asterisk-test*CLI> Really destroying SIP dialog '2114317149' Method: OPTIONS Really destroying SIP dialog '924064418' Method: ACK <--- SIP read from UDP:192.168.100.6:5060 ---> OPTIONS sip:211@192.168.100.29 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.6:5060;rport;branch=z9hG4bK1856643648 From: "clau" ;tag=1928631702 To: Call-ID: 193245645 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 70 User-Agent: Linphone/3.3.99.8 (eXosip2/3.3.0) Expires: 120 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Looking for 211 in default (domain 192.168.100.29) <--- Transmitting (no NAT) to 192.168.100.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.6:5060;rport;branch=z9hG4bK1856643648;received=192.168.100.6 From: "clau" ;tag=1928631702 To: ;tag=as7e237e61 Call-ID: 193245645 CSeq: 20 OPTIONS Server: Asterisk PBX SVN-trunk-r298686 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '193245645' in 32000 ms (Method: OPTIONS) <--- SIP read from UDP:192.168.100.6:5060 ---> INVITE sip:211@192.168.100.29 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.6:5060;rport;branch=z9hG4bK1286411818 From: "clau" ;tag=764665301 To: Call-ID: 993614872 CSeq: 20 INVITE Contact: "clau" Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.3.99.8 (eXosip2/3.3.0) Subject: Phone call Content-Length: 305 v=0 o=clau 123456 654321 IN IP4 192.168.100.6 s=A conversation c=IN IP4 192.168.100.6 b=AS:512 t=0 0 m=audio 7078 RTP/AVP 0 101 b=AS:80 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv m=video 9078 RTP/AVP 34 b=AS:422 a=rtpmap:34 H263/90000 a=sendrecv <-------------> --- (13 headers 16 lines) --- Sending to 192.168.100.6:5060 (no NAT) Using INVITE request as basis request - 993614872 No matching peer for 'clau' from '192.168.100.6:5060' == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found video description format H263 for ID 34 Capabilities: us - 0x180004 (ulaw|h263|h263p), peer - audio=0x4 (ulaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80004 (ulaw|h263) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.100.6:7078 Peer video RTP is at port 192.168.100.6:9078 Looking for 211 in default (domain 192.168.100.29) list_route: hop: <--- Transmitting (no NAT) to 192.168.100.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.6:5060;branch=z9hG4bK1286411818;received=192.168.100.6;rport=5060 From: "clau" ;tag=764665301 To: Call-ID: 993614872 CSeq: 20 INVITE Server: Asterisk PBX SVN-trunk-r298686 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [211@default:1] AGI("SIP/192.168.100.6-00000005", "agi://192.168.100.6") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_request: agi://192.168.100.6 AGI Tx >> agi_channel: SIP/192.168.100.6-00000005 AGI Tx >> agi_language: en AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1292554503.5 AGI Tx >> agi_version: SVN-trunk-r298686 AGI Tx >> agi_callerid: clau AGI Tx >> agi_calleridname: clau AGI Tx >> agi_callingpres: 0 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: 211 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: default AGI Tx >> agi_extension: 211 AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> agi_threadid: -1221358736 AGI Tx >> AGI Rx << ANSWER Audio is at 5060 Video is at 192.168.100.29:5060 Adding codec 0x4 (ulaw) to SDP Adding video codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.100.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.6:5060;branch=z9hG4bK1286411818;received=192.168.100.6;rport=5060 From: "clau" ;tag=764665301 To: ;tag=as6b8f2fe2 Call-ID: 993614872 CSeq: 20 INVITE Server: Asterisk PBX SVN-trunk-r298686 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 349 v=0 o=root 1764185867 1764185867 IN IP4 192.168.100.29 s=Asterisk PBX SVN-trunk-r298686 c=IN IP4 192.168.100.29 b=CT:384 t=0 0 m=audio 15652 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 14194 RTP/AVP 34 a=rtpmap:34 H263/90000 a=sendrecv <------------> <--- SIP read from UDP:192.168.100.6:5060 ---> ACK sip:211@192.168.100.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.6:5060;rport;branch=z9hG4bK2138976552 From: "clau" ;tag=764665301 To: ;tag=as6b8f2fe2 Call-ID: 993614872 CSeq: 20 ACK Contact: "clau" Max-Forwards: 70 User-Agent: Linphone/3.3.99.8 (eXosip2/3.3.0) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- AGI Tx >> 200 result=0 AGI Rx << ANSWER AGI Tx >> 200 result=0 AGI Rx << EXEC Wait "5" -- AGI Script Executing Application: (Wait) Options: (5) AGI Tx >> 200 result=0 AGI Rx << EXEC Wait "5" -- AGI Script Executing Application: (Wait) Options: (5) AGI Tx >> 200 result=0 AGI Rx << EXEC Wait "5" -- AGI Script Executing Application: (Wait) Options: (5) AGI Tx >> 200 result=0 AGI Rx << EXEC Wait "5" -- AGI Script Executing Application: (Wait) Options: (5) <--- SIP read from UDP:192.168.100.6:5060 ---> BYE sip:211@192.168.100.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.6:5060;rport;branch=z9hG4bK294575461 From: "clau" ;tag=764665301 To: ;tag=as6b8f2fe2 Call-ID: 993614872 CSeq: 21 BYE Contact: Max-Forwards: 70 User-Agent: Linphone/3.3.99.8 (eXosip2/3.3.0) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.100.6:5060 (no NAT) Scheduling destruction of SIP dialog '993614872' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.100.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.6:5060;branch=z9hG4bK294575461;received=192.168.100.6;rport=5060 From: "clau" ;tag=764665301 To: ;tag=as6b8f2fe2 Call-ID: 993614872 CSeq: 21 BYE Server: Asterisk PBX SVN-trunk-r298686 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> AGI Tx >> 200 result=-1 AGI Rx << EXEC Wait "5" -- AGI Script Executing Application: (Wait) Options: (5) Really destroying SIP dialog '193245645' Method: OPTIONS AGI Tx >> 200 result=0 AGI Rx << EXEC Wait "5" -- AGI Script Executing Application: (Wait) Options: (5) AGI Tx >> 200 result=0 AGI Rx << EXEC Wait "5" -- AGI Script Executing Application: (Wait) Options: (5) AGI Tx >> 200 result=0 <<<< HERE I KILL THE AHN APP AND RIGHT AFTER THAT ASTERISK HANGS UP THE CALL >>>>> -- AGI Script agi://192.168.100.6 completed, returning 0 -- Executing [h@default:1] AGI("SIP/192.168.100.6-00000005", "agi://192.168.100.6") in new stack AGI Tx >> agi_network: yes [Dec 16 21:55:47] ERROR[27914]: utils.c:1138 ast_carefulwrite: write() returned error: Connection refused [Dec 16 21:55:47] WARNING[27914]: res_agi.c:1411 launch_netscript: Connect to 'agi://192.168.100.6' failed: Connection refused -- Executing [h@default:2] Hangup("SIP/192.168.100.6-00000005", "") in new stack == Spawn extension (default, h, 2) exited non-zero on 'SIP/192.168.100.6-00000005' asterisk-test*CLI>