Asterisk 1.8.4, 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. ========================================================================= Connected to Asterisk 1.8.4 currently running on fw-office (pid = 13433) fw-office*CLI> Verbosity is at least 10 fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK57708cce;rport=5060 Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de From: "Anonymous,";tag=as3256636a To: ;tag=uh5og676-CC-28 CSeq: 103 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Reason: Q.850;cause=7;text="Call awarded and being delivered in an established channel" Contact: Content-Length: 219 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 1593371 1593371 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 23930 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 <-------------> --- (11 headers 10 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 - 0xe (gsm|ulaw|alaw), 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 213.148.136.2:23930 fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK57708cce;rport=5060 Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de From: "Anonymous,";tag=as3256636a To: ;tag=uh5og676-CC-28 CSeq: 103 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Contact: Content-Length: 219 Content-Type: application/sdp v=0 fw-office*CLI> o=HuaweiSoftX3000 1593371 1593372 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 23930 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 <-------------> --- (10 headers 10 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 - 0xe (gsm|ulaw|alaw), 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 213.148.136.2:23930 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.148.136.2:5060 Transmitting (no NAT) to 213.148.136.2:5060: ACK sip:213.148.136.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK04dae442 Max-Forwards: 70 From: "Anonymous," ;tag=as3256636a To: ;tag=uh5og676-CC-28 Contact: Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.4 Content-Length: 0 fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  -- Attempting call on SIP/05119400268@sip.qsc.de-515180 for fax_out@fax-autodial-outgoing:1 (Retry 1) == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.148.136.2:5060: INVITE sip:05119400268@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK2a012c49 Max-Forwards: 70 From: "Anonymous," ;tag=as025d3112 To: Contact: Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.4 Date: Thu, 19 May 2011 12:06:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 310 v=0 o=root 1086637560 1086637560 IN IP4 85.158.179.66 s=Asterisk PBX 1.8.4 c=IN IP4 85.158.179.66 t=0 0 m=audio 11234 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 --- fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK2a012c49;rport=5060 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: "Anonymous,";tag=as025d3112 To: CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK2a012c49;rport=5060 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: "Anonymous,";tag=as025d3112 To: ;tag=56a1c28b CSeq: 102 INVITE Proxy-Authenticate: Digest realm="qsc.de",nonce="TdUI4U3VB7VP2u7abqzz/dLaUH9jJKFg",qop="auth" Server: QSC SiP server node 03 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 213.148.136.2:5060: ACK sip:05119400268@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK2a012c49 Max-Forwards: 70 From: "Anonymous," ;tag=as025d3112 To: ;tag=56a1c28b Contact: Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.4 Content-Length: 0 --- Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.148.136.2:5060: INVITE sip:05119400268@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK7370f288 Max-Forwards: 70 From: "Anonymous," ;tag=as025d3112 To: Contact: Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.4 Proxy-Authorization: Digest username="0511515180", realm="qsc.de", algorithm=MD5, uri="sip:05119400268@sip.qsc.de", nonce="TdUI4U3VB7VP2u7abqzz/dLaUH9jJKFg", response="8bfecebe116ea834f8c7bc604b6c5055", qop=auth, cnonce="6fb2a46e", nc=00000001 Date: Thu, 19 May 2011 12:06:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 310 v=0 o=root 1086637560 1086637561 IN IP4 85.158.179.66 s=Asterisk PBX 1.8.4 c=IN IP4 85.158.179.66 t=0 0 m=audio 11234 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 --- fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK7370f288;rport=5060 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: "Anonymous,";tag=as025d3112 To: CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- fw-office*CLI>  -- Executing [fax_out@fax-autodial-outgoing:5] Set("SIP/sip.qsc.de-515180-0000005d", "FAXOPT(ecm)=yes") in new stack -- Executing [fax_out@fax-autodial-outgoing:6] Set("SIP/sip.qsc.de-515180-0000005d", "FAXOPT(maxrate)=9600") in new stack -- Executing [fax_out@fax-autodial-outgoing:7] SendFAX("SIP/sip.qsc.de-515180-0000005d", "/opt/fax/in/1305805569.72.tif") in new stack -- Channel 'SIP/sip.qsc.de-515180-0000005d' sending FAX: -- /opt/fax/in/1305805569.72.tif fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> INVITE sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKbca8c5a5a790e38710e2e8222 Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a CSeq: 1 INVITE Max-Forwards: 69 Contact: Content-Length: 529 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 1593371 1593373 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=image 23930 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy m=audio 23986 RTP/AVP 8 0 127 103 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:127 PCMU/8000 a=gpmd:127 vbd=yes a=rtpmap:103 PCMA/8000 a=gpmd:103 vbd=yes a=rtpmap:101 telephone-event/8000 a=ptime:20 a=silenceSupp:off - - - - a=ecan:fb on - a=X-fax a=fmtp:101 0-15 <-------------> --- (10 headers 23 lines) --- Sending to 213.148.136.2:5060 (no NAT) Got T.38 offer in SDP in dialog 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 127 Found RTP audio format 103 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format PCMU for ID 127 Found audio description format PCMA for ID 103 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 213.148.136.2:23986 <--- Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKbca8c5a5a790e38710e2e8222;received=213.148.136.2 From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de CSeq: 1 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> fw-office*CLI>  <--- Reliably Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKbca8c5a5a790e38710e2e8222;received=213.148.136.2 From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de CSeq: 1 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 307 v=0 o=root 1392571112 1392571114 IN IP4 85.158.179.66 s=Asterisk PBX 1.8.4 c=IN IP4 85.158.179.66 t=0 0 m=audio 0 RTP/AVP 8 0 127 103 101 m=image 4271 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> ACK sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKd685ac22ddbfd50baafd27235 Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a CSeq: 1 ACK Max-Forwards: 69 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK7370f288;rport=5060 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: "Anonymous,";tag=as025d3112 To: ;tag=sy1h6lst-CC-24 CSeq: 103 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Reason: Q.850;cause=7;text="Call awarded and being delivered in an established channel" Contact: Content-Length: 219 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 1970539 1970539 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 23990 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 <-------------> --- (11 headers 10 lines) --- fw-office*CLI> 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 - 0xe (gsm|ulaw|alaw), 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 213.148.136.2:23990 fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK7370f288;rport=5060 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: "Anonymous,";tag=as025d3112 To: ;tag=sy1h6lst-CC-24 CSeq: 103 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Contact: Content-Length: 219 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 1970539 1970540 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 23990 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 <-------------> fw-office*CLI> --- (10 headers 10 lines) --- fw-office*CLI> Found RTP audio format 8 fw-office*CLI> Found RTP audio format 101 fw-office*CLI> Found audio description format PCMA for ID 8 fw-office*CLI> Found audio description format telephone-event for ID 101 fw-office*CLI> Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) fw-office*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) fw-office*CLI> Peer audio RTP is at port 213.148.136.2:23990 fw-office*CLI> list_route: hop: fw-office*CLI> set_destination: Parsing for address/port to send to fw-office*CLI> set_destination: set destination to 213.148.136.2:5060 fw-office*CLI> Transmitting (no NAT) to 213.148.136.2:5060: ACK sip:213.148.136.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 85.158.179.66:5060;branch=z9hG4bK3bb2c0ee Max-Forwards: 70 From: "Anonymous," ;tag=as025d3112 To: ;tag=sy1h6lst-CC-24 Contact: Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.4 Content-Length: 0 --- fw-office*CLI>  > Channel SIP/sip.qsc.de-515180-0000005e was answered. -- Executing [fax_out@fax-autodial-outgoing:1] Set("SIP/sip.qsc.de-515180-0000005e", "LOCALSTATIONID=""") in new stack -- Executing [fax_out@fax-autodial-outgoing:2] Set("SIP/sip.qsc.de-515180-0000005e", "LOCALHEADERINFO=""") in new stack -- Executing [fax_out@fax-autodial-outgoing:3] Answer("SIP/sip.qsc.de-515180-0000005e", "") in new stack -- Executing [fax_out@fax-autodial-outgoing:4] Wait("SIP/sip.qsc.de-515180-0000005e", "2") in new stack fw-office*CLI> Really destroying SIP dialog '1ced3b112a0412da6785c1d46571ff2e@sip.qsc.de' Method: BYE fw-office*CLI>  -- Executing [fax_out@fax-autodial-outgoing:5] Set("SIP/sip.qsc.de-515180-0000005e", "FAXOPT(ecm)=yes") in new stack -- Executing [fax_out@fax-autodial-outgoing:6] Set("SIP/sip.qsc.de-515180-0000005e", "FAXOPT(maxrate)=9600") in new stack -- Executing [fax_out@fax-autodial-outgoing:7] SendFAX("SIP/sip.qsc.de-515180-0000005e", "/opt/fax/in/1305805569.72.tif") in new stack -- Channel 'SIP/sip.qsc.de-515180-0000005e' sending FAX: -- /opt/fax/in/1305805569.72.tif fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> INVITE sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKa378faabd56ba73211f64c4bf Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: ;tag=sy1h6lst-CC-24 To: "Anonymous,";tag=as025d3112 CSeq: 1 INVITE Max-Forwards: 69 Contact: Content-Length: 529 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 1970539 1970541 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=image 23990 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy m=audio 24022 RTP/AVP 8 0 127 103 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:127 PCMU/8000 a=gpmd:127 vbd=yes a=rtpmap:103 PCMA/8000 a=gpmd:103 vbd=yes a=rtpmap:101 telephone-event/8000 a=ptime:20 a=silenceSupp:off - - - - a=ecan:fb on - a=X-fax a=fmtp:101 0-15 <-------------> --- (10 headers 23 lines) --- Sending to 213.148.136.2:5060 (no NAT) Got T.38 offer in SDP in dialog 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 127 Found RTP audio format 103 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format PCMU for ID 127 Found audio description format PCMA for ID 103 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 213.148.136.2:24022 <--- Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKa378faabd56ba73211f64c4bf;received=213.148.136.2 From: ;tag=sy1h6lst-CC-24 To: "Anonymous,";tag=as025d3112 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de CSeq: 1 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> fw-office*CLI>  <--- Reliably Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKa378faabd56ba73211f64c4bf;received=213.148.136.2 From: ;tag=sy1h6lst-CC-24 To: "Anonymous,";tag=as025d3112 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de CSeq: 1 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 307 v=0 o=root 1086637560 1086637562 IN IP4 85.158.179.66 s=Asterisk PBX 1.8.4 c=IN IP4 85.158.179.66 t=0 0 m=audio 0 RTP/AVP 8 0 127 103 101 m=image 4191 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> ACK sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK5183f0856901eec3c038fbe32 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: ;tag=sy1h6lst-CC-24 To: "Anonymous,";tag=as025d3112 CSeq: 1 ACK Max-Forwards: 69 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> INVITE sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKb3413548a1b3f8b8dba3ef332 Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a CSeq: 2 INVITE Max-Forwards: 69 Contact: Content-Length: 219 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 1593371 1593374 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 23930 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 <-------------> --- (10 headers 10 lines) --- Sending to 213.148.136.2:5060 (no NAT) 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 - 0xe (gsm|ulaw|alaw), 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 213.148.136.2:23930 <--- Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKb3413548a1b3f8b8dba3ef332;received=213.148.136.2 From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de CSeq: 2 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bKb3413548a1b3f8b8dba3ef332;received=213.148.136.2 From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de CSeq: 2 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 1392571112 1392571115 IN IP4 85.158.179.66 s=Asterisk PBX 1.8.4 c=IN IP4 85.158.179.66 t=0 0 m=audio 11144 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 <------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> ACK sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK2f1c3fad62781b9238d779263 Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> BYE sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK33287709b2dc9bd245fde8b7a Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a CSeq: 3 BYE Max-Forwards: 69 Reason: Q.850;cause=16;text="normal call clearing" Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 213.148.136.2:5060 (no NAT) Scheduling destruction of SIP dialog '36f3892d3f88afa46413f9c4773432ba@sip.qsc.de' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK33287709b2dc9bd245fde8b7a;received=213.148.136.2 From: ;tag=uh5og676-CC-28 To: "Anonymous,";tag=as3256636a Call-ID: 36f3892d3f88afa46413f9c4773432ba@sip.qsc.de CSeq: 3 BYE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> fw-office*CLI>  == Spawn extension (fax-autodial-outgoing, fax_out, 7) exited non-zero on 'SIP/sip.qsc.de-515180-0000005d' fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:1] Verbose("SIP/sip.qsc.de-515180-0000005d", "### FAXSTATUS: FAILED") in new stack fw-office*CLI> ### FAXSTATUS: FAILED fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:2] Verbose("SIP/sip.qsc.de-515180-0000005d", "### FAXERROR: The call dropped prematurely") in new stack fw-office*CLI> ### FAXERROR: The call dropped prematurely fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:3] Verbose("SIP/sip.qsc.de-515180-0000005d", "### FAXSTATUSSTRING: The call dropped prematurely") in new stack fw-office*CLI> ### FAXSTATUSSTRING: The call dropped prematurely fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:4] Verbose("SIP/sip.qsc.de-515180-0000005d", "### FAXPAGES: 0") in new stack fw-office*CLI> ### FAXPAGES: 0 fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:5] Verbose("SIP/sip.qsc.de-515180-0000005d", "### FAXBITRATE: 14400") in new stack fw-office*CLI> ### FAXBITRATE: 14400 fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:6] Verbose("SIP/sip.qsc.de-515180-0000005d", "### FAXRESOLUTION: 0x0") in new stack fw-office*CLI> ### FAXRESOLUTION: 0x0 fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:7] Verbose("SIP/sip.qsc.de-515180-0000005d", "### REMOTESTATIONID: ") in new stack fw-office*CLI> ### REMOTESTATIONID: fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:8] Verbose("SIP/sip.qsc.de-515180-0000005d", "### FAXPATH: /opt/fax/in/") in new stack fw-office*CLI> ### FAXPATH: /opt/fax/in/ fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:9] Verbose("SIP/sip.qsc.de-515180-0000005d", "### FAXFILE: 1305805569.72.tif") in new stack fw-office*CLI> ### FAXFILE: 1305805569.72.tif fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:10] Verbose("SIP/sip.qsc.de-515180-0000005d", "### DEST: 05119400268") in new stack fw-office*CLI> ### DEST: 05119400268 fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:11] Verbose("SIP/sip.qsc.de-515180-0000005d", "### RETRYCOUNT: 29") in new stack fw-office*CLI> ### RETRYCOUNT: 29 fw-office*CLI>  -- Executing [h@fax-autodial-outgoing:12] System("SIP/sip.qsc.de-515180-0000005d", "RETRYCOUNT=\"29\" FAXSTATUS=\"FAILED\" FAXERROR=\"The call dropped prematurely\" FAXPATH=\"/opt/fax/in/\" FAXFILE=\"1305805569.72.tif\" DEST=\"05119400268\" /opt/asterisk-faxcall.pl") in new stack fw-office*CLI> [May 19 14:06:37] NOTICE[13978]: pbx_spool.c:362 attempt_thread: Call completed to SIP/05119400268@sip.qsc.de-515180 fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> hello <-------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> INVITE sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK8997bee2fb9956e86822fec5c Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: ;tag=sy1h6lst-CC-24 To: "Anonymous,";tag=as025d3112 CSeq: 2 INVITE Max-Forwards: 69 Contact: Content-Length: 219 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 1970539 1970542 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 23990 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 <-------------> fw-office*CLI> --- (10 headers 10 lines) --- fw-office*CLI> Sending to 213.148.136.2:5060 (no NAT) fw-office*CLI> Found RTP audio format 8 fw-office*CLI> Found RTP audio format 101 fw-office*CLI> Found audio description format PCMA for ID 8 fw-office*CLI> Found audio description format telephone-event for ID 101 fw-office*CLI> Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) fw-office*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) fw-office*CLI> Peer audio RTP is at port 213.148.136.2:23990 fw-office*CLI>  <--- Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK8997bee2fb9956e86822fec5c;received=213.148.136.2 From: ;tag=sy1h6lst-CC-24 To: "Anonymous,";tag=as025d3112 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de CSeq: 2 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> fw-office*CLI> Audio is at 5060 fw-office*CLI> Adding codec 0x8 (alaw) to SDP fw-office*CLI> Adding non-codec 0x1 (telephone-event) to SDP fw-office*CLI>  <--- Reliably Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK8997bee2fb9956e86822fec5c;received=213.148.136.2 From: ;tag=sy1h6lst-CC-24 To: "Anonymous,";tag=as025d3112 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de CSeq: 2 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 1086637560 1086637563 IN IP4 85.158.179.66 s=Asterisk PBX 1.8.4 c=IN IP4 85.158.179.66 t=0 0 m=audio 11234 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 <------------> fw-office*CLI>  <--- SIP read from UDP:213.148.136.2:5060 ---> ACK sip:0511515180@85.158.179.66:5060 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK0617bae6aaa470450d36fa220 Call-ID: 7f57790c596c8f8a0b1b750d18611685@sip.qsc.de From: ;tag=sy1h6lst-CC-24 To: "Anonymous,";tag=as025d3112 CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 <-------------> fw-office*CLI> --- (8 headers 0 lines) --- fw-office*CLI> Disconnected from Asterisk server