########### CALL START <--- SIP read from UDP:208.72.186.166:5060 ---> INVITE sip:9802336995@208.72.186.184 SIP/2.0 Via: SIP/2.0/UDP 208.72.186.166;rport;branch=z9hG4bKF7FF4m3NDDFvr Max-Forwards: 69 From: "6468271109" ;tag=aeD6rcKD45X3S To: Call-ID: 990e1414-78eb-4c6b-a2be-b9fc2691291f CSeq: 9424033 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.5pre9-hacked Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 267 X-FS-Support: update_display Remote-Party-ID: "6468271109" ;party=calling;screen=yes;privacy=off v=0 o=Sonus_UAC 27238 16778 IN IP4 209.249.3.74 s=SIP Media Capabilities c=IN IP4 209.249.3.80 t=0 0 m=audio 7042 RTP/AVP 0 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 a=ptime:20 <-------------> --- (17 headers 12 lines) --- == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Sending to 208.72.186.166 : 5060 (no NAT) Using INVITE request as basis request - 990e1414-78eb-4c6b-a2be-b9fc2691291f Found peer 'ivrin' for '6468271109' from 208.72.186.166:5060 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 127 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 127 Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.249.3.80:7042 Looking for 9802336995 in ivrin (domain 208.72.186.184) list_route: hop: <--- Transmitting (no NAT) to 208.72.186.166:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 208.72.186.166;branch=z9hG4bKF7FF4m3NDDFvr;received=208.72.186.166;rport=5060 From: "6468271109" ;tag=aeD6rcKD45X3S To: Call-ID: 990e1414-78eb-4c6b-a2be-b9fc2691291f CSeq: 9424033 INVITE Server: StanaCard IVR Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [9802336995@ivrin:1] Set("SIP/ivrin-00000001", "callId=990e1414-78eb-4c6b-a2be-b9fc2691291f") in new stack -- Executing [9802336995@ivrin:2] SIPAddHeader("SIP/ivrin-00000001", "Cisco-Guid: 990e1414-78eb-4c6b-a2be-b9fc2691291f") in new stack -- Executing [9802336995@ivrin:3] AGI("SIP/ivrin-00000001", "agi://10.5.60.115:2222?lcr") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_request: agi://10.5.60.115:2222?lcr AGI Tx >> agi_channel: SIP/ivrin-00000001 AGI Tx >> agi_language: en AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1299520194.1 AGI Tx >> agi_version: 1.6.2.17 AGI Tx >> agi_callerid: 6468271109 AGI Tx >> agi_calleridname: 6468271109 AGI Tx >> agi_callingpres: 0 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: 9802336995 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: ivrin AGI Tx >> agi_extension: 9802336995 AGI Tx >> agi_priority: 3 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> agi_threadid: 1090324800 AGI Tx >> AGI Rx << EXEC Set LIMIT_TIMEOUT_FILE=sc-timeout -- AGI Script Executing Application: (Set) Options: (LIMIT_TIMEOUT_FILE=sc-timeout) AGI Tx >> 200 result=0 AGI Rx << EXEC Set LIMIT_TIMEOUT_FILE=sc-timeout -- AGI Script Executing Application: (Set) Options: (LIMIT_TIMEOUT_FILE=sc-timeout) AGI Tx >> 200 result=0 AGI Rx << EXEC Set TIMEOUT(digit)=8 -- AGI Script Executing Application: (Set) Options: (TIMEOUT(digit)=8) -- Digit timeout set to 8.000 AGI Tx >> 200 result=0 AGI Rx << EXEC Background silence/0 -- AGI Script Executing Application: (Background) Options: (silence/0) Audio is at 208.72.186.184 port 19620 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 208.72.186.166:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.72.186.166;branch=z9hG4bKF7FF4m3NDDFvr;received=208.72.186.166;rport=5060 From: "6468271109" ;tag=aeD6rcKD45X3S To: ;tag=as3bb199ed Call-ID: 990e1414-78eb-4c6b-a2be-b9fc2691291f CSeq: 9424033 INVITE Server: StanaCard IVR Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 268 v=0 o=root 1182966147 1182966147 IN IP4 208.72.186.184 s=Asterisk PBX 1.6.2.17 c=IN IP4 208.72.186.184 t=0 0 m=audio 19620 RTP/AVP 0 127 a=rtpmap:0 PCMU/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:208.72.186.166:5060 ---> ACK sip:9802336995@208.72.186.184 SIP/2.0 Via: SIP/2.0/UDP 208.72.186.166;rport;branch=z9hG4bKgg975FmSap5em Max-Forwards: 70 From: "6468271109" ;tag=aeD6rcKD45X3S To: ;tag=as3bb199ed Call-ID: 990e1414-78eb-4c6b-a2be-b9fc2691291f CSeq: 9424033 ACK Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- Playing 'silence/0.ulaw' (language 'en') AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE SIP_HEADER(Contact) AGI Tx >> 200 result=1 () AGI Rx << GET VARIABLE SIP_HEADER(X-Src-IP) AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE SIP_HEADER(Call-ID) AGI Tx >> 200 result=1 (990e1414-78eb-4c6b-a2be-b9fc2691291f) AGI Rx << Exec Read pin,ss-pin,14,,, -- AGI Script Executing Application: (Read) Options: (pin,ss-pin,14,,,) -- Accepting a maximum of 14 digits. -- Playing 'ss-pin.ulaw' (language 'en') ########### HERE I HANGED UP <--- SIP read from UDP:208.72.186.166:5060 ---> BYE sip:9802336995@208.72.186.184 SIP/2.0 Via: SIP/2.0/UDP 208.72.186.166;rport;branch=z9hG4bK7Ze2eyQa825ve Max-Forwards: 70 From: "6468271109" ;tag=aeD6rcKD45X3S To: ;tag=as3bb199ed Call-ID: 990e1414-78eb-4c6b-a2be-b9fc2691291f CSeq: 9424034 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.5pre9-hacked Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 208.72.186.166 : 5060 (no NAT) <--- Transmitting (no NAT) to 208.72.186.166:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.72.186.166;branch=z9hG4bK7Ze2eyQa825ve;received=208.72.186.166;rport=5060 From: "6468271109" ;tag=aeD6rcKD45X3S To: ;tag=as3bb199ed Call-ID: 990e1414-78eb-4c6b-a2be-b9fc2691291f CSeq: 9424034 BYE Server: StanaCard IVR Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- User disconnected AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE pin AGI Tx >> 200 result=1 () AGI Rx << Exec Playback ss-pininvalid -- AGI Script Executing Application: (Playback) Options: (ss-pininvalid) [Mar 7 12:49:57] WARNING[23939]: file.c:753 ast_readaudio_callback: Failed to write frame -- Playing 'ss-pininvalid.ulaw' (language 'en') [Mar 7 12:49:57] WARNING[23939]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/ivrin-00000001 for ss-pininvalid AGI Tx >> 200 result=0 AGI Rx << Exec Read pin,ss-pin,14,,, -- AGI Script Executing Application: (Read) Options: (pin,ss-pin,14,,,) -- Accepting a maximum of 14 digits. [Mar 7 12:49:57] WARNING[23939]: file.c:753 ast_readaudio_callback: Failed to write frame -- Playing 'ss-pin.ulaw' (language 'en') -- User disconnected AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE pin AGI Tx >> 200 result=1 () AGI Rx << Exec Playback ss-pininvalid -- AGI Script Executing Application: (Playback) Options: (ss-pininvalid) [Mar 7 12:49:57] WARNING[23939]: file.c:753 ast_readaudio_callback: Failed to write frame -- Playing 'ss-pininvalid.ulaw' (language 'en') [Mar 7 12:49:57] WARNING[23939]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/ivrin-00000001 for ss-pininvalid AGI Tx >> 200 result=0 AGI Rx << Exec Read pin,ss-pin,14,,, -- AGI Script Executing Application: (Read) Options: (pin,ss-pin,14,,,) -- Accepting a maximum of 14 digits. [Mar 7 12:49:57] WARNING[23939]: file.c:753 ast_readaudio_callback: Failed to write frame -- Playing 'ss-pin.ulaw' (language 'en') -- User disconnected AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE pin AGI Tx >> 200 result=1 () AGI Rx << Exec Playback ss-pininvalid -- AGI Script Executing Application: (Playback) Options: (ss-pininvalid) [Mar 7 12:49:57] WARNING[23939]: file.c:753 ast_readaudio_callback: Failed to write frame -- Playing 'ss-pininvalid.ulaw' (language 'en') [Mar 7 12:49:57] WARNING[23939]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/ivrin-00000001 for ss-pininvalid AGI Tx >> 200 result=0 AGI Rx << Exec Playback ss-goodbye -- AGI Script Executing Application: (Playback) Options: (ss-goodbye) [Mar 7 12:49:57] WARNING[23939]: file.c:753 ast_readaudio_callback: Failed to write frame -- Playing 'ss-goodbye.ulaw' (language 'en') [Mar 7 12:49:57] WARNING[23939]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/ivrin-00000001 for ss-goodbye AGI Tx >> 200 result=0 AGI Rx << Exec Playtones busy -- AGI Script Executing Application: (Playtones) Options: (busy) AGI Tx >> 200 result=0 AGI Rx << Exec Wait 5 -- AGI Script Executing Application: (Wait) Options: (5) AGI Tx >> 200 result=-1 ########### GOT HANGUP after mutliple Playbacks AGI Rx << HANGUP AGI Tx >> 200 result=1 -- AGI Script agi://10.5.60.115:2222?lcr completed, returning 0 Really destroying SIP dialog '990e1414-78eb-4c6b-a2be-b9fc2691291f' Method: BYE