[Mar 1 12:35:57] DEBUG[8269] chan_sip.c: Auto destroying SIP dialog 'yjoevijubpguvtr@192.168.7.55' [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: Destroying SIP dialog yjoevijubpguvtr@192.168.7.55 [Mar 1 12:35:57] VERBOSE[8269] chan_sip.c: Really destroying SIP dialog 'yjoevijubpguvtr@192.168.7.55' Method: ACK [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: ---------- SIP HISTORY for 'yjoevijubpguvtr@192.168.7.55' [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: * SIP Call [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: 001. Rx INVITE / 760 INVITE / sip:123@dev-sip.tele500.com [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: 002. NewChan Channel SIP/dev-sip.tele500.com-082016d8 - from yjoevijubpguvtr [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: 003. TxResp SIP/2.0 / 760 INVITE - 100 Trying [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: 004. Cancel Cause No user responding [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: 005. SchedDestroy 32000 ms [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: 006. TxRespRel SIP/2.0 / 760 INVITE - 408 Request Timeout [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: 007. Rx ACK / 760 ACK / sip:123@dev-sip.tele500.com [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: 008. AutoDestroy yjoevijubpguvtr@192.168.7.55 [Mar 1 12:35:57] DEBUG[8269] chan_sip.c: ---------- END SIP HISTORY for 'yjoevijubpguvtr@192.168.7.55' [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: <--- SIP read from UDP://{OpenSER_IP}:5060 ---> INVITE sip:123@dev-sip.tele500.com SIP/2.0 Record-Route: Via: SIP/2.0/UDP {OpenSER_IP};branch=z9hG4bK18a8.ca91afd.0 Via: SIP/2.0/UDP {UA_IP}:5061;received={UA_IP};rport=5061;branch=z9hG4bKdjvhchet Max-Forwards: 69 To: From: "10000" ;tag=qjkyu Call-ID: btomwmxyvjserjv@192.168.7.55 CSeq: 708 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.1 Content-Length: 252 X-auth: 10000 v=0 o=10000 1334618652 2083910151 IN IP4 192.168.7.55 s=- c=IN IP4 {UA_IP} t=0 0 m=audio 8002 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 0 [ 42]: INVITE sip:123@dev-sip.tele500.com SIP/2.0 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 1 [ 49]: Record-Route: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 2 [ 58]: Via: SIP/2.0/UDP {OpenSER_IP};branch=z9hG4bK18a8.ca91afd.0 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 3 [ 90]: Via: SIP/2.0/UDP {UA_IP}:5061;received={UA_IP};rport=5061;branch=z9hG4bKdjvhchet [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 4 [ 16]: Max-Forwards: 69 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 5 [ 33]: To: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 6 [ 55]: From: "10000" ;tag=qjkyu [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 7 [ 37]: Call-ID: btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 8 [ 16]: CSeq: 708 INVITE [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 9 [ 38]: Contact: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 11 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 12 [ 37]: Supported: replaces,norefersub,100rel [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 13 [ 23]: User-Agent: Twinkle/1.1 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 14 [ 19]: Content-Length: 252 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 15 [ 13]: X-auth: 10000 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 16 [ 0]: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 0 [ 3]: v=0 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 1 [ 49]: o=10000 1334618652 2083910151 IN IP4 192.168.7.55 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 2 [ 3]: s=- [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 3 [ 21]: c=IN IP4 {UA_IP} [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 5 [ 30]: m=audio 8002 RTP/AVP 8 0 3 101 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: --- (16 headers 12 lines) --- [Mar 1 12:36:06] DEBUG[8269] acl.c: Found IP address for this socket [Mar 1 12:36:06] VERBOSE[8269] netsock.c: == Using SIP RTP CoS mark 5 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Setting NAT on RTP to Off [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Allocating new SIP dialog for btomwmxyvjserjv@192.168.7.55 - INVITE (With RTP) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Begin: parsing SIP "Supported: replaces,norefersub,100rel" [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Found SIP option: -replaces- [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Matched SIP option: replaces [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Found SIP option: -norefersub- [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Matched SIP option: norefersub [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Found SIP option: -100rel- [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Matched SIP option: 100rel [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Sending to {OpenSER_IP} : 5060 (no NAT) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Initializing initreq for method INVITE - callid btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Using INVITE request as basis request - btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found peer 'OpenSER' for '10000' from {OpenSER_IP}:5060 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Setting NAT on RTP to Off [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found RTP audio format 8 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found RTP audio format 0 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found RTP audio format 3 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found RTP audio format 101 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Peer audio RTP is at port {UA_IP}:8002 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found audio description format GSM for ID 3 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Got unsupported a:fmtp in SDP offer [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Capabilities: us - 0x140f (g723|gsm|ulaw|alaw|ilbc|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Peer audio RTP is at port {UA_IP}:8002 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Checking SIP call limits for device [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Updating call counter for incoming call [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: Looking for 123 in common (domain dev-sip.tele500.com) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: *** Our native formats are 0x8 (alaw) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: *** Our capabilities are 0x140f (g723|gsm|ulaw|alaw|ilbc|g722) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: This channel will not be able to handle video. [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: build_route: Record-Route hop: [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: list_route: hop: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Session timer started: 13 - btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: SIP/dev-sip.tele500.com-08206af0: New call is still down.... Trying... [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: <--- Transmitting (no NAT) to {OpenSER_IP}:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP {OpenSER_IP};branch=z9hG4bK18a8.ca91afd.0;received={OpenSER_IP} Via: SIP/2.0/UDP {UA_IP}:5061;received={UA_IP};rport=5061;branch=z9hG4bKdjvhchet Record-Route: From: "10000" ;tag=qjkyu To: Call-ID: btomwmxyvjserjv@192.168.7.55 CSeq: 708 INVITE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 1 [ 80]: Via: SIP/2.0/UDP {OpenSER_IP};branch=z9hG4bK18a8.ca91afd.0;received={OpenSER_IP} [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 2 [ 90]: Via: SIP/2.0/UDP {UA_IP}:5061;received={UA_IP};rport=5061;branch=z9hG4bKdjvhchet [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 3 [ 49]: Record-Route: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 4 [ 55]: From: "10000" ;tag=qjkyu [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 5 [ 33]: To: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 6 [ 37]: Call-ID: btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 7 [ 16]: CSeq: 708 INVITE [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 8 [ 18]: Server: Media GW 1 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 9 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 11 [ 31]: Contact: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 13 [ 0]: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for {OpenSER_IP}:5060 [Mar 1 12:36:06] DEBUG[8266] devicestate.c: No provider found, checking channel drivers for SIP - dev-sip.tele500.com [Mar 1 12:36:06] DEBUG[8266] chan_sip.c: Checking device state for peer dev-sip.tele500.com [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8290] pbx.c: Launching 'Dial' [Mar 1 12:36:06] VERBOSE[8290] pbx.c: -- Executing [123@common:1] Dial("SIP/dev-sip.tele500.com-08206af0", "SIP/442088123456@Magrathea,45") in new stack [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] channel.c: Avoiding initial deadlock for channel '0x82049d8' [Mar 1 12:36:06] DEBUG[8266] devicestate.c: Changing state for SIP/dev-sip.tele500.com - state 2 (In use) [Mar 1 12:36:06] DEBUG[8266] devicestate.c: device 'SIP/dev-sip.tele500.com' state '2' [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Mar 1 12:36:06] VERBOSE[8290] netsock.c: == Using SIP RTP CoS mark 5 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Allocating new SIP dialog for 1701bf4f73ad4c871973ca2b0c6031b0@{Asterisk_IP} - INVITE (With RTP) [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Setting NAT on RTP to Off [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: OBPROXY: Applying global OBproxy to this call [Mar 1 12:36:06] DEBUG[8290] acl.c: Found IP address for this socket [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: *** Our native formats are 0x8 (alaw) [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: *** Our capabilities are 0x140f (g723|gsm|ulaw|alaw|ilbc|g722) [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: This channel will not be able to handle video. [Mar 1 12:36:06] DEBUG[8265] devicestate.c: Processing device state change for 'SIP/dev-sip.tele500.com' [Mar 1 12:36:06] DEBUG[8265] devicestate.c: Adding per-server state of 'In use' for 'SIP/dev-sip.tele500.com' [Mar 1 12:36:06] DEBUG[8265] devicestate.c: Aggregate devstate result is 2 [Mar 1 12:36:06] DEBUG[8265] devicestate.c: Aggregate state for device 'SIP/dev-sip.tele500.com' has changed to 'In use' [Mar 1 12:36:06] DEBUG[8290] channel.c: Not copying variable DIALEDTIME. [Mar 1 12:36:06] DEBUG[8290] channel.c: Not copying variable ANSWEREDTIME. [Mar 1 12:36:06] DEBUG[8290] channel.c: Not copying variable DIALEDPEERNAME. [Mar 1 12:36:06] DEBUG[8290] channel.c: Not copying variable DIALEDPEERNUMBER. [Mar 1 12:36:06] DEBUG[8290] channel.c: Not copying variable DIALSTATUS. [Mar 1 12:36:06] DEBUG[8290] channel.c: Not copying variable SIPCALLID. [Mar 1 12:36:06] DEBUG[8290] channel.c: Not copying variable SIPDOMAIN. [Mar 1 12:36:06] DEBUG[8290] channel.c: Not copying variable SIPURI. [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Outgoing Call for 442088123456 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Updating call counter for outgoing call [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False Text flag: False [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Mar 1 12:36:06] VERBOSE[8290] chan_sip.c: Audio is at {Asterisk_IP} port 18220 [Mar 1 12:36:06] VERBOSE[8290] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Mar 1 12:36:06] VERBOSE[8290] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Mar 1 12:36:06] VERBOSE[8290] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Mar 1 12:36:06] VERBOSE[8290] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: -- Done with adding codecs to SDP [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Initializing initreq for method INVITE - callid 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 0 [ 47]: INVITE sip:442088123456@pstn.wima.co.uk SIP/2.0 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 3 [ 53]: From: "10000" ;tag=as25e845f5 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 4 [ 38]: To: [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 5 [ 33]: Contact: [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 6 [ 54]: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 8 [ 22]: User-Agent: Media GW 1 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 9 [ 35]: Date: Sun, 01 Mar 2009 12:36:06 GMT [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 12 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 15 [ 19]: Content-Length: 300 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 16 [ 0]: [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 0 [ 3]: v=0 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 1 [ 48]: o=root 2057975154 2057975154 IN IP4 {Asterisk_IP} [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 2 [ 12]: s=Media GW 1 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 3 [ 21]: c=IN IP4 {Asterisk_IP} [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 5 [ 31]: m=audio 18220 RTP/AVP 8 0 3 101 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 13 [ 10]: a=sendrecv [Mar 1 12:36:06] VERBOSE[8290] chan_sip.c: Reliably Transmitting (no NAT) to {OpenSER_IP}:5060: INVITE sip:442088123456@pstn.wima.co.uk SIP/2.0 Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport Max-Forwards: 70 From: "10000" ;tag=as25e845f5 To: Contact: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} CSeq: 102 INVITE User-Agent: Media GW 1 Date: Sun, 01 Mar 2009 12:36:06 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 300 v=0 o=root 2057975154 2057975154 IN IP4 {Asterisk_IP} s=Media GW 1 c=IN IP4 {Asterisk_IP} t=0 0 m=audio 18220 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 0 [ 47]: INVITE sip:442088123456@pstn.wima.co.uk SIP/2.0 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 3 [ 53]: From: "10000" ;tag=as25e845f5 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 4 [ 38]: To: [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 5 [ 33]: Contact: [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 6 [ 54]: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 8 [ 22]: User-Agent: Media GW 1 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 9 [ 35]: Date: Sun, 01 Mar 2009 12:36:06 GMT [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 12 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 15 [ 19]: Content-Length: 300 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Header 16 [ 0]: [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 0 [ 3]: v=0 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 1 [ 48]: o=root 2057975154 2057975154 IN IP4 {Asterisk_IP} [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 2 [ 12]: s=Media GW 1 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 3 [ 21]: c=IN IP4 {Asterisk_IP} [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 5 [ 31]: m=audio 18220 RTP/AVP 8 0 3 101 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Body 13 [ 10]: a=sendrecv [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Mar 1 12:36:06] DEBUG[8290] chan_sip.c: Trying to put 'INVITE sip' onto UDP socket destined for {OpenSER_IP}:5060 [Mar 1 12:36:06] VERBOSE[8290] app_dial.c: -- Called 442088123456@Magrathea [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: <--- SIP read from UDP://{OpenSER_IP}:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport=5060 From: "10000" ;tag=as25e845f5 To: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} CSeq: 102 INVITE Server: SIP Proxy 1 Content-Length: 0 <-------------> [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport=5060 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 2 [ 53]: From: "10000" ;tag=as25e845f5 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 3 [ 38]: To: [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 4 [ 54]: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 6 [ 19]: Server: SIP Proxy 1 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: Header 8 [ 0]: [Mar 1 12:36:06] VERBOSE[8269] chan_sip.c: --- (8 headers 0 lines) --- [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: *** SIP TIMER: Cancelling retransmission #14 - INVITE (got response) [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}' Request 102: Found [Mar 1 12:36:06] DEBUG[8269] chan_sip.c: SIP response 100 to standard invite [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: <--- SIP read from UDP://{OpenSER_IP}:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport=5060 From: "10000" ;tag=as25e845f5 To: ;tag=78b11732fd1718be878e22d27662fcac-6cbc Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} CSeq: 102 INVITE Server: SIP Proxy 1 Content-Length: 0 <-------------> [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 0 [ 27]: SIP/2.0 408 Request Timeout [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport=5060 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 2 [ 53]: From: "10000" ;tag=as25e845f5 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 3 [ 80]: To: ;tag=78b11732fd1718be878e22d27662fcac-6cbc [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 4 [ 54]: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 6 [ 19]: Server: SIP Proxy 1 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 8 [ 0]: [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: --- (8 headers 0 lines) --- [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Acked pending invite 102 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Stopping retransmission on '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}' of Request 102: Match Found [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: SIP response 408 to standard invite [Mar 1 12:36:08] WARNING[8269] chan_sip.c: Re-invite to non-existing call leg on other UA. SIP dialog '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}'. Giving up. [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: Transmitting (no NAT) to {OpenSER_IP}:5060: ACK sip:442088123456@pstn.wima.co.uk SIP/2.0 Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport Max-Forwards: 70 From: "10000" ;tag=as25e845f5 To: ;tag=78b11732fd1718be878e22d27662fcac-6cbc Contact: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} CSeq: 102 ACK User-Agent: Media GW 1 Content-Length: 0 --- [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 0 [ 44]: ACK sip:442088123456@pstn.wima.co.uk SIP/2.0 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK16ab7e1a;rport [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 3 [ 53]: From: "10000" ;tag=as25e845f5 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 4 [ 80]: To: ;tag=78b11732fd1718be878e22d27662fcac-6cbc [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 5 [ 33]: Contact: [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 6 [ 54]: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 8 [ 22]: User-Agent: Media GW 1 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 10 [ 0]: [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Trying to put 'ACK sip:44' onto UDP socket destined for {OpenSER_IP}:5060 [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: Scheduling destruction of SIP dialog '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}' in 32000 ms (Method: INVITE) [Mar 1 12:36:08] VERBOSE[8290] app_dial.c: -- SIP/Magrathea-0820cf20 is circuit-busy [Mar 1 12:36:08] DEBUG[8290] channel.c: Hanging up channel 'SIP/Magrathea-0820cf20' [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Hangup call SIP/Magrathea-0820cf20, SIP callid 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:08] VERBOSE[8290] chan_sip.c: Scheduling destruction of SIP dialog '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}' in 32000 ms (Method: INVITE) [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Session timer stopped: -1 - 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:08] VERBOSE[8290] chan_sip.c: Reliably Transmitting (no NAT) to {OpenSER_IP}:5060: BYE sip:442088123456@pstn.wima.co.uk SIP/2.0 Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK04892ce3;rport Max-Forwards: 70 From: "10000" ;tag=as25e845f5 To: ;tag=78b11732fd1718be878e22d27662fcac-6cbc Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} CSeq: 103 BYE User-Agent: Media GW 1 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0 --- [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 0 [ 44]: BYE sip:442088123456@pstn.wima.co.uk SIP/2.0 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK04892ce3;rport [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 3 [ 53]: From: "10000" ;tag=as25e845f5 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 4 [ 80]: To: ;tag=78b11732fd1718be878e22d27662fcac-6cbc [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 5 [ 54]: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 6 [ 13]: CSeq: 103 BYE [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 7 [ 22]: User-Agent: Media GW 1 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 8 [ 42]: X-Asterisk-HangupCause: No user responding [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 18 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 11 [ 0]: [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Trying to put 'BYE sip:44' onto UDP socket destined for {OpenSER_IP}:5060 [Mar 1 12:36:08] DEBUG[8290] cdr.c: Dropping CDR ! [Mar 1 12:36:08] VERBOSE[8290] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0) [Mar 1 12:36:08] DEBUG[8290] rtp.c: Channel '' has no RTP, not doing anything [Mar 1 12:36:08] DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=CONGESTION. [Mar 1 12:36:08] DEBUG[8290] pbx.c: Launching 'Hangup' [Mar 1 12:36:08] VERBOSE[8290] pbx.c: -- Executing [123@common:2] Hangup("SIP/dev-sip.tele500.com-08206af0", "") in new stack [Mar 1 12:36:08] DEBUG[8290] pbx.c: Spawn extension (common,123,2) exited non-zero on 'SIP/dev-sip.tele500.com-08206af0' [Mar 1 12:36:08] VERBOSE[8290] pbx.c: == Spawn extension (common, 123, 2) exited non-zero on 'SIP/dev-sip.tele500.com-08206af0' [Mar 1 12:36:08] DEBUG[8290] channel.c: Soft-Hanging up channel 'SIP/dev-sip.tele500.com-08206af0' [Mar 1 12:36:08] DEBUG[8290] channel.c: Hanging up channel 'SIP/dev-sip.tele500.com-08206af0' [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Hangup call SIP/dev-sip.tele500.com-08206af0, SIP callid btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Hanging up channel in state Ring (not UP) [Mar 1 12:36:08] VERBOSE[8290] chan_sip.c: Scheduling destruction of SIP dialog 'btomwmxyvjserjv@192.168.7.55' in 32000 ms (Method: INVITE) [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Session timer stopped: -1 - btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:08] VERBOSE[8290] chan_sip.c: <--- Reliably Transmitting (no NAT) to {OpenSER_IP}:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP {OpenSER_IP};branch=z9hG4bK18a8.ca91afd.0;received={OpenSER_IP} Via: SIP/2.0/UDP {UA_IP}:5061;received={UA_IP};rport=5061;branch=z9hG4bKdjvhchet From: "10000" ;tag=qjkyu To: ;tag=as5cb187d2 Call-ID: btomwmxyvjserjv@192.168.7.55 CSeq: 708 INVITE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 0 [ 27]: SIP/2.0 408 Request Timeout [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 1 [ 80]: Via: SIP/2.0/UDP {OpenSER_IP};branch=z9hG4bK18a8.ca91afd.0;received={OpenSER_IP} [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 2 [ 90]: Via: SIP/2.0/UDP {UA_IP}:5061;received={UA_IP};rport=5061;branch=z9hG4bKdjvhchet [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 3 [ 55]: From: "10000" ;tag=qjkyu [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 4 [ 48]: To: ;tag=as5cb187d2 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 5 [ 37]: Call-ID: btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 6 [ 16]: CSeq: 708 INVITE [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 7 [ 18]: Server: Media GW 1 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 8 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Header 11 [ 0]: [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #20 [Mar 1 12:36:08] DEBUG[8290] chan_sip.c: Trying to put 'SIP/2.0 40' onto UDP socket destined for {OpenSER_IP}:5060 [Mar 1 12:36:08] DEBUG[8290] cdr.c: Dropping CDR ! [Mar 1 12:36:08] DEBUG[8266] devicestate.c: No provider found, checking channel drivers for SIP - Magrathea [Mar 1 12:36:08] DEBUG[8266] chan_sip.c: Checking device state for peer Magrathea [Mar 1 12:36:08] DEBUG[8266] devicestate.c: Changing state for SIP/Magrathea - state 1 (Not in use) [Mar 1 12:36:08] DEBUG[8266] devicestate.c: device 'SIP/Magrathea' state '1' [Mar 1 12:36:08] DEBUG[8266] devicestate.c: No provider found, checking channel drivers for SIP - dev-sip.tele500.com [Mar 1 12:36:08] DEBUG[8266] chan_sip.c: Checking device state for peer dev-sip.tele500.com [Mar 1 12:36:08] DEBUG[8266] devicestate.c: Changing state for SIP/dev-sip.tele500.com - state 1 (Not in use) [Mar 1 12:36:08] DEBUG[8266] devicestate.c: device 'SIP/dev-sip.tele500.com' state '1' [Mar 1 12:36:08] DEBUG[8265] devicestate.c: Processing device state change for 'SIP/Magrathea' [Mar 1 12:36:08] DEBUG[8265] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/Magrathea' [Mar 1 12:36:08] DEBUG[8265] devicestate.c: Aggregate devstate result is 1 [Mar 1 12:36:08] DEBUG[8265] devicestate.c: Aggregate state for device 'SIP/Magrathea' has not changed from 'Not in use' [Mar 1 12:36:08] DEBUG[8265] devicestate.c: Processing device state change for 'SIP/dev-sip.tele500.com' [Mar 1 12:36:08] DEBUG[8265] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/dev-sip.tele500.com' [Mar 1 12:36:08] DEBUG[8265] devicestate.c: Aggregate devstate result is 1 [Mar 1 12:36:08] DEBUG[8265] devicestate.c: Aggregate state for device 'SIP/dev-sip.tele500.com' has changed to 'Not in use' [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: <--- SIP read from UDP://{OpenSER_IP}:5060 ---> ACK sip:123@dev-sip.tele500.com SIP/2.0 Via: SIP/2.0/UDP {OpenSER_IP};branch=z9hG4bK18a8.ca91afd.0 From: "10000" ;tag=qjkyu Call-ID: btomwmxyvjserjv@192.168.7.55 To: ;tag=as5cb187d2 CSeq: 708 ACK Max-Forwards: 70 User-Agent: SIP Proxy 1 Content-Length: 0 <-------------> [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 0 [ 39]: ACK sip:123@dev-sip.tele500.com SIP/2.0 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP {OpenSER_IP};branch=z9hG4bK18a8.ca91afd.0 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 2 [ 55]: From: "10000" ;tag=qjkyu [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 3 [ 37]: Call-ID: btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 4 [ 48]: To: ;tag=as5cb187d2 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 5 [ 13]: CSeq: 708 ACK [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 7 [ 23]: User-Agent: SIP Proxy 1 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 9 [ 0]: [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: --- (9 headers 0 lines) --- [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #20 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Stopping retransmission on 'btomwmxyvjserjv@192.168.7.55' of Response 708: Match Found [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: <--- SIP read from UDP://{OpenSER_IP}:5060 ---> SIP/2.0 404 Not here Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK04892ce3;rport=5060 From: "10000" ;tag=as25e845f5 To: ;tag=78b11732fd1718be878e22d27662fcac-6cbc Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} CSeq: 103 BYE Server: SIP Proxy 1 Content-Length: 0 <-------------> [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 0 [ 20]: SIP/2.0 404 Not here [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP {Asterisk_IP}:5060;branch=z9hG4bK04892ce3;rport=5060 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 2 [ 53]: From: "10000" ;tag=as25e845f5 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 3 [ 80]: To: ;tag=78b11732fd1718be878e22d27662fcac-6cbc [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 4 [ 54]: Call-ID: 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 6 [ 19]: Server: SIP Proxy 1 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Header 8 [ 0]: [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: --- (8 headers 0 lines) --- [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Stopping retransmission on '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}' of Request 103: Match Found [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: Destroying SIP dialog 58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP} [Mar 1 12:36:08] VERBOSE[8269] chan_sip.c: Really destroying SIP dialog '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}' Method: INVITE [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: ---------- SIP HISTORY for '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}' [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: * SIP Call [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 001. OBproxy Using global obproxy {OpenSER_IP} [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 002. NewChan Channel SIP/Magrathea-0820cf20 - from 58730b377ad8e3324f7d07501 [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 003. TxReqRel INVITE / 102 INVITE - INVITE [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 100 Giving a try [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 005. Rx SIP/2.0 / 102 INVITE / 408 Request Timeout [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 006. TxReq ACK / 102 ACK - ACK [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 007. SchedDestroy 32000 ms [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 008. Hangup Cause No user responding [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 009. CancelDestroy [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 010. SchedDestroy 32000 ms [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 011. TxReqRel BYE / 103 BYE - BYE [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 012. RTCPaudio Quality: [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: 013. Rx SIP/2.0 / 103 BYE / 404 Not here [Mar 1 12:36:08] DEBUG[8269] chan_sip.c: ---------- END SIP HISTORY for '58730b377ad8e3324f7d07501c1b3717@{Asterisk_IP}' [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: Auto destroying SIP dialog 'btomwmxyvjserjv@192.168.7.55' [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: Destroying SIP dialog btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:40] VERBOSE[8269] chan_sip.c: Really destroying SIP dialog 'btomwmxyvjserjv@192.168.7.55' Method: ACK [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: ---------- SIP HISTORY for 'btomwmxyvjserjv@192.168.7.55' [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: * SIP Call [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: 001. Rx INVITE / 708 INVITE / sip:123@dev-sip.tele500.com [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: 002. NewChan Channel SIP/dev-sip.tele500.com-08206af0 - from btomwmxyvjserjv [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: 003. TxResp SIP/2.0 / 708 INVITE - 100 Trying [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: 004. Cancel Cause No user responding [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: 005. SchedDestroy 32000 ms [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: 006. TxRespRel SIP/2.0 / 708 INVITE - 408 Request Timeout [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: 007. Rx ACK / 708 ACK / sip:123@dev-sip.tele500.com [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: 008. AutoDestroy btomwmxyvjserjv@192.168.7.55 [Mar 1 12:36:40] DEBUG[8269] chan_sip.c: ---------- END SIP HISTORY for 'btomwmxyvjserjv@192.168.7.55'