Asterisk 1.6.2.3-rc2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.3-rc2 currently running on sip3 (pid = 8156) sip3*CLI> Verbosity was 0 and is now 45 <--- SIP read from UDP:219.90.198.19:5060 ---> <-------------> == Manager 'admin' logged on from 127.0.0.1 -- Executing [0754474090@dialler-originate:1] Set("Local/0754474090@dialler-originate-6d36;2", "__CAMPAIGNID=1") in new stack -- Executing [0754474090@dialler-originate:2] Set("Local/0754474090@dialler-originate-6d36;2", "ORGPHONE=0754474090") in new stack -- Executing [0754474090@dialler-originate:3] Set("Local/0754474090@dialler-originate-6d36;2", "PHONE=0754474090") in new stack -- Executing [0754474090@dialler-originate:4] Set("Local/0754474090@dialler-originate-6d36;2", "PHONE=61754474090") in new stack -- Executing [0754474090@dialler-originate:5] Goto("Local/0754474090@dialler-originate-6d36;2", "61754474090,4") in new stack -- Goto (dialler-originate,61754474090,4) -- Executing [61754474090@dialler-originate:4] Set("Local/0754474090@dialler-originate-6d36;2", "DNCCALLDATE=2010-02-12 14:46:02") in new stack -- Executing [61754474090@dialler-originate:5] Set("Local/0754474090@dialler-originate-6d36;2", "DNCCALLDATESHORT=2010-02-12") in new stack -- Executing [61754474090@dialler-originate:6] Set("Local/0754474090@dialler-originate-6d36;2", "GROUP()=1") in new stack -- Executing [61754474090@dialler-originate:7] GotoIf("Local/0754474090@dialler-originate-6d36;2", "0?103") in new stack -- Executing [61754474090@dialler-originate:8] Goto("Local/0754474090@dialler-originate-6d36;2", "check,1") in new stack -- Goto (dialler-originate,check,1) -- Executing [check@dialler-originate:1] GotoIf("Local/0754474090@dialler-originate-6d36;2", "?2:3") in new stack -- Goto (dialler-originate,check,3) -- Executing [check@dialler-originate:3] GotoIf("Local/0754474090@dialler-originate-6d36;2", "?7") in new stack -- Executing [check@dialler-originate:4] AGI("Local/0754474090@dialler-originate-6d36;2", "dnc-check.agi.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dnc-check.agi.php == Manager 'admin' logged off from 127.0.0.1 -- AGI Script dnc-check.agi.php completed, returning 0 -- Executing [check@dialler-originate:5] GotoIf("Local/0754474090@dialler-originate-6d36;2", "?61754474090,103") in new stack -- Executing [check@dialler-originate:6] GotoIf("Local/0754474090@dialler-originate-6d36;2", "1?record,1") in new stack -- Goto (dialler-originate,record,1) -- Executing [record@dialler-originate:1] GotoIf("Local/0754474090@dialler-originate-6d36;2", "?2:4") in new stack -- Goto (dialler-originate,record,4) -- Executing [record@dialler-originate:4] NoOp("Local/0754474090@dialler-originate-6d36;2", "skipping call outcome because we moved it") in new stack -- Executing [record@dialler-originate:5] Goto("Local/0754474090@dialler-originate-6d36;2", "61754474090,50") in new stack -- Goto (dialler-originate,61754474090,50) -- Executing [61754474090@dialler-originate:50] AGI("Local/0754474090@dialler-originate-6d36;2", "dial-start.agi.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dial-start.agi.php -- AGI Script dial-start.agi.php completed, returning 0 -- Executing [61754474090@dialler-originate:51] GotoIf("Local/0754474090@dialler-originate-6d36;2", "?check,4") in new stack -- Executing [61754474090@dialler-originate:52] NoOp("Local/0754474090@dialler-originate-6d36;2", "") in new stack -- Executing [61754474090@dialler-originate:53] Originate("Local/0754474090@dialler-originate-6d36;2", "SIP/trunk-16/0754474090,exten,dialler-answered-voicecast,61754474090,1") in new stack == Using SIP RTP CoS mark 5 Audio is at 67.228.218.209 port 11242 Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 202.85.243.87:5060: INVITE sip:0754474090@sip.pennytel.com:5060 SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK71bad523;rport Max-Forwards: 70 From: "asterisk" ;tag=as71c6cbb1 T o: Contact: Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.3-rc2 Date: Fri, 12 Feb 2010 04:16:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 318 v=0 o=root 1532157213 1532157213 IN IP4 67.228.218.209 s=Asterisk PBX 1.6.2.3-rc2 c=IN IP4 67.228.218.209 t=0 0 m=audio 11242 RTP/AVP 3 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:202.85.243.87:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK71bad523;rport=5060 From: "asterisk" ;tag=as71c6cbb1 To: Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 102 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:202.85.243.87:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK71bad523;rport=5060 Record-Route: From: asterisk ;tag=as71c6cbb1 To: Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 102 INVITE Server: Sippy WWW-Authenticate: Digest realm="sip.pennytel.com",nonce="a5e1c9386f2f86af76ab553ebac9b57dac2a" <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 202.85.243.87:5060: ACK sip:0754474090@sip.pennytel.com:5060 SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK71bad523;rport Max-Forwards: 70 From: "asterisk" ;tag=as71c6cbb1 To: Contact: Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.3-rc2 Content-Length: 0 --- Audio is at 67.228.218.209 port 11242 Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 202.85.243.87:5060: INVITE sip:0754474090@sip.pennytel.com:5060 SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK77ce8021;rport Max-Forwards: 70 From: "asterisk" ;tag=as71c6cbb1 To: Contact: Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.3-rc2 Authorization: Digest username="8889209288", realm="sip.pennytel.com", algorithm=MD5, uri="sip:0754474090@sip.pennytel.com:5060", nonce="a5e1c9386f2f86af76ab553ebac9b57dac2a", response="c6884a9af0dee0d7bf1f4c2a5f814b5c" Date: Fri, 12 Feb 2010 04:16:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 318 v=0 o=root 1532157213 1532157214 IN IP4 67.228.218.209 s=Asterisk PBX 1.6.2.3-rc2 c=IN IP4 67.228.218.209 t=0 0 m=audio 11242 RTP/AVP 3 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:202.85.243.87:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK77ce8021;rport=5060 From: "asterisk" ;tag=as71c6cbb1 To: Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 103 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '2466462266eba932403334107cd67c91@67.228.218.209' Method: REGISTER Really destroying SIP dialog '36a40f261bff24a15fdc32cd3e824198@67.228.218.209' Method: REGISTER Really destroying SIP dialog '624fa77256cc3dc111819fde6484cec7@67.228.218.209' Method: REGISTER Really destroying SIP dialog '2e8257014ec7668176eed89d63c7dabc@67.228.218.209' Method: REGISTER Really destroying SIP dialog '4239e6ec789bdb8d6ff892516506b48d@67.228.218.209' Method: REGISTER -- Remote UNIX connection -- Remote UNIX connection disconnected Really destroying SIP dialog '666493653b906f5a4d0d75a36ec43c62@67.228.218.209' Method: REGISTER <--- SIP read from UDP:202.85.243.87:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK77ce8021;rport=5060 Record-Route: From: asterisk ;tag=as71c6cbb1 To: ;tag=103b6f792d43a1d933f7550c3b96cce0 Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 103 INVITE Server: Sippy Content-Length: 194 Content-Type: application/sdp v=0 o=Sippy 187117196 1 IN IP4 202.85.243.87 s=SIP Call t=0 0 m=audio 19652 RTP/AVP 8 101 c=IN IP4 202.85.241.100 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (10 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 202.85.241.100:19652 [Feb 12 14:46:13] NOTICE[9361]: rtp.c:1130 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 202.85.241.100 == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 <--- SIP read from UDP:219.90.198.19:5060 ---> <-------------> <--- SIP read from UDP:202.85.243.87:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK77ce8021;rport=5060 Record-Route: From: asterisk ;tag=as71c6cbb1 To: ;tag=103b6f792d43a1d933f7550c3b96cce0 Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 103 INVITE Server: Sippy Contact: Anonymous Content-Length: 194 Content-Type: application/sdp v=0 o=Sippy 187117196 2 IN IP4 202.85.243.87 s=SIP Call t=0 0 m=audio 19652 RTP/AVP 8 101 c=IN IP4 202.85.241.100 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 202.85.241.100:19652 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 202.85.243.87, port 5060 Transmitting (no NAT) to 202.85.243.87:5060: ACK sip:202.85.243.87:5061 SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK14792bba;rport Route: Max-Forwards: 70 From: "asterisk" ;tag=as71c6cbb1 To: ;tag=103b6f792d43a1d933f7550c3b96cce0 Contact: Call-ID: 64dee7ed17c94b1e34ff2ae622bc2b37@sip.pennytel.com CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.2.3-rc2 Content-Length: 0 --- -- Executing [61754474090@dialler-answered-voicecast:1] Set("SIP/trunk-16-00000000", "PHONE=61754474090") in new stack -- Executing [61754474090@dialler-answered-voicecast:2] BackGround("SIP/trunk-16-00000000", "VCast-Excite-Mobile-Sexy-Part1") in new stack -- Playing 'VCast-Excite-Mobile-Sexy-Part1.slin' (language 'en') -- Executing [61754474090@dialler-originate:54] AGI("Local/0754474090@dialler-originate-6d36;2", "dial-stop.agi.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dial-stop.agi.php == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- AGI Script dial-stop.agi.php completed, returning 0 -- Executing [61754474090@dialler-originate:55] Goto("Local/0754474090@dialler-originate-6d36;2", "61754474090,103") in new stack -- Goto (dialler-originate,61754474090,103) -- Executing [61754474090@dialler-originate:103] Hangup("Local/0754474090@dialler-originate-6d36;2", "") in new stack == Spawn extension (dialler-originate, 61754474090, 103) exited non-zero on 'Local/0754474090@dialler-originate-6d36;2' Reliably Transmitting (no NAT) to 202.122.22.2:5060: OPTIONS sip:202.122.22.2 SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK35ac56e7;rport Max-Forwards: 70 From: "asterisk" ;tag=as71e7a212 To: Contact: Call-ID: 0d41f575526564567e595e80315e86ff@67.228.218.209 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.3-rc2 Date: Fri, 12 Feb 2010 04:16:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:202.122.22.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK35ac56e7;rport From: "asterisk" ;tag=as71e7a212 Call-ID: 0d41f575526564567e595e80315e86ff@67.228.218.209 CSeq: 102 OPTIONS To: ;tag=abcdefg Accept: application/sdp Record-Route: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '0d41f575526564567e595e80315e86ff@67.228.218.209' Method: OPTIONS Reliably Transmitting (no NAT) to 202.169.178.10:5060: OPTIONS sip:sip.gotalk.com SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK1624bdcf;rport Max-Forwards: 70 From: "asterisk" ;tag=as78a18117 To: Contact: Call-ID: 0a71d301158e70963f178473546754c5@67.228.218.209 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.3-rc2 Date: Fri, 12 Feb 2010 04:16:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 202.169.178.10:5060: OPTIONS sip:sip.gotalk.com SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK7fdc7dd1;rport Max-Forwards: 70 From: "asterisk" ;tag=as60bd54e9 To: Contact: Call-ID: 2a5f7767160293b12f35eeea5df5dcfb@67.228.218.209 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.3-rc2 Date: Fri, 12 Feb 2010 04:16:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 202.169.178.10:5060: OPTIONS sip:sip.gotalk.com SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK149ddfbc;rport Max-Forwards: 70 From: "asterisk" ;tag=as188f624d To: Contact: Call-ID: 1926eb9873d51b397acfab213e0d11af@67.228.218.209 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.3-rc2 Date: Fri, 12 Feb 2010 04:16:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:202.169.178.10:5060 ---> SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK1624bdcf;rport To: From: "asterisk";tag=as78a18117 Call-ID: 0a71d301158e70963f178473546754c5@67.228.218.209 CSeq: 102 OPTIONS User-Agent: ENSR2.5.46.8-IS1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '0a71d301158e70963f178473546754c5@67.228.218.209' Method: OPTIONS <--- SIP read from UDP:202.169.178.10:5060 ---> SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK7fdc7dd1;rport To: From: "asterisk";tag=as60bd54e9 Call-ID: 2a5f7767160293b12f35eeea5df5dcfb@67.228.218.209 CSeq: 102 OPTIONS User-Agent: ENSR2.5.46.8-IS1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '2a5f7767160293b12f35eeea5df5dcfb@67.228.218.209' Method: OPTIONS <--- SIP read from UDP:202.169.178.10:5060 ---> SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK149ddfbc;rport To: From: "asterisk";tag=as188f624d Call-ID: 1926eb9873d51b397acfab213e0d11af@67.228.218.209 CSeq: 102 OPTIONS User-Agent: ENSR2.5.46.8-IS1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '1926eb9873d51b397acfab213e0d11af@67.228.218.209' Method: OPTIONS Reliably Transmitting (no NAT) to 202.169.178.10:5060: OPTIONS sip:sip.gotalk.com SIP/2.0 Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK6c2d8c2b;rport Max-Forwards: 70 From: "asterisk" ;tag=as539bb1bf To: Contact: Call-ID: 66e6a5ed618cac603b5f76954b410ee1@67.228.218.209 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.3-rc2 Date: Fri, 12 Feb 2010 04:16:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:202.169.178.10:5060 ---> SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 67.228.218.209:5060;branch=z9hG4bK6c2d8c2b;rport To: From: "asterisk";tag=as539bb1bf Call-ID: 66e6a5ed618cac603b5f76954b410ee1@67.228.218.209 CSeq: 102 OPTIONS User-Agent: ENSR2.5.46.8-IS1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '66e6a5ed618cac603b5f76954b410ee1@67.228.218.209' Method: OPTIONS <--- SIP read from UDP:219.90.198.19:5060 ---> <-------------> -- Executing [61754474090@dialler-answered-voicecast:3] WaitExten("SIP/trunk-16-00000000", "2") in new stack -- Timeout on SIP/trunk-16-00000000, continuing... -- Executing [61754474090@dialler-answered-voicecast:4] BackGround("SIP/trunk-16-00000000", "VCast-Answer-Machine-Excite-Mobile-Sexy-bit 1") in new stack -- Playing 'VCast-Answer-Machine-Excite-Mobile-Sexy-bit 1.slin' (language 'en') <--- SIP read from UDP:219.90.198.19:5060 ---> <-------------> Disconnected from Asterisk server Executing last minute cleanups