=5060 From: "37233" ;tag=as2a775170 To: ;tag=as27b2cbfb Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com CSeq: 102 INVITE Server: Asterisk PBX 13.9.1 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: 253 v=0 o=root 2090856424 2090856424 IN IP4 10.0.0.31 s=Asterisk PBX 13.9.1 c=IN IP4 10.0.0.31 t=0 0 m=audio 49560 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> [Jun 20 13:23:55] DEBUG[21342][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Jun 20 13:23:55] DEBUG[21342][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:23:55] DEBUG[21342][C-00000002] bridge.c: Chose bridge technology simple_bridge [Jun 20 13:23:55] DEBUG[21342][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: calling simple_bridge technology constructor [Jun 20 13:23:55] DEBUG[21342][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: calling simple_bridge technology start [Jun 20 13:23:55] DEBUG[21145] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:23:55] DEBUG[21145] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:23:55] DEBUG[21145] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:23:55] DEBUG[21145] devicestate.c: No provider found, checking channel drivers for SIP - ISP_SIPtrunk [Jun 20 13:23:55] DEBUG[21145] chan_sip.c: Checking device state for peer ISP_SIPtrunk [Jun 20 13:23:55] DEBUG[21145] devicestate.c: Changing state for SIP/ISP_SIPtrunk - state 1 (Not in use) [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Session timer started: 2 - 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com 900000ms [Jun 20 13:23:55] DEBUG[21345][C-00000002] bridge_channel.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: 0x1fa08d8(SIP/PROXY-out-00000005) is joining [Jun 20 13:23:55] DEBUG[21345][C-00000002] bridge_channel.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: pushing 0x1fa08d8(SIP/PROXY-out-00000005) [Jun 20 13:23:55] VERBOSE[21345][C-00000002] bridge_channel.c: Channel SIP/PROXY-out-00000005 joined 'simple_bridge' basic-bridge <38337361-db13-48c3-b493-4bb5c46d072a> [Jun 20 13:23:55] DEBUG[21345][C-00000002] bridge.c: Chose bridge technology simple_bridge [Jun 20 13:23:55] DEBUG[21345][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a is already using the new technology. [Jun 20 13:23:55] DEBUG[21345][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: 0x1fa08d8(SIP/PROXY-out-00000005) is joining simple_bridge technology [Jun 20 13:23:55] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jun 20 13:23:55] DEBUG[21342][C-00000002] bridge_channel.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: 0x2085cb8(SIP/ISP_SIPtrunk-00000004) is joining [Jun 20 13:23:55] DEBUG[21342][C-00000002] bridge_channel.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: pushing 0x2085cb8(SIP/ISP_SIPtrunk-00000004) [Jun 20 13:23:55] VERBOSE[21342][C-00000002] bridge_channel.c: Channel SIP/ISP_SIPtrunk-00000004 joined 'simple_bridge' basic-bridge <38337361-db13-48c3-b493-4bb5c46d072a> [Jun 20 13:23:55] DEBUG[21342][C-00000002] bridge.c: Chose bridge technology simple_bridge [Jun 20 13:23:55] DEBUG[21342][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a is already using the new technology. [Jun 20 13:23:55] DEBUG[21342][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: 0x2085cb8(SIP/ISP_SIPtrunk-00000004) is joining simple_bridge technology [Jun 20 13:23:55] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jun 20 13:23:55] DEBUG[21149] cdr.c: Finalized CDR for SIP/PROXY-out-00000005 - start 1466421833.883112 answer 1466421835.295830 end 1466421835.296786 dispo ANSWERED [Jun 20 13:23:55] VERBOSE[21156] 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=z9hG4bK41d353ce Max-Forwards: 70 From: "37233" ;tag=as2a775170 To: ;tag=as27b2cbfb Contact: Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com CSeq: 102 ACK User-Agent: Digium Gateway Content-Length: 0 <-------------> [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 0 [ 40]: ACK sip:23830@10.0.0.31:5100 SIP/2.0 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK41d353ce [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 3 [ 57]: From: "37233" ;tag=as2a775170 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 4 [ 52]: To: ;tag=as27b2cbfb [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 5 [ 38]: Contact: [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 6 [ 58]: Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 20 13:23:55] DEBUG[21156] chan_sip.c: Header 10 [ 0]: [Jun 20 13:23:55] VERBOSE[21156] chan_sip.c: --- (10 headers 0 lines) --- [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16 [Jun 20 13:23:55] DEBUG[21156][C-00000002] chan_sip.c: Stopping retransmission on '11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com' of Response 102: Match Found [Jun 20 13:23:55] DEBUG[21342][C-00000002] res_rtp_asterisk.c: 0x7f2ec4031390 -- Probation learning mode pass with source address 10.0.0.9:10142 [Jun 20 13:23:55] VERBOSE[21342][C-00000002] res_rtp_asterisk.c: 0x7f2ec4031390 -- Probation passed - setting RTP source address to 10.0.0.9:10142 [Jun 20 13:23:55] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Ooh, format changed from none to alaw [Jun 20 13:23:55] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x1fbd428' [Jun 20 13:23:55] DEBUG[21345][C-00000002] res_rtp_asterisk.c: 0x2090a40 -- Probation learning mode pass with source address 10.0.0.174:35116 [Jun 20 13:23:55] VERBOSE[21345][C-00000002] res_rtp_asterisk.c: 0x2090a40 -- Probation passed - setting RTP source address to 10.0.0.174:35116 [Jun 20 13:23:55] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Ooh, format changed from none to alaw [Jun 20 13:23:56] VERBOSE[21156] 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=z9hG4bKbf49.9c6ab7f.0 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKbf49.8c6ab7f.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKbf49.d6e2a3b1.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKe89e915094A8F53B From: "userB" ;tag=C4769972-BA58EA4D To: "37233" ;tag=as29fc0405 CSeq: 1 INVITE Call-ID: 7086c41e68689783760a32b471464e08@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=- 1466366352 1466366353 IN IP4 10.11.198.230 s=Polycom IP Phone c=IN IP4 10.0.0.174 t=0 0 a=sendonly m=audio 35116 RTP/AVP 8 101 a=sendonly a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=nortpproxy:yes <-------------> [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 0 [ 43]: INVITE sip:37233@10.0.0.31:5100 SIP/2.0 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 1 [ 67]: Record-Route: [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 2 [ 67]: Record-Route: [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 3 [ 68]: Record-Route: [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 4 [ 64]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKbf49.9c6ab7f.0 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 5 [ 64]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKbf49.8c6ab7f.0 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKbf49.d6e2a3b1.0 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 7 [ 72]: Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKe89e915094A8F53B [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 8 [ 86]: From: "userB" ;tag=C4769972-BA58EA4D [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 9 [ 62]: To: "37233" ;tag=as29fc0405 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 10 [ 14]: CSeq: 1 INVITE [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 11 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 12 [ 33]: Contact: [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 13 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 14 [ 34]: Allow-Events: talk,hold,conference [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 15 [ 16]: Max-Forwards: 11 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 17 [ 19]: Content-Length: 233 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 18 [ 16]: P-RTP-Proxy: Yes [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 19 [ 0]: [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 0 [ 3]: v=0 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 1 [ 46]: o=- 1466366352 1466366353 IN IP4 10.11.198.230 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 3 [ 23]: c=IN IP4 10.0.0.174 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 5 [ 10]: a=sendonly [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 6 [ 27]: m=audio 35116 RTP/AVP 8 101 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 7 [ 10]: a=sendonly [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Body 10 [ 16]: a=nortpproxy:yes [Jun 20 13:23:56] VERBOSE[21156] chan_sip.c: --- (19 headers 11 lines) --- [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 20 13:23:56] DEBUG[21156][C-00000002] netsock2.c: Splitting '10.0.0.37:5070' into... [Jun 20 13:23:56] DEBUG[21156][C-00000002] netsock2.c: ...host '10.0.0.37' and port '5070'. [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Sending to 10.0.0.37:5070 (NAT) [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP o=- 1466366352 1466366353 IN IP4 10.11.198.230... OK. [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Jun 20 13:23:56] DEBUG[21156][C-00000002] netsock2.c: Splitting '10.0.0.174' into... [Jun 20 13:23:56] DEBUG[21156][C-00000002] netsock2.c: ...host '10.0.0.174' and port ''. [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.174... OK. [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing session-level SDP a=sendonly... OK. [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Found RTP audio format 8 [Jun 20 13:23:56] DEBUG[21156][C-00000002] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f2ed8180fb0 [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Found RTP audio format 101 [Jun 20 13:23:56] DEBUG[21156][C-00000002] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f2ed8180fb0 [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=nortpproxy:yes... UNSUPPORTED OR FAILED. [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 20 13:23:56] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1fbd428' [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Peer audio RTP is at port 10.0.0.174:35116 [Jun 20 13:23:56] DEBUG[21156][C-00000002] rtp_engine.c: Copying payload 8 (0x7f2ec4006fc0) from 0x7f2ed8180fb0 to 0x1fbd5f0 [Jun 20 13:23:56] DEBUG[21156][C-00000002] rtp_engine.c: Copying payload 101 (0x7f2ec40070c0) from 0x7f2ed8180fb0 to 0x1fbd5f0 [Jun 20 13:23:56] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x1fbd428' [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: We're settling with these formats: (alaw) [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Jun 20 13:23:56] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1fbd428' [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Got a SIP re-invite for call 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: SIP/PROXY-out-00000005: This call is UP.... [Jun 20 13:23:56] VERBOSE[21156][C-00000002] 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=z9hG4bKbf49.9c6ab7f.0;received=10.0.0.37;rport=5070 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKbf49.8c6ab7f.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKbf49.d6e2a3b1.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKe89e915094A8F53B Record-Route: Record-Route: Record-Route: From: "userB" ;tag=C4769972-BA58EA4D To: "37233" ;tag=as29fc0405 Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 1 INVITE Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: ** Our prefcodec: (alaw) [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Audio is at 43472 [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Adding codec alaw to SDP [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Adding codec g729 to SDP [Jun 20 13:23:56] VERBOSE[21156][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Setting framing on incoming call: 20 [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Jun 20 13:23:56] VERBOSE[21156][C-00000002] 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=z9hG4bKbf49.9c6ab7f.0;received=10.0.0.37;rport=5070 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKbf49.8c6ab7f.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKbf49.d6e2a3b1.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bKe89e915094A8F53B Record-Route: Record-Route: Record-Route: From: "userB" ;tag=C4769972-BA58EA4D To: "37233" ;tag=as29fc0405 Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 1 INVITE Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 1200515678 1200515679 IN IP4 10.0.0.31 s=Asterisk PBX 13.9.1 c=IN IP4 10.0.0.31 t=0 0 m=audio 43472 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=maxptime:150 a=recvonly <------------> [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:23:56] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:23:56] VERBOSE[21342][C-00000002] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/ISP_SIPtrunk-00000004' [Jun 20 13:23:56] DEBUG[21342][C-00000002] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jun 20 13:23:56] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 20 13:23:56] DEBUG[21345][C-00000002] res_rtp_asterisk.c: 0x2090a40 -- Probation learning mode pass with source address 10.0.0.174:35116 [Jun 20 13:23:56] VERBOSE[21345][C-00000002] res_rtp_asterisk.c: 0x2090a40 -- Probation passed - setting RTP source address to 10.0.0.174:35116 [Jun 20 13:23:56] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1fbd428' [Jun 20 13:23:56] DEBUG[21342][C-00000002] channel.c: Channel SIP/ISP_SIPtrunk-00000004 setting write format path: slin -> alaw [Jun 20 13:23:56] DEBUG[21342][C-00000002] res_musiconhold.c: SIP/ISP_SIPtrunk-00000004 Opened file 0 '/opt/asterisk-ibercom/var/lib/asterisk/moh/reno_project-system' [Jun 20 13:23:56] VERBOSE[21156] 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=z9hG4bKbf49.8c6ab7f.2 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKbf49.8c6ab7f.2 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKbf49.d6e2a3b1.2 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK809586ce1E370C9 From: "userB" ;tag=C4769972-BA58EA4D To: "37233" ;tag=as29fc0405 CSeq: 1 ACK Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com Contact: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 Max-Forwards: 11 Content-Length: 0 <-------------> [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 0 [ 40]: ACK sip:37233@10.0.0.31:5100 SIP/2.0 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKbf49.8c6ab7f.2 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 2 [ 64]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bKbf49.8c6ab7f.2 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 3 [ 60]: Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bKbf49.d6e2a3b1.2 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 4 [ 71]: Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK809586ce1E370C9 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 5 [ 86]: From: "userB" ;tag=C4769972-BA58EA4D [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 6 [ 62]: To: "37233" ;tag=as29fc0405 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 7 [ 11]: CSeq: 1 ACK [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 8 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 9 [ 33]: Contact: [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 10 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 11 [ 16]: Max-Forwards: 11 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:23:56] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:23:56] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Jun 20 13:23:56] DEBUG[21156][C-00000002] chan_sip.c: Stopping retransmission on '7086c41e68689783760a32b471464e08@mydomain.com' of Response 1: Match Found [Jun 20 13:23:57] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:23:57] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:23:57] DEBUG[21342][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:23:57] DEBUG[21342][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:23:57] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:23:57] DEBUG[21342][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:00] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Got RTCP report of 80 bytes [Jun 20 13:24:00] DEBUG[21345][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:00] DEBUG[21345][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:00] DEBUG[21345][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:00] DEBUG[21345][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:00] DEBUG[21345][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:00] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:00] DEBUG[21156] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:00] DEBUG[21156] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:00] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:00] DEBUG[21156] acl.c: Attached to given IP address [Jun 20 13:24:01] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:01] DEBUG[21156] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:01] DEBUG[21156] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:01] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:01] DEBUG[21156] acl.c: Attached to given IP address [Jun 20 13:24:02] VERBOSE[21156] 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=z9hG4bK8f49.4798bf56.0 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK8f49.3798bf56.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK8f49.082578d7.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK7d5cad3e85B473B9 From: "userB" ;tag=C4769972-BA58EA4D To: "37233" ;tag=as29fc0405 CSeq: 2 REFER Call-ID: 7086c41e68689783760a32b471464e08@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 13:24:02] DEBUG[21156] chan_sip.c: Header 0 [ 42]: REFER sip:37233@10.0.0.31:5100 SIP/2.0 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 1 [ 68]: Record-Route: [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 2 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK8f49.4798bf56.0 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 3 [ 65]: Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK8f49.3798bf56.0 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK8f49.082578d7.0 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 5 [ 72]: Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK7d5cad3e85B473B9 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 6 [ 86]: From: "userB" ;tag=C4769972-BA58EA4D [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 7 [ 62]: To: "37233" ;tag=as29fc0405 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 8 [ 13]: CSeq: 2 REFER [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 9 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 10 [ 33]: Contact: [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 11 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 12 [138]: Refer-To: [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 13 [ 58]: Referred-By: [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 14 [ 16]: Max-Forwards: 11 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 16 [ 0]: [Jun 20 13:24:02] VERBOSE[21156] chan_sip.c: --- (16 headers 0 lines) --- [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Jun 20 13:24:02] VERBOSE[21156][C-00000002] chan_sip.c: Call 7086c41e68689783760a32b471464e08@mydomain.com got a SIP call transfer from caller: (REFER)! [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 78525333-d698bd06-a0c37d41@10.11.198.230 (No check of from/to tags) [Jun 20 13:24:02] VERBOSE[21156][C-00000002] chan_sip.c: SIP transfer to extension u6-2@station_transferUC by userB@mydomain.com [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: This SIP transfer is to a remote SIP extension (remote domain 10.0.0.37?Replaces=78525333-d698bd06-a0c37d41%4010.11.198.230%3Bto-tag%3D883327568%3Bfrom-tag%3D34483AF7-212442AA) [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: SIP attended transfer: Transferer channel SIP/PROXY-out-00000005 [Jun 20 13:24:02] VERBOSE[21156][C-00000002] 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=z9hG4bK8f49.4798bf56.0;received=10.0.0.37;rport=5070 Via: SIP/2.0/UDP 10.0.0.37:5070;branch=z9hG4bK8f49.3798bf56.0 Via: SIP/2.0/UDP 10.0.0.37;branch=z9hG4bK8f49.082578d7.0 Via: SIP/2.0/UDP 10.11.198.230;rport=5060;branch=z9hG4bK7d5cad3e85B473B9 Record-Route: From: "userB" ;tag=C4769972-BA58EA4D To: "37233" ;tag=as29fc0405 Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 2 REFER Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: Looking for callid 78525333-d698bd06-a0c37d41@10.11.198.230 (fromtag 34483AF7-212442AA totag 883327568) [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: SIP attended transfer: Error: No owner of target call [Jun 20 13:24:02] VERBOSE[21156][C-00000002] 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=z9hG4bK55444c3d;rport Route: ,, Max-Forwards: 70 From: "37233" ;tag=as29fc0405 To: "userB" ;tag=C4769972-BA58EA4D Contact: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 103 NOTIFY User-Agent: Asterisk PBX 13.9.1 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: 33 SIP/2.0 503 Service Unavailable --- [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: SIP message could not be handled, bad request: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:24:02] VERBOSE[21156] 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=z9hG4bK55444c3d;rport=5100 From: "37233" ;tag=as29fc0405 To: "userB" ;tag=C4769972-BA58EA4D CSeq: 103 NOTIFY Call-ID: 7086c41e68689783760a32b471464e08@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 13:24:02] DEBUG[21156] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK55444c3d;rport=5100 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as29fc0405 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 3 [ 84]: To: "userB" ;tag=C4769972-BA58EA4D [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 4 [ 16]: CSeq: 103 NOTIFY [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 5 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 6 [ 33]: Contact: [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 7 [ 61]: Record-Route: [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 8 [ 17]: Event: refer;id=2 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 10 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jun 20 13:24:02] DEBUG[21156] chan_sip.c: Header 12 [ 0]: [Jun 20 13:24:02] VERBOSE[21156] chan_sip.c: --- (12 headers 0 lines) --- [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8 [Jun 20 13:24:02] DEBUG[21156][C-00000002] chan_sip.c: Stopping retransmission on '7086c41e68689783760a32b471464e08@mydomain.com' of Request 103: Match Found [Jun 20 13:24:02] NOTICE[21156][C-00000002] chan_sip.c: Got OK on REFER Notify message [Jun 20 13:24:02] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:24:02] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:02] DEBUG[21342][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:02] DEBUG[21342][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:02] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:02] DEBUG[21342][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:05] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Got RTCP report of 60 bytes [Jun 20 13:24:05] DEBUG[21345][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:05] DEBUG[21345][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:05] DEBUG[21345][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:05] DEBUG[21345][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:05] DEBUG[21345][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:05] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:05] DEBUG[21156] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:05] DEBUG[21156] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:05] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:05] DEBUG[21156] acl.c: Attached to given IP address [Jun 20 13:24:06] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:06] DEBUG[21156] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:06] DEBUG[21156] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:06] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:06] DEBUG[21156] acl.c: Attached to given IP address [Jun 20 13:24:07] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:24:07] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:07] DEBUG[21342][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:07] DEBUG[21342][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:07] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:07] DEBUG[21342][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:10] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Got RTCP report of 60 bytes [Jun 20 13:24:10] DEBUG[21345][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:10] DEBUG[21345][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:10] DEBUG[21345][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:10] DEBUG[21345][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:10] DEBUG[21345][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:10] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:10] DEBUG[21156] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:10] DEBUG[21156] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:10] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:10] DEBUG[21156] acl.c: Attached to given IP address [Jun 20 13:24:11] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:11] DEBUG[21156] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:11] DEBUG[21156] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:11] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:11] DEBUG[21156] acl.c: Attached to given IP address [Jun 20 13:24:12] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:24:12] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:12] DEBUG[21342][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:12] DEBUG[21342][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:12] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:12] DEBUG[21342][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:15] VERBOSE[21341] asterisk.c: Remote UNIX connection disconnected [Jun 20 13:24:15] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Got RTCP report of 60 bytes [Jun 20 13:24:15] DEBUG[21345][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:15] DEBUG[21345][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:15] DEBUG[21345][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:15] DEBUG[21345][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:15] DEBUG[21345][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:15] DEBUG[21344] threadpool.c: Worker thread idle timeout reached. Dying. [Jun 20 13:24:15] DEBUG[21133] threadpool.c: Destroying worker thread 8 [Jun 20 13:24:15] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:15] DEBUG[21156] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:15] DEBUG[21156] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:15] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:15] DEBUG[21156] acl.c: Attached to given IP address [Jun 20 13:24:16] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:16] DEBUG[21156] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:16] DEBUG[21156] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:16] DEBUG[21156] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:16] DEBUG[21156] acl.c: Attached to given IP address [Jun 20 13:24:17] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Jun 20 13:24:17] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:17] DEBUG[21342][C-00000002] netsock2.c: Splitting 'asterisk' into... [Jun 20 13:24:17] DEBUG[21342][C-00000002] netsock2.c: ...host 'asterisk' and port ''. [Jun 20 13:24:17] DEBUG[21342][C-00000002] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 20 13:24:17] DEBUG[21342][C-00000002] acl.c: Attached to given IP address [Jun 20 13:24:19] VERBOSE[21156] 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=z9hG4bK66161e3d Max-Forwards: 70 From: "37233" ;tag=as2a775170 To: ;tag=as27b2cbfb Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com CSeq: 103 BYE User-Agent: Digium Gateway X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 0 [ 40]: BYE sip:23830@10.0.0.31:5100 SIP/2.0 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK66161e3d [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 3 [ 57]: From: "37233" ;tag=as2a775170 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 4 [ 52]: To: ;tag=as27b2cbfb [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 5 [ 58]: Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 6 [ 13]: CSeq: 103 BYE [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 7 [ 26]: User-Agent: Digium Gateway [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:24:19] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:24:19] DEBUG[21156][C-00000002] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jun 20 13:24:19] DEBUG[21156][C-00000002] chan_sip.c: Initializing initreq for method BYE - callid 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:24:19] DEBUG[21156][C-00000002] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:24:19] DEBUG[21156][C-00000002] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:24:19] VERBOSE[21156][C-00000002] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:24:19] DEBUG[21156][C-00000002] chan_sip.c: Setting SIP_ALREADYGONE on dialog 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:24:19] DEBUG[21156][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2ec401c358' [Jun 20 13:24:19] VERBOSE[21156][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com' in 6400 ms (Method: BYE) [Jun 20 13:24:19] DEBUG[21156][C-00000002] chan_sip.c: Received bye, issuing owner hangup [Jun 20 13:24:19] VERBOSE[21156][C-00000002] 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=z9hG4bK66161e3d;received=10.0.0.9;rport=5060 From: "37233" ;tag=as2a775170 To: ;tag=as27b2cbfb Call-ID: 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com CSeq: 103 BYE Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jun 20 13:24:19] DEBUG[21156][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Session timer stopped: 2 - 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:24:19] DEBUG[21133] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jun 20 13:24:19] VERBOSE[21342][C-00000002] res_musiconhold.c: Stopped music on hold on SIP/ISP_SIPtrunk-00000004 [Jun 20 13:24:19] DEBUG[21342][C-00000002] channel.c: Channel SIP/ISP_SIPtrunk-00000004 setting write format path: alaw -> alaw [Jun 20 13:24:19] DEBUG[21342][C-00000002] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 20 13:24:19] DEBUG[21342][C-00000002] bridge_channel.c: Setting 0x2085cb8(SIP/ISP_SIPtrunk-00000004) state from:0 to:1 [Jun 20 13:24:19] DEBUG[21342][C-00000002] bridge_channel.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: pulling 0x2085cb8(SIP/ISP_SIPtrunk-00000004) [Jun 20 13:24:19] VERBOSE[21342][C-00000002] bridge_channel.c: Channel SIP/ISP_SIPtrunk-00000004 left 'simple_bridge' basic-bridge <38337361-db13-48c3-b493-4bb5c46d072a> [Jun 20 13:24:19] DEBUG[21342][C-00000002] bridge_channel.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: 0x2085cb8(SIP/ISP_SIPtrunk-00000004) is leaving simple_bridge technology [Jun 20 13:24:19] DEBUG[21342][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: dissolving bridge with cause 16(Normal Clearing) [Jun 20 13:24:19] DEBUG[21342][C-00000002] bridge_channel.c: Setting 0x1fa08d8(SIP/PROXY-out-00000005) state from:0 to:2 [Jun 20 13:24:19] DEBUG[21342][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: queueing action type:13 sub:1001 [Jun 20 13:24:19] DEBUG[21342][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a is dissolved, not performing smart bridge operation. [Jun 20 13:24:19] DEBUG[21342][C-00000002] bridge_channel.c: Channel SIP/ISP_SIPtrunk-00000004 simulating UNHOLD for bridge end. [Jun 20 13:24:19] DEBUG[21342][C-00000002] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jun 20 13:24:19] DEBUG[21342][C-00000002] app_macro.c: Spawn extension (macro-llamadaEntranteUC,s,23) exited non-zero on 'SIP/ISP_SIPtrunk-00000004' in macro 'llamadaEntranteUC' [Jun 20 13:24:19] VERBOSE[21342][C-00000002] app_macro.c: Spawn extension (macro-llamadaEntranteUC, s, 23) exited non-zero on 'SIP/ISP_SIPtrunk-00000004' in macro 'llamadaEntranteUC' [Jun 20 13:24:19] DEBUG[21342][C-00000002] pbx.c: Spawn extension (incoming,23830,1) exited non-zero on 'SIP/ISP_SIPtrunk-00000004' [Jun 20 13:24:19] VERBOSE[21342][C-00000002] pbx.c: Spawn extension (incoming, 23830, 1) exited non-zero on 'SIP/ISP_SIPtrunk-00000004' [Jun 20 13:24:19] DEBUG[21342][C-00000002] channel.c: Soft-Hanging (0x10) up channel 'SIP/ISP_SIPtrunk-00000004' [Jun 20 13:24:19] DEBUG[21342][C-00000002] channel.c: Hanging up channel 'SIP/ISP_SIPtrunk-00000004' [Jun 20 13:24:19] DEBUG[21342][C-00000002] chan_sip.c: Hangup call SIP/ISP_SIPtrunk-00000004, SIP callid 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:24:19] DEBUG[21342][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2ec401c358' [Jun 20 13:24:19] DEBUG[21149] cdr.c: Finalized CDR for SIP/ISP_SIPtrunk-00000004 - start 1466421833.879112 answer 1466421835.295858 end 1466421859.525032 dispo ANSWERED [Jun 20 13:24:19] DEBUG[21345][C-00000002] bridge_channel.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: pulling 0x1fa08d8(SIP/PROXY-out-00000005) [Jun 20 13:24:19] VERBOSE[21345][C-00000002] bridge_channel.c: Channel SIP/PROXY-out-00000005 left 'simple_bridge' basic-bridge <38337361-db13-48c3-b493-4bb5c46d072a> [Jun 20 13:24:19] DEBUG[21345][C-00000002] bridge_channel.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: 0x1fa08d8(SIP/PROXY-out-00000005) is leaving simple_bridge technology [Jun 20 13:24:19] DEBUG[21345][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a is dissolved, not performing smart bridge operation. [Jun 20 13:24:19] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1018141467 to 805563919 due to a source change [Jun 20 13:24:19] DEBUG[21345][C-00000002] channel.c: Hanging up channel 'SIP/PROXY-out-00000005' [Jun 20 13:24:19] DEBUG[21345][C-00000002] chan_sip.c: Hangup call SIP/PROXY-out-00000005, SIP callid 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:24:19] DEBUG[21345][C-00000002] chan_sip.c: update_call_counter(23830) - decrement call limit counter on hangup [Jun 20 13:24:19] DEBUG[21345][C-00000002] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:24:19] DEBUG[21345][C-00000002] chan_sip.c: Call to peer 'PROXY-out' removed from call limit 0 [Jun 20 13:24:19] DEBUG[21345][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x1fbd428' [Jun 20 13:24:19] VERBOSE[21345][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '7086c41e68689783760a32b471464e08@mydomain.com' in 32000 ms (Method: REFER) [Jun 20 13:24:19] VERBOSE[21345][C-00000002] chan_sip.c: Reliably Transmitting (NAT) to 10.0.0.37:5070: BYE sip:u3-0@10.0.0.37 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK67036b91;rport Route: ,, Max-Forwards: 70 From: "37233" ;tag=as29fc0405 To: "userB" ;tag=C4769972-BA58EA4D Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com CSeq: 104 BYE User-Agent: Asterisk PBX 13.9.1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jun 20 13:24:19] DEBUG[21345][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Jun 20 13:24:19] DEBUG[21345][C-00000002] chan_sip.c: Trying to put 'BYE sip:u3-' onto UDP socket destined for 10.0.0.37:5070 [Jun 20 13:24:19] DEBUG[21144][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: actually destroying basic bridge, nobody wants it anymore [Jun 20 13:24:19] DEBUG[21144][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: calling basic bridge destructor [Jun 20 13:24:19] DEBUG[21144][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: calling simple_bridge technology stop [Jun 20 13:24:19] DEBUG[21144][C-00000002] bridge.c: Bridge 38337361-db13-48c3-b493-4bb5c46d072a: calling simple_bridge technology destructor [Jun 20 13:24:19] DEBUG[21145] devicestate.c: No provider found, checking channel drivers for SIP - ISP_SIPtrunk [Jun 20 13:24:19] DEBUG[21145] chan_sip.c: Checking device state for peer ISP_SIPtrunk [Jun 20 13:24:19] DEBUG[21145] devicestate.c: Changing state for SIP/ISP_SIPtrunk - state 1 (Not in use) [Jun 20 13:24:19] DEBUG[21145] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:24:19] DEBUG[21145] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:24:19] DEBUG[21145] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:24:19] DEBUG[21145] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:24:19] DEBUG[21145] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:24:19] DEBUG[21145] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:24:19] DEBUG[21149] cdr.c: CDR for SIP/PROXY-out-00000005 is dialed and has no Party B; discarding [Jun 20 13:24:19] VERBOSE[21156] 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=z9hG4bK67036b91;rport=5100 From: "37233" ;tag=as29fc0405 To: "userB" ;tag=C4769972-BA58EA4D CSeq: 104 BYE Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com Contact: 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 13:24:19] DEBUG[21156] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;received=10.0.0.31;branch=z9hG4bK67036b91;rport=5100 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 2 [ 64]: From: "37233" ;tag=as29fc0405 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 3 [ 84]: To: "userB" ;tag=C4769972-BA58EA4D [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 4 [ 13]: CSeq: 104 BYE [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 5 [ 60]: Call-ID: 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 6 [ 33]: Contact: [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.2.0413 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 8 [ 40]: Accept-Language: pl-pl,pl;q=0.9,en;q=0.8 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Header 10 [ 0]: [Jun 20 13:24:19] VERBOSE[21156] chan_sip.c: --- (10 headers 0 lines) --- [Jun 20 13:24:19] DEBUG[21156][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Jun 20 13:24:19] DEBUG[21156][C-00000002] chan_sip.c: Stopping retransmission on '7086c41e68689783760a32b471464e08@mydomain.com' of Request 104: Match Found [Jun 20 13:24:19] VERBOSE[21156][C-00000002] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Destroying SIP dialog 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:24:19] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '7086c41e68689783760a32b471464e08@mydomain.com' Method: REFER [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Updating call counter for outgoing call [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: Call to peer 'PROXY-out' removed from call limit 0 [Jun 20 13:24:19] DEBUG[21156] chan_sip.c: This call did not properly clean up call limits. Call ID 7086c41e68689783760a32b471464e08@mydomain.com [Jun 20 13:24:19] DEBUG[21156] rtp_engine.c: Destroyed RTP instance '0x1fbd428' [Jun 20 13:24:19] DEBUG[21145] devicestate.c: No provider found, checking channel drivers for SIP - PROXY-out [Jun 20 13:24:19] DEBUG[21145] chan_sip.c: Checking device state for peer PROXY-out [Jun 20 13:24:19] DEBUG[21145] devicestate.c: Changing state for SIP/PROXY-out - state 1 (Not in use) [Jun 20 13:24:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '3f97ac085f1663545e55d67c658e3c19@10.0.0.9:5060' [Jun 20 13:24:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 3f97ac085f1663545e55d67c658e3c19@10.0.0.9:5060 [Jun 20 13:24:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '3f97ac085f1663545e55d67c658e3c19@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:24:25] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com' [Jun 20 13:24:25] DEBUG[21156] chan_sip.c: Destroying SIP dialog 11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com [Jun 20 13:24:25] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '11b9a0af470f95f66325b63e6aea8676@asterisk.mydomain.com' Method: BYE [Jun 20 13:24:25] DEBUG[21156] rtp_engine.c: Destroyed RTP instance '0x7f2ec401c358' [Jun 20 13:24:39] DEBUG[21370] threadpool.c: Worker thread idle timeout reached. Dying. [Jun 20 13:24:39] DEBUG[21133] threadpool.c: Destroying worker thread 9 [Jun 20 13:24:39] DEBUG[21343] threadpool.c: Worker thread idle timeout reached. Dying. [Jun 20 13:24:39] DEBUG[21133] threadpool.c: Destroying worker thread 7 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 2cd82e61361e369267774e0a58a431d9@mydomain.com - OPTIONS (No RTP) [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:24:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '2cd82e61361e369267774e0a58a431d9@mydomain.com' to '1e59981c5bafedfb514ac501545c1d6a@mydomain.com' [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 1e59981c5bafedfb514ac501545c1d6a@mydomain.com [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK0a8daa87;rport [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as7c7d0df9 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 1e59981c5bafedfb514ac501545c1d6a@mydomain.com [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:24:42 GMT [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:24:42] VERBOSE[21156] 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=z9hG4bK0a8daa87;rport Max-Forwards: 70 From: "asterisk" ;tag=as7c7d0df9 To: Contact: Call-ID: 1e59981c5bafedfb514ac501545c1d6a@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:24:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:24:42] VERBOSE[21156] 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=z9hG4bK0a8daa87;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as7c7d0df9 To: ;tag=as6c5c381f Call-ID: 1e59981c5bafedfb514ac501545c1d6a@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 13:24:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK0a8daa87;received=10.0.0.31;rport=5100 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as7c7d0df9 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as6c5c381f [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 1e59981c5bafedfb514ac501545c1d6a@mydomain.com [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:24:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16 [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '1e59981c5bafedfb514ac501545c1d6a@mydomain.com' of Request 102: Match Found [Jun 20 13:24:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 1e59981c5bafedfb514ac501545c1d6a@mydomain.com [Jun 20 13:24:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '1e59981c5bafedfb514ac501545c1d6a@mydomain.com' Method: OPTIONS [Jun 20 13:24:49] VERBOSE[21156] 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=z9hG4bK17021fb6 Max-Forwards: 70 From: "asterisk" ;tag=as0b28b20a To: Contact: Call-ID: 29f875ed6377648f6405e92747ba23fe@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:24:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK17021fb6 [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as0b28b20a [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 29f875ed6377648f6405e92747ba23fe@10.0.0.9:5060 [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:24:49 GMT [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:24:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:24:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:24:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:24:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:24:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 29f875ed6377648f6405e92747ba23fe@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:24:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:24:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:24:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:24:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:24:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:24:49] VERBOSE[21156] 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=z9hG4bK17021fb6;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as0b28b20a To: ;tag=as155ed703 Call-ID: 29f875ed6377648f6405e92747ba23fe@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:24:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:24:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '29f875ed6377648f6405e92747ba23fe@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:25:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '29f875ed6377648f6405e92747ba23fe@10.0.0.9:5060' [Jun 20 13:25:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 29f875ed6377648f6405e92747ba23fe@10.0.0.9:5060 [Jun 20 13:25:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '29f875ed6377648f6405e92747ba23fe@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 4ec26d006b064c5806511ff320785729@mydomain.com - OPTIONS (No RTP) [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:25:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '4ec26d006b064c5806511ff320785729@mydomain.com' to '2e1abab01c4876d94a3cb9d211c5dfde@mydomain.com' [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 2e1abab01c4876d94a3cb9d211c5dfde@mydomain.com [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK02e79296;rport [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as03abb0d3 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 2e1abab01c4876d94a3cb9d211c5dfde@mydomain.com [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:25:42 GMT [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:25:42] VERBOSE[21156] 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=z9hG4bK02e79296;rport Max-Forwards: 70 From: "asterisk" ;tag=as03abb0d3 To: Contact: Call-ID: 2e1abab01c4876d94a3cb9d211c5dfde@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:25:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:25:42] VERBOSE[21156] 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=z9hG4bK02e79296;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as03abb0d3 To: ;tag=as075508d4 Call-ID: 2e1abab01c4876d94a3cb9d211c5dfde@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 13:25:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK02e79296;received=10.0.0.31;rport=5100 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as03abb0d3 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as075508d4 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 2e1abab01c4876d94a3cb9d211c5dfde@mydomain.com [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:25:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11 [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '2e1abab01c4876d94a3cb9d211c5dfde@mydomain.com' of Request 102: Match Found [Jun 20 13:25:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 2e1abab01c4876d94a3cb9d211c5dfde@mydomain.com [Jun 20 13:25:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '2e1abab01c4876d94a3cb9d211c5dfde@mydomain.com' Method: OPTIONS [Jun 20 13:25:49] VERBOSE[21156] 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=z9hG4bK53a97b82 Max-Forwards: 70 From: "asterisk" ;tag=as10cd296d To: Contact: Call-ID: 1c0845de159592cc4afb42824534edf0@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:25:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK53a97b82 [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as10cd296d [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 1c0845de159592cc4afb42824534edf0@10.0.0.9:5060 [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:25:49 GMT [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:25:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:25:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:25:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:25:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:25:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 1c0845de159592cc4afb42824534edf0@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:25:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:25:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:25:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:25:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:25:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:25:49] VERBOSE[21156] 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=z9hG4bK53a97b82;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as10cd296d To: ;tag=as1d8bac8b Call-ID: 1c0845de159592cc4afb42824534edf0@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:25:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:25:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '1c0845de159592cc4afb42824534edf0@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:26:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '1c0845de159592cc4afb42824534edf0@10.0.0.9:5060' [Jun 20 13:26:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 1c0845de159592cc4afb42824534edf0@10.0.0.9:5060 [Jun 20 13:26:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '1c0845de159592cc4afb42824534edf0@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 124507a72b67bdbb1cb5752339cc81e8@mydomain.com - OPTIONS (No RTP) [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:26:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '124507a72b67bdbb1cb5752339cc81e8@mydomain.com' to '76aa73135ef2dac53b9a8682081a5c17@mydomain.com' [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 76aa73135ef2dac53b9a8682081a5c17@mydomain.com [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK4b4d45c4;rport [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as3f0e96d6 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 76aa73135ef2dac53b9a8682081a5c17@mydomain.com [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:26:42 GMT [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:26:42] VERBOSE[21156] 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=z9hG4bK4b4d45c4;rport Max-Forwards: 70 From: "asterisk" ;tag=as3f0e96d6 To: Contact: Call-ID: 76aa73135ef2dac53b9a8682081a5c17@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:26:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:26:42] VERBOSE[21156] 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=z9hG4bK4b4d45c4;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as3f0e96d6 To: ;tag=as2351d276 Call-ID: 76aa73135ef2dac53b9a8682081a5c17@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 13:26:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK4b4d45c4;received=10.0.0.31;rport=5100 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as3f0e96d6 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as2351d276 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 76aa73135ef2dac53b9a8682081a5c17@mydomain.com [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:26:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '76aa73135ef2dac53b9a8682081a5c17@mydomain.com' of Request 102: Match Found [Jun 20 13:26:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 76aa73135ef2dac53b9a8682081a5c17@mydomain.com [Jun 20 13:26:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '76aa73135ef2dac53b9a8682081a5c17@mydomain.com' Method: OPTIONS [Jun 20 13:26:49] VERBOSE[21156] 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=z9hG4bK4e451b47 Max-Forwards: 70 From: "asterisk" ;tag=as1a36e3f7 To: Contact: Call-ID: 37ef6a220a852d975a28b30d55432a69@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:26:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK4e451b47 [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as1a36e3f7 [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 37ef6a220a852d975a28b30d55432a69@10.0.0.9:5060 [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:26:49 GMT [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:26:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:26:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:26:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:26:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:26:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 37ef6a220a852d975a28b30d55432a69@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:26:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:26:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:26:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:26:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:26:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:26:49] VERBOSE[21156] 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=z9hG4bK4e451b47;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as1a36e3f7 To: ;tag=as628bf38f Call-ID: 37ef6a220a852d975a28b30d55432a69@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:26:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:26:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '37ef6a220a852d975a28b30d55432a69@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:27:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '37ef6a220a852d975a28b30d55432a69@10.0.0.9:5060' [Jun 20 13:27:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 37ef6a220a852d975a28b30d55432a69@10.0.0.9:5060 [Jun 20 13:27:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '37ef6a220a852d975a28b30d55432a69@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 644bb70b434787314349aecc330cfec1@mydomain.com - OPTIONS (No RTP) [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:27:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '644bb70b434787314349aecc330cfec1@mydomain.com' to '157734a71f03f85e6359257b7253bafb@mydomain.com' [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 157734a71f03f85e6359257b7253bafb@mydomain.com [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK2221e6d3;rport [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as64884142 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 157734a71f03f85e6359257b7253bafb@mydomain.com [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:27:42 GMT [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:27:42] VERBOSE[21156] 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=z9hG4bK2221e6d3;rport Max-Forwards: 70 From: "asterisk" ;tag=as64884142 To: Contact: Call-ID: 157734a71f03f85e6359257b7253bafb@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:27:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #10 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:27:42] VERBOSE[21156] 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=z9hG4bK2221e6d3;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as64884142 To: ;tag=as28563d51 Call-ID: 157734a71f03f85e6359257b7253bafb@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 13:27:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK2221e6d3;received=10.0.0.31;rport=5100 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as64884142 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as28563d51 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 157734a71f03f85e6359257b7253bafb@mydomain.com [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:27:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10 [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '157734a71f03f85e6359257b7253bafb@mydomain.com' of Request 102: Match Found [Jun 20 13:27:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 157734a71f03f85e6359257b7253bafb@mydomain.com [Jun 20 13:27:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '157734a71f03f85e6359257b7253bafb@mydomain.com' Method: OPTIONS [Jun 20 13:27:49] VERBOSE[21156] 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=z9hG4bK2b98ac2e Max-Forwards: 70 From: "asterisk" ;tag=as57ac2fad To: Contact: Call-ID: 6e608cd7159231f863bc0fe670ef8935@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:27:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK2b98ac2e [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as57ac2fad [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 6e608cd7159231f863bc0fe670ef8935@10.0.0.9:5060 [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:27:49 GMT [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:27:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:27:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:27:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:27:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:27:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 6e608cd7159231f863bc0fe670ef8935@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:27:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:27:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:27:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:27:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:27:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:27:49] VERBOSE[21156] 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=z9hG4bK2b98ac2e;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as57ac2fad To: ;tag=as4064da36 Call-ID: 6e608cd7159231f863bc0fe670ef8935@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:27:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:27:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '6e608cd7159231f863bc0fe670ef8935@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:28:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '6e608cd7159231f863bc0fe670ef8935@10.0.0.9:5060' [Jun 20 13:28:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 6e608cd7159231f863bc0fe670ef8935@10.0.0.9:5060 [Jun 20 13:28:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '6e608cd7159231f863bc0fe670ef8935@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 745e2127702a78c1274889b142d80c17@mydomain.com - OPTIONS (No RTP) [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:28:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '745e2127702a78c1274889b142d80c17@mydomain.com' to '4767bca658ae6a160eff6c2429c69074@mydomain.com' [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 4767bca658ae6a160eff6c2429c69074@mydomain.com [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1ad5bde1;rport [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as5b9b2cee [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 4767bca658ae6a160eff6c2429c69074@mydomain.com [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:28:42 GMT [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:28:42] VERBOSE[21156] 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=z9hG4bK1ad5bde1;rport Max-Forwards: 70 From: "asterisk" ;tag=as5b9b2cee To: Contact: Call-ID: 4767bca658ae6a160eff6c2429c69074@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:28:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:28:42] VERBOSE[21156] 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=z9hG4bK1ad5bde1;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as5b9b2cee To: ;tag=as72ec3152 Call-ID: 4767bca658ae6a160eff6c2429c69074@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 13:28:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1ad5bde1;received=10.0.0.31;rport=5100 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as5b9b2cee [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as72ec3152 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 4767bca658ae6a160eff6c2429c69074@mydomain.com [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:28:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '4767bca658ae6a160eff6c2429c69074@mydomain.com' of Request 102: Match Found [Jun 20 13:28:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 4767bca658ae6a160eff6c2429c69074@mydomain.com [Jun 20 13:28:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '4767bca658ae6a160eff6c2429c69074@mydomain.com' Method: OPTIONS [Jun 20 13:28:49] VERBOSE[21156] 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=z9hG4bK530a6674 Max-Forwards: 70 From: "asterisk" ;tag=as7dd31776 To: Contact: Call-ID: 230196c8050e83454a20fb8011095ab0@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:28:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK530a6674 [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as7dd31776 [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 230196c8050e83454a20fb8011095ab0@10.0.0.9:5060 [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:28:49 GMT [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:28:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:28:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:28:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:28:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:28:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 230196c8050e83454a20fb8011095ab0@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:28:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:28:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:28:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:28:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:28:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:28:49] VERBOSE[21156] 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=z9hG4bK530a6674;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as7dd31776 To: ;tag=as6863ff17 Call-ID: 230196c8050e83454a20fb8011095ab0@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:28:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:28:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '230196c8050e83454a20fb8011095ab0@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:29:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '230196c8050e83454a20fb8011095ab0@10.0.0.9:5060' [Jun 20 13:29:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 230196c8050e83454a20fb8011095ab0@10.0.0.9:5060 [Jun 20 13:29:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '230196c8050e83454a20fb8011095ab0@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 216d44e573f6957d76a025055670262c@mydomain.com - OPTIONS (No RTP) [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:29:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '216d44e573f6957d76a025055670262c@mydomain.com' to '7b420c16459d69ed1a1223a21da7a0e2@mydomain.com' [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 7b420c16459d69ed1a1223a21da7a0e2@mydomain.com [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK56a52be2;rport [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as2b0ad81c [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 7b420c16459d69ed1a1223a21da7a0e2@mydomain.com [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:29:42 GMT [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:29:42] VERBOSE[21156] 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=z9hG4bK56a52be2;rport Max-Forwards: 70 From: "asterisk" ;tag=as2b0ad81c To: Contact: Call-ID: 7b420c16459d69ed1a1223a21da7a0e2@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:29:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:29:42] VERBOSE[21156] 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=z9hG4bK56a52be2;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as2b0ad81c To: ;tag=as4985c899 Call-ID: 7b420c16459d69ed1a1223a21da7a0e2@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 13:29:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK56a52be2;received=10.0.0.31;rport=5100 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as2b0ad81c [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as4985c899 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 7b420c16459d69ed1a1223a21da7a0e2@mydomain.com [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:29:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '7b420c16459d69ed1a1223a21da7a0e2@mydomain.com' of Request 102: Match Found [Jun 20 13:29:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 7b420c16459d69ed1a1223a21da7a0e2@mydomain.com [Jun 20 13:29:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '7b420c16459d69ed1a1223a21da7a0e2@mydomain.com' Method: OPTIONS [Jun 20 13:29:49] VERBOSE[21156] 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=z9hG4bK011a4ff3 Max-Forwards: 70 From: "asterisk" ;tag=as24e48987 To: Contact: Call-ID: 4476b931475a379d11ef80a744d388b2@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:29:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK011a4ff3 [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as24e48987 [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 4476b931475a379d11ef80a744d388b2@10.0.0.9:5060 [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:29:49 GMT [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:29:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:29:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:29:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:29:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:29:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 4476b931475a379d11ef80a744d388b2@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:29:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:29:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:29:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:29:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:29:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:29:49] VERBOSE[21156] 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=z9hG4bK011a4ff3;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as24e48987 To: ;tag=as6d59afca Call-ID: 4476b931475a379d11ef80a744d388b2@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:29:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:29:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '4476b931475a379d11ef80a744d388b2@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:30:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '4476b931475a379d11ef80a744d388b2@10.0.0.9:5060' [Jun 20 13:30:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 4476b931475a379d11ef80a744d388b2@10.0.0.9:5060 [Jun 20 13:30:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '4476b931475a379d11ef80a744d388b2@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 52436b32630fd79d0833c0de6cff4788@mydomain.com - OPTIONS (No RTP) [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:30:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '52436b32630fd79d0833c0de6cff4788@mydomain.com' to '06ee95654c883092732030927b7746e1@mydomain.com' [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 06ee95654c883092732030927b7746e1@mydomain.com [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK56caf4a3;rport [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as4d6dd10b [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 06ee95654c883092732030927b7746e1@mydomain.com [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:30:42 GMT [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:30:42] VERBOSE[21156] 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=z9hG4bK56caf4a3;rport Max-Forwards: 70 From: "asterisk" ;tag=as4d6dd10b To: Contact: Call-ID: 06ee95654c883092732030927b7746e1@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:30:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:30:42] VERBOSE[21156] 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=z9hG4bK56caf4a3;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as4d6dd10b To: ;tag=as643fea95 Call-ID: 06ee95654c883092732030927b7746e1@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 13:30:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK56caf4a3;received=10.0.0.31;rport=5100 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as4d6dd10b [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as643fea95 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 06ee95654c883092732030927b7746e1@mydomain.com [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:30:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '06ee95654c883092732030927b7746e1@mydomain.com' of Request 102: Match Found [Jun 20 13:30:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 06ee95654c883092732030927b7746e1@mydomain.com [Jun 20 13:30:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '06ee95654c883092732030927b7746e1@mydomain.com' Method: OPTIONS [Jun 20 13:30:49] VERBOSE[21156] 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=z9hG4bK0379e643 Max-Forwards: 70 From: "asterisk" ;tag=as4f281109 To: Contact: Call-ID: 7fd028ac22f7eea4019744ec30881b18@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:30:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK0379e643 [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as4f281109 [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 7fd028ac22f7eea4019744ec30881b18@10.0.0.9:5060 [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:30:49 GMT [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:30:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:30:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:30:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:30:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:30:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 7fd028ac22f7eea4019744ec30881b18@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:30:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:30:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:30:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:30:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:30:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:30:49] VERBOSE[21156] 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=z9hG4bK0379e643;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as4f281109 To: ;tag=as0ac48023 Call-ID: 7fd028ac22f7eea4019744ec30881b18@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:30:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:30:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '7fd028ac22f7eea4019744ec30881b18@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:31:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '7fd028ac22f7eea4019744ec30881b18@10.0.0.9:5060' [Jun 20 13:31:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 7fd028ac22f7eea4019744ec30881b18@10.0.0.9:5060 [Jun 20 13:31:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '7fd028ac22f7eea4019744ec30881b18@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 5a45d75549cc8c27488ca41825f39640@mydomain.com - OPTIONS (No RTP) [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:31:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '5a45d75549cc8c27488ca41825f39640@mydomain.com' to '63114fda674e3a782b483d6e7dac0761@mydomain.com' [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 63114fda674e3a782b483d6e7dac0761@mydomain.com [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK2a8855a1;rport [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as269689e0 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 63114fda674e3a782b483d6e7dac0761@mydomain.com [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:31:42 GMT [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:31:42] VERBOSE[21156] 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=z9hG4bK2a8855a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as269689e0 To: Contact: Call-ID: 63114fda674e3a782b483d6e7dac0761@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:31:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:31:42] VERBOSE[21156] 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=z9hG4bK2a8855a1;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as269689e0 To: ;tag=as443d746e Call-ID: 63114fda674e3a782b483d6e7dac0761@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 13:31:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK2a8855a1;received=10.0.0.31;rport=5100 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as269689e0 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as443d746e [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 63114fda674e3a782b483d6e7dac0761@mydomain.com [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:31:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16 [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '63114fda674e3a782b483d6e7dac0761@mydomain.com' of Request 102: Match Found [Jun 20 13:31:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 63114fda674e3a782b483d6e7dac0761@mydomain.com [Jun 20 13:31:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '63114fda674e3a782b483d6e7dac0761@mydomain.com' Method: OPTIONS [Jun 20 13:31:49] VERBOSE[21156] 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=z9hG4bK1828398f Max-Forwards: 70 From: "asterisk" ;tag=as7cb84a5f To: Contact: Call-ID: 75f88665265ee3e27f847fdb1bb4b1df@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:31:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK1828398f [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as7cb84a5f [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 75f88665265ee3e27f847fdb1bb4b1df@10.0.0.9:5060 [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:31:49 GMT [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:31:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:31:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:31:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:31:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:31:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 75f88665265ee3e27f847fdb1bb4b1df@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:31:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:31:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:31:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:31:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:31:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:31:49] VERBOSE[21156] 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=z9hG4bK1828398f;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as7cb84a5f To: ;tag=as79534ece Call-ID: 75f88665265ee3e27f847fdb1bb4b1df@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:31:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:31:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '75f88665265ee3e27f847fdb1bb4b1df@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:32:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '75f88665265ee3e27f847fdb1bb4b1df@10.0.0.9:5060' [Jun 20 13:32:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 75f88665265ee3e27f847fdb1bb4b1df@10.0.0.9:5060 [Jun 20 13:32:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '75f88665265ee3e27f847fdb1bb4b1df@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 430152220d9ab6f36bef8c79426600db@mydomain.com - OPTIONS (No RTP) [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:32:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '430152220d9ab6f36bef8c79426600db@mydomain.com' to '3650bbc06cd659c6689fbe1040c9f294@mydomain.com' [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 3650bbc06cd659c6689fbe1040c9f294@mydomain.com [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK5622286f;rport [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as425e03bb [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 3650bbc06cd659c6689fbe1040c9f294@mydomain.com [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:32:42 GMT [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:32:42] VERBOSE[21156] 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=z9hG4bK5622286f;rport Max-Forwards: 70 From: "asterisk" ;tag=as425e03bb To: Contact: Call-ID: 3650bbc06cd659c6689fbe1040c9f294@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:32:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:32:42] VERBOSE[21156] 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=z9hG4bK5622286f;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as425e03bb To: ;tag=as59dac095 Call-ID: 3650bbc06cd659c6689fbe1040c9f294@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 13:32:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK5622286f;received=10.0.0.31;rport=5100 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as425e03bb [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as59dac095 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 3650bbc06cd659c6689fbe1040c9f294@mydomain.com [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:32:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11 [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '3650bbc06cd659c6689fbe1040c9f294@mydomain.com' of Request 102: Match Found [Jun 20 13:32:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 3650bbc06cd659c6689fbe1040c9f294@mydomain.com [Jun 20 13:32:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '3650bbc06cd659c6689fbe1040c9f294@mydomain.com' Method: OPTIONS [Jun 20 13:32:49] VERBOSE[21156] 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=z9hG4bK290c68e6 Max-Forwards: 70 From: "asterisk" ;tag=as087cd4e0 To: Contact: Call-ID: 10261e7b7e4260a41d7349590cd456ea@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:32:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK290c68e6 [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as087cd4e0 [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 10261e7b7e4260a41d7349590cd456ea@10.0.0.9:5060 [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:32:49 GMT [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:32:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:32:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:32:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:32:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:32:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 10261e7b7e4260a41d7349590cd456ea@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:32:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:32:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:32:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:32:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:32:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:32:49] VERBOSE[21156] 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=z9hG4bK290c68e6;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as087cd4e0 To: ;tag=as286eae86 Call-ID: 10261e7b7e4260a41d7349590cd456ea@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:32:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:32:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '10261e7b7e4260a41d7349590cd456ea@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:33:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '10261e7b7e4260a41d7349590cd456ea@10.0.0.9:5060' [Jun 20 13:33:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 10261e7b7e4260a41d7349590cd456ea@10.0.0.9:5060 [Jun 20 13:33:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '10261e7b7e4260a41d7349590cd456ea@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 4a001371329246ef4135e3ef63992046@mydomain.com - OPTIONS (No RTP) [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:33:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '4a001371329246ef4135e3ef63992046@mydomain.com' to '22fbac964fe392393eebc8674053b1bb@mydomain.com' [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 22fbac964fe392393eebc8674053b1bb@mydomain.com [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK707db8f4;rport [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as5b41b66e [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 22fbac964fe392393eebc8674053b1bb@mydomain.com [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:33:42 GMT [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:33:42] VERBOSE[21156] 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=z9hG4bK707db8f4;rport Max-Forwards: 70 From: "asterisk" ;tag=as5b41b66e To: Contact: Call-ID: 22fbac964fe392393eebc8674053b1bb@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:33:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:33:42] VERBOSE[21156] 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=z9hG4bK707db8f4;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as5b41b66e To: ;tag=as267c2cf2 Call-ID: 22fbac964fe392393eebc8674053b1bb@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 13:33:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK707db8f4;received=10.0.0.31;rport=5100 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as5b41b66e [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as267c2cf2 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 22fbac964fe392393eebc8674053b1bb@mydomain.com [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:33:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '22fbac964fe392393eebc8674053b1bb@mydomain.com' of Request 102: Match Found [Jun 20 13:33:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 22fbac964fe392393eebc8674053b1bb@mydomain.com [Jun 20 13:33:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '22fbac964fe392393eebc8674053b1bb@mydomain.com' Method: OPTIONS [Jun 20 13:33:49] VERBOSE[21156] 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=z9hG4bK4c544943 Max-Forwards: 70 From: "asterisk" ;tag=as4c4c5822 To: Contact: Call-ID: 3f120907360fac3d040739b1316484c4@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:33:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK4c544943 [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as4c4c5822 [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 3f120907360fac3d040739b1316484c4@10.0.0.9:5060 [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:33:49 GMT [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:33:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:33:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:33:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:33:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:33:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 3f120907360fac3d040739b1316484c4@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:33:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:33:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:33:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:33:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:33:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:33:49] VERBOSE[21156] 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=z9hG4bK4c544943;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as4c4c5822 To: ;tag=as140a90dc Call-ID: 3f120907360fac3d040739b1316484c4@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:33:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:33:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '3f120907360fac3d040739b1316484c4@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:34:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '3f120907360fac3d040739b1316484c4@10.0.0.9:5060' [Jun 20 13:34:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 3f120907360fac3d040739b1316484c4@10.0.0.9:5060 [Jun 20 13:34:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '3f120907360fac3d040739b1316484c4@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 51575fe01cbcacf273866f495f5d3e16@mydomain.com - OPTIONS (No RTP) [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:34:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '51575fe01cbcacf273866f495f5d3e16@mydomain.com' to '0557e0ab79c4464240b50ada4ef21379@mydomain.com' [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 0557e0ab79c4464240b50ada4ef21379@mydomain.com [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK6549761b;rport [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as1914574f [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 0557e0ab79c4464240b50ada4ef21379@mydomain.com [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:34:42 GMT [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:34:42] VERBOSE[21156] 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=z9hG4bK6549761b;rport Max-Forwards: 70 From: "asterisk" ;tag=as1914574f To: Contact: Call-ID: 0557e0ab79c4464240b50ada4ef21379@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:34:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #10 [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:34:42] VERBOSE[21156] 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=z9hG4bK6549761b;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as1914574f To: ;tag=as6af788cb Call-ID: 0557e0ab79c4464240b50ada4ef21379@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 13:34:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK6549761b;received=10.0.0.31;rport=5100 [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as1914574f [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as6af788cb [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 0557e0ab79c4464240b50ada4ef21379@mydomain.com [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:34:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10 [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '0557e0ab79c4464240b50ada4ef21379@mydomain.com' of Request 102: Match Found [Jun 20 13:34:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 0557e0ab79c4464240b50ada4ef21379@mydomain.com [Jun 20 13:34:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '0557e0ab79c4464240b50ada4ef21379@mydomain.com' Method: OPTIONS [Jun 20 13:34:49] VERBOSE[21156] 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=z9hG4bK56252bc5 Max-Forwards: 70 From: "asterisk" ;tag=as79a2f082 To: Contact: Call-ID: 3fdfc8f55e4b12a757786084093b19c7@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:34:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK56252bc5 [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as79a2f082 [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 3fdfc8f55e4b12a757786084093b19c7@10.0.0.9:5060 [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:34:49 GMT [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:34:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:34:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:34:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:34:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:34:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 3fdfc8f55e4b12a757786084093b19c7@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:34:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:34:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:34:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:34:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:34:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:34:49] VERBOSE[21156] 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=z9hG4bK56252bc5;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as79a2f082 To: ;tag=as69780383 Call-ID: 3fdfc8f55e4b12a757786084093b19c7@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:34:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:34:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '3fdfc8f55e4b12a757786084093b19c7@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:35:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '3fdfc8f55e4b12a757786084093b19c7@10.0.0.9:5060' [Jun 20 13:35:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 3fdfc8f55e4b12a757786084093b19c7@10.0.0.9:5060 [Jun 20 13:35:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '3fdfc8f55e4b12a757786084093b19c7@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 2c4919412d116d595281577476b5a997@mydomain.com - OPTIONS (No RTP) [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:35:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '2c4919412d116d595281577476b5a997@mydomain.com' to '0506c5ba3f9e936b70e328133f828e0e@mydomain.com' [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 0506c5ba3f9e936b70e328133f828e0e@mydomain.com [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK3b60cb6a;rport [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as0388f6ef [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 0506c5ba3f9e936b70e328133f828e0e@mydomain.com [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:35:42 GMT [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:35:42] VERBOSE[21156] 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=z9hG4bK3b60cb6a;rport Max-Forwards: 70 From: "asterisk" ;tag=as0388f6ef To: Contact: Call-ID: 0506c5ba3f9e936b70e328133f828e0e@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:35:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:35:42] VERBOSE[21156] 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=z9hG4bK3b60cb6a;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as0388f6ef To: ;tag=as4ee79055 Call-ID: 0506c5ba3f9e936b70e328133f828e0e@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 13:35:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK3b60cb6a;received=10.0.0.31;rport=5100 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as0388f6ef [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as4ee79055 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 0506c5ba3f9e936b70e328133f828e0e@mydomain.com [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:35:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '0506c5ba3f9e936b70e328133f828e0e@mydomain.com' of Request 102: Match Found [Jun 20 13:35:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 0506c5ba3f9e936b70e328133f828e0e@mydomain.com [Jun 20 13:35:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '0506c5ba3f9e936b70e328133f828e0e@mydomain.com' Method: OPTIONS [Jun 20 13:35:49] VERBOSE[21156] 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=z9hG4bK49093ba2 Max-Forwards: 70 From: "asterisk" ;tag=as7f289dc2 To: Contact: Call-ID: 5db774e04b7de5121716257d7c8d173c@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:35:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK49093ba2 [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as7f289dc2 [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 5db774e04b7de5121716257d7c8d173c@10.0.0.9:5060 [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:35:49 GMT [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:35:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:35:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:35:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:35:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:35:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 5db774e04b7de5121716257d7c8d173c@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:35:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:35:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:35:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:35:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:35:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:35:49] VERBOSE[21156] 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=z9hG4bK49093ba2;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as7f289dc2 To: ;tag=as3d116578 Call-ID: 5db774e04b7de5121716257d7c8d173c@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:35:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:35:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '5db774e04b7de5121716257d7c8d173c@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:36:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '5db774e04b7de5121716257d7c8d173c@10.0.0.9:5060' [Jun 20 13:36:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 5db774e04b7de5121716257d7c8d173c@10.0.0.9:5060 [Jun 20 13:36:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '5db774e04b7de5121716257d7c8d173c@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 27c7346f291318911b2326ad1d70b084@mydomain.com - OPTIONS (No RTP) [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:36:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '27c7346f291318911b2326ad1d70b084@mydomain.com' to '1c93397f67d0d2657573706244a94dc6@mydomain.com' [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 1c93397f67d0d2657573706244a94dc6@mydomain.com [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK06ff9dc4;rport [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as21fc4777 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 1c93397f67d0d2657573706244a94dc6@mydomain.com [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:36:42 GMT [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:36:42] VERBOSE[21156] 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=z9hG4bK06ff9dc4;rport Max-Forwards: 70 From: "asterisk" ;tag=as21fc4777 To: Contact: Call-ID: 1c93397f67d0d2657573706244a94dc6@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:36:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:36:42] VERBOSE[21156] 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=z9hG4bK06ff9dc4;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as21fc4777 To: ;tag=as1047547a Call-ID: 1c93397f67d0d2657573706244a94dc6@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 13:36:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK06ff9dc4;received=10.0.0.31;rport=5100 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as21fc4777 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as1047547a [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 1c93397f67d0d2657573706244a94dc6@mydomain.com [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:36:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '1c93397f67d0d2657573706244a94dc6@mydomain.com' of Request 102: Match Found [Jun 20 13:36:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 1c93397f67d0d2657573706244a94dc6@mydomain.com [Jun 20 13:36:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '1c93397f67d0d2657573706244a94dc6@mydomain.com' Method: OPTIONS [Jun 20 13:36:49] VERBOSE[21156] 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=z9hG4bK51a14f20 Max-Forwards: 70 From: "asterisk" ;tag=as3f174b58 To: Contact: Call-ID: 7f09dad1632aa9866a3fee8b7900d8b7@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:36:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK51a14f20 [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as3f174b58 [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 7f09dad1632aa9866a3fee8b7900d8b7@10.0.0.9:5060 [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:36:49 GMT [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:36:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:36:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:36:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:36:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:36:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 7f09dad1632aa9866a3fee8b7900d8b7@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:36:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:36:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:36:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:36:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:36:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:36:49] VERBOSE[21156] 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=z9hG4bK51a14f20;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as3f174b58 To: ;tag=as467d5d52 Call-ID: 7f09dad1632aa9866a3fee8b7900d8b7@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:36:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:36:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '7f09dad1632aa9866a3fee8b7900d8b7@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:37:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '7f09dad1632aa9866a3fee8b7900d8b7@10.0.0.9:5060' [Jun 20 13:37:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 7f09dad1632aa9866a3fee8b7900d8b7@10.0.0.9:5060 [Jun 20 13:37:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '7f09dad1632aa9866a3fee8b7900d8b7@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 61c8641f2981faf16f3188d859afd7da@mydomain.com - OPTIONS (No RTP) [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:37:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '61c8641f2981faf16f3188d859afd7da@mydomain.com' to '6733f8dd754a21c002e650694a82274a@mydomain.com' [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 6733f8dd754a21c002e650694a82274a@mydomain.com [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1349b1c1;rport [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as5b9c7f97 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 6733f8dd754a21c002e650694a82274a@mydomain.com [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:37:42 GMT [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:37:42] VERBOSE[21156] 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=z9hG4bK1349b1c1;rport Max-Forwards: 70 From: "asterisk" ;tag=as5b9c7f97 To: Contact: Call-ID: 6733f8dd754a21c002e650694a82274a@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:37:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:37:42] VERBOSE[21156] 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=z9hG4bK1349b1c1;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as5b9c7f97 To: ;tag=as17e67ae7 Call-ID: 6733f8dd754a21c002e650694a82274a@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 13:37:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1349b1c1;received=10.0.0.31;rport=5100 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as5b9c7f97 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as17e67ae7 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 6733f8dd754a21c002e650694a82274a@mydomain.com [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:37:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '6733f8dd754a21c002e650694a82274a@mydomain.com' of Request 102: Match Found [Jun 20 13:37:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 6733f8dd754a21c002e650694a82274a@mydomain.com [Jun 20 13:37:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '6733f8dd754a21c002e650694a82274a@mydomain.com' Method: OPTIONS [Jun 20 13:37:49] VERBOSE[21156] 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=z9hG4bK39c90e3c Max-Forwards: 70 From: "asterisk" ;tag=as7a7da535 To: Contact: Call-ID: 27281c6a04f519c645b87c0c450173c5@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:37:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK39c90e3c [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as7a7da535 [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 27281c6a04f519c645b87c0c450173c5@10.0.0.9:5060 [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:37:49 GMT [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:37:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:37:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:37:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:37:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:37:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 27281c6a04f519c645b87c0c450173c5@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:37:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:37:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:37:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:37:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:37:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:37:49] VERBOSE[21156] 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=z9hG4bK39c90e3c;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as7a7da535 To: ;tag=as7392b58c Call-ID: 27281c6a04f519c645b87c0c450173c5@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:37:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:37:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '27281c6a04f519c645b87c0c450173c5@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:38:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '27281c6a04f519c645b87c0c450173c5@10.0.0.9:5060' [Jun 20 13:38:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 27281c6a04f519c645b87c0c450173c5@10.0.0.9:5060 [Jun 20 13:38:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '27281c6a04f519c645b87c0c450173c5@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 62b0e5537ff28a91609d3b010ee0a54a@mydomain.com - OPTIONS (No RTP) [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:38:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '62b0e5537ff28a91609d3b010ee0a54a@mydomain.com' to '11f951f41fd22a1a27de77cd4ba2b22a@mydomain.com' [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 11f951f41fd22a1a27de77cd4ba2b22a@mydomain.com [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK518c64fb;rport [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as3dc014f0 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 11f951f41fd22a1a27de77cd4ba2b22a@mydomain.com [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:38:42 GMT [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:38:42] VERBOSE[21156] 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=z9hG4bK518c64fb;rport Max-Forwards: 70 From: "asterisk" ;tag=as3dc014f0 To: Contact: Call-ID: 11f951f41fd22a1a27de77cd4ba2b22a@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:38:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:38:42] VERBOSE[21156] 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=z9hG4bK518c64fb;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as3dc014f0 To: ;tag=as70457414 Call-ID: 11f951f41fd22a1a27de77cd4ba2b22a@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 13:38:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK518c64fb;received=10.0.0.31;rport=5100 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as3dc014f0 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as70457414 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 11f951f41fd22a1a27de77cd4ba2b22a@mydomain.com [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:38:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16 [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '11f951f41fd22a1a27de77cd4ba2b22a@mydomain.com' of Request 102: Match Found [Jun 20 13:38:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 11f951f41fd22a1a27de77cd4ba2b22a@mydomain.com [Jun 20 13:38:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '11f951f41fd22a1a27de77cd4ba2b22a@mydomain.com' Method: OPTIONS [Jun 20 13:38:49] VERBOSE[21156] 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=z9hG4bK399651fb Max-Forwards: 70 From: "asterisk" ;tag=as5932782e To: Contact: Call-ID: 3ff942f552b1625a212d6c1956489c03@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:38:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK399651fb [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5932782e [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 3ff942f552b1625a212d6c1956489c03@10.0.0.9:5060 [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:38:49 GMT [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:38:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:38:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:38:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:38:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:38:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 3ff942f552b1625a212d6c1956489c03@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:38:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:38:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:38:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:38:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:38:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:38:49] VERBOSE[21156] 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=z9hG4bK399651fb;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as5932782e To: ;tag=as7782823d Call-ID: 3ff942f552b1625a212d6c1956489c03@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:38:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:38:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '3ff942f552b1625a212d6c1956489c03@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:39:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '3ff942f552b1625a212d6c1956489c03@10.0.0.9:5060' [Jun 20 13:39:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 3ff942f552b1625a212d6c1956489c03@10.0.0.9:5060 [Jun 20 13:39:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '3ff942f552b1625a212d6c1956489c03@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 5c78f3c51c5244d844f652966ed7193a@mydomain.com - OPTIONS (No RTP) [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:39:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '5c78f3c51c5244d844f652966ed7193a@mydomain.com' to '4fb94f41016003720ed97c480b84c491@mydomain.com' [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 4fb94f41016003720ed97c480b84c491@mydomain.com [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK438c60ce;rport [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as30146852 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 4fb94f41016003720ed97c480b84c491@mydomain.com [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:39:42 GMT [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:39:42] VERBOSE[21156] 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=z9hG4bK438c60ce;rport Max-Forwards: 70 From: "asterisk" ;tag=as30146852 To: Contact: Call-ID: 4fb94f41016003720ed97c480b84c491@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:39:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:39:42] VERBOSE[21156] 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=z9hG4bK438c60ce;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as30146852 To: ;tag=as524c9f81 Call-ID: 4fb94f41016003720ed97c480b84c491@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 13:39:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK438c60ce;received=10.0.0.31;rport=5100 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as30146852 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as524c9f81 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 4fb94f41016003720ed97c480b84c491@mydomain.com [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:39:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11 [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '4fb94f41016003720ed97c480b84c491@mydomain.com' of Request 102: Match Found [Jun 20 13:39:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 4fb94f41016003720ed97c480b84c491@mydomain.com [Jun 20 13:39:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '4fb94f41016003720ed97c480b84c491@mydomain.com' Method: OPTIONS [Jun 20 13:39:49] VERBOSE[21156] 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=z9hG4bK4be8e432 Max-Forwards: 70 From: "asterisk" ;tag=as6bd1bde8 To: Contact: Call-ID: 41c759cc2278dd493a96bb3473c0afa8@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:39:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK4be8e432 [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6bd1bde8 [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 41c759cc2278dd493a96bb3473c0afa8@10.0.0.9:5060 [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:39:49 GMT [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:39:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:39:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:39:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:39:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:39:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 41c759cc2278dd493a96bb3473c0afa8@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:39:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:39:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:39:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:39:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:39:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:39:49] VERBOSE[21156] 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=z9hG4bK4be8e432;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as6bd1bde8 To: ;tag=as29a1a572 Call-ID: 41c759cc2278dd493a96bb3473c0afa8@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:39:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:39:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '41c759cc2278dd493a96bb3473c0afa8@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:40:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '41c759cc2278dd493a96bb3473c0afa8@10.0.0.9:5060' [Jun 20 13:40:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 41c759cc2278dd493a96bb3473c0afa8@10.0.0.9:5060 [Jun 20 13:40:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '41c759cc2278dd493a96bb3473c0afa8@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 20639311352da58545074947622ed4b0@mydomain.com - OPTIONS (No RTP) [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:40:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '20639311352da58545074947622ed4b0@mydomain.com' to '477eabbc0bd102d93ed26e5f544f7523@mydomain.com' [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 477eabbc0bd102d93ed26e5f544f7523@mydomain.com [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK05b2f829;rport [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as410a6e71 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 477eabbc0bd102d93ed26e5f544f7523@mydomain.com [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:40:42 GMT [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:40:42] VERBOSE[21156] 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=z9hG4bK05b2f829;rport Max-Forwards: 70 From: "asterisk" ;tag=as410a6e71 To: Contact: Call-ID: 477eabbc0bd102d93ed26e5f544f7523@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:40:42] VERBOSE[21156] 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=z9hG4bK05b2f829;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as410a6e71 To: ;tag=as3cfe58a2 Call-ID: 477eabbc0bd102d93ed26e5f544f7523@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 13:40:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK05b2f829;received=10.0.0.31;rport=5100 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as410a6e71 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as3cfe58a2 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 477eabbc0bd102d93ed26e5f544f7523@mydomain.com [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:40:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '477eabbc0bd102d93ed26e5f544f7523@mydomain.com' of Request 102: Match Found [Jun 20 13:40:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 477eabbc0bd102d93ed26e5f544f7523@mydomain.com [Jun 20 13:40:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '477eabbc0bd102d93ed26e5f544f7523@mydomain.com' Method: OPTIONS [Jun 20 13:40:49] VERBOSE[21156] 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=z9hG4bK377c8796 Max-Forwards: 70 From: "asterisk" ;tag=as5629fefc To: Contact: Call-ID: 6a0d3f5a66551e1c5473e2fa1ea215a8@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:40:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK377c8796 [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5629fefc [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 6a0d3f5a66551e1c5473e2fa1ea215a8@10.0.0.9:5060 [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:40:49 GMT [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:40:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:40:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:40:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:40:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:40:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 6a0d3f5a66551e1c5473e2fa1ea215a8@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:40:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:40:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:40:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:40:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:40:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:40:49] VERBOSE[21156] 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=z9hG4bK377c8796;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as5629fefc To: ;tag=as03db4897 Call-ID: 6a0d3f5a66551e1c5473e2fa1ea215a8@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:40:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:40:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '6a0d3f5a66551e1c5473e2fa1ea215a8@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:41:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '6a0d3f5a66551e1c5473e2fa1ea215a8@10.0.0.9:5060' [Jun 20 13:41:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 6a0d3f5a66551e1c5473e2fa1ea215a8@10.0.0.9:5060 [Jun 20 13:41:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '6a0d3f5a66551e1c5473e2fa1ea215a8@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 582646e14810fbd61a45a11151d2cab2@mydomain.com - OPTIONS (No RTP) [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:41:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '582646e14810fbd61a45a11151d2cab2@mydomain.com' to '0bc122d1297f28e93d98bb4b0e630cf1@mydomain.com' [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 0bc122d1297f28e93d98bb4b0e630cf1@mydomain.com [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1133f168;rport [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as4ac2aac4 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 0bc122d1297f28e93d98bb4b0e630cf1@mydomain.com [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:41:42 GMT [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:41:42] VERBOSE[21156] 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=z9hG4bK1133f168;rport Max-Forwards: 70 From: "asterisk" ;tag=as4ac2aac4 To: Contact: Call-ID: 0bc122d1297f28e93d98bb4b0e630cf1@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:41:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #10 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:41:42] VERBOSE[21156] 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=z9hG4bK1133f168;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as4ac2aac4 To: ;tag=as027c6621 Call-ID: 0bc122d1297f28e93d98bb4b0e630cf1@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 13:41:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK1133f168;received=10.0.0.31;rport=5100 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as4ac2aac4 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as027c6621 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 0bc122d1297f28e93d98bb4b0e630cf1@mydomain.com [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:41:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10 [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '0bc122d1297f28e93d98bb4b0e630cf1@mydomain.com' of Request 102: Match Found [Jun 20 13:41:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 0bc122d1297f28e93d98bb4b0e630cf1@mydomain.com [Jun 20 13:41:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '0bc122d1297f28e93d98bb4b0e630cf1@mydomain.com' Method: OPTIONS [Jun 20 13:41:49] VERBOSE[21156] 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=z9hG4bK06a9dda2 Max-Forwards: 70 From: "asterisk" ;tag=as6f75d957 To: Contact: Call-ID: 49217fc84d0ac8896dd4b9725d9309d4@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:41:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK06a9dda2 [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6f75d957 [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 49217fc84d0ac8896dd4b9725d9309d4@10.0.0.9:5060 [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:41:49 GMT [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:41:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:41:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:41:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:41:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:41:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 49217fc84d0ac8896dd4b9725d9309d4@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:41:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:41:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:41:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:41:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:41:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:41:49] VERBOSE[21156] 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=z9hG4bK06a9dda2;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as6f75d957 To: ;tag=as5a3466ff Call-ID: 49217fc84d0ac8896dd4b9725d9309d4@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:41:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:41:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '49217fc84d0ac8896dd4b9725d9309d4@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:42:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '49217fc84d0ac8896dd4b9725d9309d4@10.0.0.9:5060' [Jun 20 13:42:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 49217fc84d0ac8896dd4b9725d9309d4@10.0.0.9:5060 [Jun 20 13:42:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '49217fc84d0ac8896dd4b9725d9309d4@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 0625a83f1ee705f14c237f9d4f8210df@mydomain.com - OPTIONS (No RTP) [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:42:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '0625a83f1ee705f14c237f9d4f8210df@mydomain.com' to '63aa7cd51f2da8e1570ecd5d392492fa@mydomain.com' [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 63aa7cd51f2da8e1570ecd5d392492fa@mydomain.com [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK67c2fe29;rport [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as015a38fb [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 63aa7cd51f2da8e1570ecd5d392492fa@mydomain.com [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:42:42 GMT [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:42:42] VERBOSE[21156] 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=z9hG4bK67c2fe29;rport Max-Forwards: 70 From: "asterisk" ;tag=as015a38fb To: Contact: Call-ID: 63aa7cd51f2da8e1570ecd5d392492fa@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:42:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:42:42] VERBOSE[21156] 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=z9hG4bK67c2fe29;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as015a38fb To: ;tag=as779d747f Call-ID: 63aa7cd51f2da8e1570ecd5d392492fa@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 13:42:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK67c2fe29;received=10.0.0.31;rport=5100 [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as015a38fb [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as779d747f [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 63aa7cd51f2da8e1570ecd5d392492fa@mydomain.com [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:42:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '63aa7cd51f2da8e1570ecd5d392492fa@mydomain.com' of Request 102: Match Found [Jun 20 13:42:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 63aa7cd51f2da8e1570ecd5d392492fa@mydomain.com [Jun 20 13:42:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '63aa7cd51f2da8e1570ecd5d392492fa@mydomain.com' Method: OPTIONS [Jun 20 13:42:49] VERBOSE[21156] 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=z9hG4bK66879e80 Max-Forwards: 70 From: "asterisk" ;tag=as644ec4f5 To: Contact: Call-ID: 42f9599f5e35c44118b6d3f128f7922e@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:42:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK66879e80 [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as644ec4f5 [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 42f9599f5e35c44118b6d3f128f7922e@10.0.0.9:5060 [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:42:49 GMT [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:42:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:42:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:42:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:42:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:42:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 42f9599f5e35c44118b6d3f128f7922e@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:42:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:42:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:42:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:42:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:42:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:42:49] VERBOSE[21156] 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=z9hG4bK66879e80;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as644ec4f5 To: ;tag=as49635a11 Call-ID: 42f9599f5e35c44118b6d3f128f7922e@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:42:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:42:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '42f9599f5e35c44118b6d3f128f7922e@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:43:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '42f9599f5e35c44118b6d3f128f7922e@10.0.0.9:5060' [Jun 20 13:43:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 42f9599f5e35c44118b6d3f128f7922e@10.0.0.9:5060 [Jun 20 13:43:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '42f9599f5e35c44118b6d3f128f7922e@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 5c3ca75959bf58ee50f02e0a219d5009@mydomain.com - OPTIONS (No RTP) [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:43:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '5c3ca75959bf58ee50f02e0a219d5009@mydomain.com' to '79574997615610921acec53e75d28ac0@mydomain.com' [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 79574997615610921acec53e75d28ac0@mydomain.com [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK3462875d;rport [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as10c1f322 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 79574997615610921acec53e75d28ac0@mydomain.com [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:43:42 GMT [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:43:42] VERBOSE[21156] 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=z9hG4bK3462875d;rport Max-Forwards: 70 From: "asterisk" ;tag=as10c1f322 To: Contact: Call-ID: 79574997615610921acec53e75d28ac0@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:43:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:43:42] VERBOSE[21156] 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=z9hG4bK3462875d;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as10c1f322 To: ;tag=as5d638c09 Call-ID: 79574997615610921acec53e75d28ac0@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 13:43:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK3462875d;received=10.0.0.31;rport=5100 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as10c1f322 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as5d638c09 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 79574997615610921acec53e75d28ac0@mydomain.com [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:43:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '79574997615610921acec53e75d28ac0@mydomain.com' of Request 102: Match Found [Jun 20 13:43:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 79574997615610921acec53e75d28ac0@mydomain.com [Jun 20 13:43:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '79574997615610921acec53e75d28ac0@mydomain.com' Method: OPTIONS [Jun 20 13:43:49] VERBOSE[21156] 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=z9hG4bK17a4ae47 Max-Forwards: 70 From: "asterisk" ;tag=as5b63103e To: Contact: Call-ID: 4d5af5757d81db5c266c300b74e0768a@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:43:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK17a4ae47 [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as5b63103e [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 4d5af5757d81db5c266c300b74e0768a@10.0.0.9:5060 [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:43:49 GMT [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:43:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:43:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:43:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:43:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:43:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 4d5af5757d81db5c266c300b74e0768a@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:43:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:43:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:43:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:43:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:43:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:43:49] VERBOSE[21156] 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=z9hG4bK17a4ae47;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as5b63103e To: ;tag=as011ac21a Call-ID: 4d5af5757d81db5c266c300b74e0768a@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:43:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:43:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '4d5af5757d81db5c266c300b74e0768a@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:44:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '4d5af5757d81db5c266c300b74e0768a@10.0.0.9:5060' [Jun 20 13:44:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 4d5af5757d81db5c266c300b74e0768a@10.0.0.9:5060 [Jun 20 13:44:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '4d5af5757d81db5c266c300b74e0768a@10.0.0.9:5060' Method: OPTIONS [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 1fe4a1ad1f4a256e50a7f8957955c5ed@mydomain.com - OPTIONS (No RTP) [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 20 13:44:42] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: SIP call-id changed from '1fe4a1ad1f4a256e50a7f8957955c5ed@mydomain.com' to '53fb36304ca14e8031a973ee460cc63a@mydomain.com' [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Initializing initreq for method OPTIONS - callid 53fb36304ca14e8031a973ee460cc63a@mydomain.com [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 0 [ 32]: OPTIONS sip:10.0.0.9 SIP/2.0 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK7c341ae7;rport [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 3 [ 70]: From: "asterisk" ;tag=as3392f072 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 4 [ 22]: To: [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 5 [ 42]: Contact: [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 6 [ 60]: Call-ID: 53fb36304ca14e8031a973ee460cc63a@mydomain.com [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.9.1 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:44:42 GMT [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:44:42] VERBOSE[21156] 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=z9hG4bK7c341ae7;rport Max-Forwards: 70 From: "asterisk" ;tag=as3392f072 To: Contact: Call-ID: 53fb36304ca14e8031a973ee460cc63a@mydomain.com CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.9.1 Date: Mon, 20 Jun 2016 11:44:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:44:42] VERBOSE[21156] 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=z9hG4bK7c341ae7;received=10.0.0.31;rport=5100 From: "asterisk" ;tag=as3392f072 To: ;tag=as0a9fe569 Call-ID: 53fb36304ca14e8031a973ee460cc63a@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 13:44:42] DEBUG[21156] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 10.0.0.31:5100;branch=z9hG4bK7c341ae7;received=10.0.0.31;rport=5100 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 2 [ 70]: From: "asterisk" ;tag=as3392f072 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 3 [ 37]: To: ;tag=as0a9fe569 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 4 [ 60]: Call-ID: 53fb36304ca14e8031a973ee460cc63a@mydomain.com [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 6 [ 22]: Server: Digium Gateway [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 9 [ 23]: Accept: application/sdp [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Header 11 [ 0]: [Jun 20 13:44:42] VERBOSE[21156] chan_sip.c: --- (11 headers 0 lines) --- [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Stopping retransmission on '53fb36304ca14e8031a973ee460cc63a@mydomain.com' of Request 102: Match Found [Jun 20 13:44:42] DEBUG[21156] chan_sip.c: Destroying SIP dialog 53fb36304ca14e8031a973ee460cc63a@mydomain.com [Jun 20 13:44:42] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '53fb36304ca14e8031a973ee460cc63a@mydomain.com' Method: OPTIONS [Jun 20 13:44:49] VERBOSE[21156] 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=z9hG4bK55091b2f Max-Forwards: 70 From: "asterisk" ;tag=as6adc5773 To: Contact: Call-ID: 214e78a2355859ef156e4d736ac91481@10.0.0.9:5060 CSeq: 102 OPTIONS User-Agent: Digium Gateway Date: Mon, 20 Jun 2016 11:44:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 0 [ 36]: OPTIONS sip:asterisk.mydomain.com SIP/2.0 [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK55091b2f [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6adc5773 [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 4 [ 26]: To: [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 5 [ 41]: Contact: [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 6 [ 59]: Call-ID: 214e78a2355859ef156e4d736ac91481@10.0.0.9:5060 [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 8 [ 26]: User-Agent: Digium Gateway [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 9 [ 35]: Date: Mon, 20 Jun 2016 11:44:49 GMT [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Header 13 [ 0]: [Jun 20 13:44:49] VERBOSE[21156] chan_sip.c: --- (13 headers 0 lines) --- [Jun 20 13:44:49] DEBUG[21156] acl.c: For destination '10.0.0.9', our source address is '10.0.0.31'. [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.0.0.31:5100 [Jun 20 13:44:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9:5060' into... [Jun 20 13:44:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port '5060'. [Jun 20 13:44:49] VERBOSE[21156] chan_sip.c: Sending to 10.0.0.9:5060 (NAT) [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Allocating new SIP dialog for 214e78a2355859ef156e4d736ac91481@10.0.0.9:5060 - OPTIONS (No RTP) [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jun 20 13:44:49] DEBUG[21156] netsock2.c: Splitting 'asterisk.mydomain.com' into... [Jun 20 13:44:49] DEBUG[21156] netsock2.c: ...host 'asterisk.mydomain.com' and port ''. [Jun 20 13:44:49] DEBUG[21156] netsock2.c: Splitting '10.0.0.9' into... [Jun 20 13:44:49] DEBUG[21156] netsock2.c: ...host '10.0.0.9' and port ''. [Jun 20 13:44:49] VERBOSE[21156] chan_sip.c: Looking for s in default (domain asterisk.mydomain.com) [Jun 20 13:44:49] VERBOSE[21156] 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=z9hG4bK55091b2f;received=10.0.0.9;rport=5060 From: "asterisk" ;tag=as6adc5773 To: ;tag=as449c7b7e Call-ID: 214e78a2355859ef156e4d736ac91481@10.0.0.9:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.9.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jun 20 13:44:49] DEBUG[21156] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.0.9:5060 [Jun 20 13:44:49] VERBOSE[21156] chan_sip.c: Scheduling destruction of SIP dialog '214e78a2355859ef156e4d736ac91481@10.0.0.9:5060' in 32000 ms (Method: OPTIONS) [Jun 20 13:45:21] DEBUG[21156] chan_sip.c: Auto destroying SIP dialog '214e78a2355859ef156e4d736ac91481@10.0.0.9:5060' [Jun 20 13:45:21] DEBUG[21156] chan_sip.c: Destroying SIP dialog 214e78a2355859ef156e4d736ac91481@10.0.0.9:5060 [Jun 20 13:45:21] VERBOSE[21156] chan_sip.c: Really destroying SIP dialog '214e78a2355859ef156e4d736ac91481@10.0.0.9:5060' Method: OPTIONS