-- Attempting call on SIP/0711654321@qsc for application AGI(routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif) (Retry 1) == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 93.189.169.91 port 19886 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.148.136.2:5060: INVITE sip:0711654321@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK68eba344;rport Max-Forwards: 70 From: "0711123456" ;tag=as2fe9ef7b To: Contact: Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Date: Sat, 25 Jul 2009 10:39:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 1639369616 1639369616 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 19886 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 --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK68eba344;rport=5060 Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de From: "0711123456";tag=as2fe9ef7b To: CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK68eba344;rport=5060 Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de From: "0711123456";tag=as2fe9ef7b To: ;tag=746dedca CSeq: 102 INVITE Proxy-Authenticate: Digest realm="qsc.de",nonce="4a6ae21dc59736628d8a9b04a9f313cd1087356d",qop="auth" Server: QSC SIP Router Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 213.148.136.2:5060: ACK sip:0711654321@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK68eba344;rport Max-Forwards: 70 From: "0711123456" ;tag=as2fe9ef7b To: ;tag=746dedca Contact: Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Content-Length: 0 --- Audio is at 93.189.169.91 port 19886 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.148.136.2:5060: INVITE sip:0711654321@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK081dd1e8;rport Max-Forwards: 70 From: "0711123456" ;tag=as2fe9ef7b To: Contact: Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Proxy-Authorization: Digest username="02152994430", realm="qsc.de", algorithm=MD5, uri="sip:0711654321@sip.qsc.de", nonce="4a6ae21dc59736628d8a9b04a9f313cd1087356d", response="359e60fb47167ab62620b44b454a67fa", qop=auth, cnonce="749d244f", nc=00000001 Date: Sat, 25 Jul 2009 10:39:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 1639369616 1639369617 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 19886 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 --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK081dd1e8;rport=5060 Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de From: "0711123456";tag=as2fe9ef7b To: CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK081dd1e8;rport=5060 Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de From: "0711123456";tag=as2fe9ef7b To: ;tag=393403df 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 9768123 9768123 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 12126 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 <-------------> --- (10 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.148.136.2:12126 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/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 213.148.136.2:12126 stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK081dd1e8;rport=5060 Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de From: "0711123456";tag=as2fe9ef7b To: ;tag=393403df CSeq: 103 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Supported: 100rel,replaces,timer,precondition,histinfo Contact: Content-Length: 219 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 9768123 9768124 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 12126 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 <-------------> --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.148.136.2:12126 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/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 213.148.136.2:12126 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.148.136.2, port 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 93.189.169.91:5060;branch=z9hG4bK45a515d7;rport Max-Forwards: 70 From: "0711123456" ;tag=as2fe9ef7b To: ;tag=393403df Contact: Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Content-Length: 0 --- > Channel SIP/qsc-08231078 was answered. > Launching AGI(routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif) on SIP/qsc-08231078 -- Launched AGI Script /var/lib/asterisk/agi-bin/routing.php routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_request' => 'routing.php' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_channel' => 'SIP/qsc-08231078' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_language' => 'en' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_type' => 'SIP' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_uniqueid' => '1248518385.3' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_version' => 'SVN-branch-1.6.0-r208752' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_callerid' => '0711123456' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_calleridname' => 'unknown' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_callingpres' => '0' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_callingani2' => '0' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_callington' => '0' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_callingtns' => '0' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_dnid' => 'unknown' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_rdnis' => 'unknown' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_context' => 'default' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_extension' => '' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_priority' => '1' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_enhanced' => '0.0' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_accountcode' => '' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_threadid' => '-1225352304' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: 'agi_arg_1' => '/tmp/email2fax/1248518385-467933000/test.pdf.tif' routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: string(57) "unknown ; SIP/qsc-08231078 ; 1248518385.3 ; ; ; unknown"n routing.php,/tmp/email2fax/1248518385-467933000/test.pdf.tif: >> EXEC SendFAX /tmp/email2fax/1248518385-467933000/test.pdf.tif -- AGI Script Executing Application: (SendFAX) Options: (/tmp/email2fax/1248518385-467933000/test.pdf.tif) stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> INVITE sip:02152994430@93.189.169.91 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK83b42a94fac64bead692e7e94 Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b CSeq: 1 INVITE Max-Forwards: 50 Contact: Content-Length: 431 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 9768123 9768125 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=image 12126 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy m=audio 12260 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 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 19 lines) --- Sending to 213.148.136.2 : 5060 (no NAT) Got T.38 offer in SDP in dialog 2cb10cfa055d03110b6832b82e2cc468@qsc.de Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.148.136.2:12260 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/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 213.148.136.2:12260 stars24*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=z9hG4bK83b42a94fac64bead692e7e94;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> stars24*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=z9hG4bK83b42a94fac64bead692e7e94;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1639369616 1639369618 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 0 RTP/AVP 8 0 101 m=image 4583 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <------------> Sent UDPTL packet to 213.148.136.2:12126 (type 0, seq 0, len 6) Sent UDPTL packet to 213.148.136.2:12126 (type 0, seq 1, len 8) stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> Retransmitting #1 (no NAT) to 213.148.136.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK83b42a94fac64bead692e7e94;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1639369616 1639369618 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 0 RTP/AVP 8 0 101 m=image 4583 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy --- Retransmitting #2 (no NAT) to 213.148.136.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK83b42a94fac64bead692e7e94;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1639369616 1639369618 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 0 RTP/AVP 8 0 101 m=image 4583 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy --- Retransmitting #3 (no NAT) to 213.148.136.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK83b42a94fac64bead692e7e94;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1639369616 1639369618 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 0 RTP/AVP 8 0 101 m=image 4583 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy --- Retransmitting #4 (no NAT) to 213.148.136.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK83b42a94fac64bead692e7e94;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1639369616 1639369618 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 0 RTP/AVP 8 0 101 m=image 4583 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> Retransmitting #5 (no NAT) to 213.148.136.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK83b42a94fac64bead692e7e94;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1639369616 1639369618 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 0 RTP/AVP 8 0 101 m=image 4583 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy --- Retransmitting #6 (no NAT) to 213.148.136.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK83b42a94fac64bead692e7e94;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1639369616 1639369618 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208752 c=IN IP4 93.189.169.91 t=0 0 m=audio 0 RTP/AVP 8 0 101 m=image 4583 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> [Jul 25 10:40:16] WARNING[26314]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission 2cb10cfa055d03110b6832b82e2cc468@qsc.de for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [Jul 25 10:40:16] WARNING[26314]: chan_sip.c:2944 retrans_pkt: Hanging up call 2cb10cfa055d03110b6832b82e2cc468@qsc.de - no reply to our critical packet (see doc/sip-retransmit.txt). [Jul 25 10:40:16] WARNING[26683]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Jul 25 10:40:16] WARNING[26683]: app_fax.c:677 transmit: Transmission error -- AGI Script routing.php completed, returning -1 [Jul 25 10:40:16] NOTICE[26683]: pbx_spool.c:357 attempt_thread: Call completed to SIP/0711654321@qsc Really destroying SIP dialog '2cb10cfa055d03110b6832b82e2cc468@qsc.de' Method: INVITE stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> BYE sip:02152994430@93.189.169.91 SIP/2.0 Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK5a568209a3f4615e9b4643d2e Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b CSeq: 2 BYE Max-Forwards: 50 Reason: Q.850;cause=16;text="normal call clearing" Content-Length: 0 <-------------> --- (9 headers 0 lines) --- stars24*CLI> <--- Transmitting (no NAT) to 213.148.136.2:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 213.148.136.2:5060;branch=z9hG4bK5a568209a3f4615e9b4643d2e;received=213.148.136.2 From: ;tag=393403df To: "0711123456";tag=as2fe9ef7b Call-ID: 2cb10cfa055d03110b6832b82e2cc468@qsc.de CSeq: 2 BYE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208752 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> stars24*CLI>