DIAL PLAN: [voxbone-in] ; exten => 16477245778,1,Answer() exten => 16477245778,2,Set(TIMEOUT(digit)=59) exten => 16477245778,3,Set(FAXFILE=/tmp/incomingfax.tif) exten => 16477245778,4,RxFAX(${FAXFILE}) ; exten => t,1,Hangup() ; ; ################################################# LOG ########################################### atpserver104*CLI> <-- SIP read from 81.201.84.26:5060: INVITE sip:16477245778@66.165.164.45 SIP/2.0 Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3f87916b;rport From: "17868667762" ;tag=as5684bd40 To: Contact: Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 102 INVITE User-Agent: Vox Callcontrol Max-Forwards: 70 Date: Thu, 24 May 2007 15:05:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 311 v=0 o=root 25697 25697 IN IP4 81.201.84.26 s=session c=IN IP4 81.201.84.26 t=0 0 m=audio 12150 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 15 lines) --- Using INVITE request as basis request - 275022702403d37275677f4513807370@81.201.84.26 Sending to 81.201.84.26 : 5060 (NAT) Found peer '81.201.84.26' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 81.201.84.26:12150 Found description format PCMA Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 16477245778 in voxbone-in (domain 66.165.164.45) list_route: hop: Transmitting (NAT) to 81.201.84.26:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3f87916b;rport;received=81.201.84.26 From: "17868667762" ;tag=as5684bd40 To: Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 102 INVITE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- We're at 66.165.164.45 port 11396 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 81.201.84.26:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3f87916b;rport;received=81.201.84.26 From: "17868667762" ;tag=as5684bd40 To: ;tag=as6eb23e61 Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 102 INVITE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 269 v=0 o=root 25483 25483 IN IP4 66.165.164.45 s=session c=IN IP4 66.165.164.45 t=0 0 m=audio 11396 RTP/AVP 0 8 122 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:122 t38/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- atpserver104*CLI> <-- SIP read from 81.201.84.26:5060: ACK sip:16477245778@66.165.164.45 SIP/2.0 Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK35c64c71;rport From: "17868667762" ;tag=as5684bd40 To: ;tag=as6eb23e61 Contact: Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 102 ACK User-Agent: Vox Callcontrol Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 81.201.84.26, port 5060 T.38 UDPTL is at port 66.165.164.45:11396 13 headers, 15 lines Reliably Transmitting (NAT) to 81.201.84.26:5060: INVITE sip:17868667762@81.201.84.26 SIP/2.0 Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK573868d2;rport From: ;tag=as6eb23e61 To: "17868667762" ;tag=as5684bd40 Contact: Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 102 INVITE User-Agent: CallWeaver Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-callweaver-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 346 v=0 o=root 25483 25484 IN IP4 66.165.164.45 s=session c=IN IP4 66.165.164.45 t=0 0 m=image 11396 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPFEC --- May 24 11:33:03 NOTICE[3062512560]: channel.c:1651 opbx_read: Dropping incompatible voice frame on SIP/81.201.84.26-b6703008 of format slin since our native format has changed to alaw atpserver104*CLI> <-- SIP read from 81.201.84.26:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK573868d2;received=66.165.164.45;rport=5060 From: ;tag=as6eb23e61 To: "17868667762" ;tag=as5684bd40 Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 102 INVITE User-Agent: Vox Callcontrol Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- (12 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 81.201.84.26, port 5060 Transmitting (NAT) to 81.201.84.26:5060: ACK sip:17868667762@81.201.84.26 SIP/2.0 Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK573868d2;rport From: ;tag=as6eb23e61 To: "17868667762" ;tag=as5684bd40 Contact: Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 102 ACK User-Agent: CallWeaver Max-Forwards: 70 Content-Length: 0 --- atpserver104*CLI> <-- SIP read from 81.201.84.26:5060: BYE sip:16477245778@66.165.164.45 SIP/2.0 Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3e390820;rport From: "17868667762" ;tag=as5684bd40 To: ;tag=as6eb23e61 Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 103 BYE User-Agent: Vox Callcontrol Max-Forwards: 70 Content-Length: 0 --- (9 headers 0 lines) --- Sending to 81.201.84.26 : 5060 (NAT) Transmitting (NAT) to 81.201.84.26:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3e390820;received=81.201.84.26;rport=5060 From: "17868667762" ;tag=as5684bd40 To: ;tag=as6eb23e61 Call-ID: 275022702403d37275677f4513807370@81.201.84.26 CSeq: 103 BYE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 X-CallWeaver-HangupCause: Bearer capability not available --- Destroying call '275022702403d37275677f4513807370@81.201.84.26' atpserver104*CLI> <-- SIP read from 81.201.84.27:5060: INVITE sip:16477245778@66.165.164.45 SIP/2.0 Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK55a6bdc9;rport From: "17868667762" ;tag=as1cc96db7 To: Contact: Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 102 INVITE User-Agent: Vox Callcontrol Max-Forwards: 70 Date: Thu, 24 May 2007 15:07:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 309 v=0 o=root 8947 8947 IN IP4 81.201.84.27 s=session c=IN IP4 81.201.84.27 t=0 0 m=audio 18150 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 15 lines) --- Using INVITE request as basis request - 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 Sending to 81.201.84.27 : 5060 (NAT) Found peer '81.201.84.27' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 81.201.84.27:18150 Found description format PCMA Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 16477245778 in voxbone-in (domain 66.165.164.45) list_route: hop: Transmitting (NAT) to 81.201.84.27:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK55a6bdc9;rport;received=81.201.84.27 From: "17868667762" ;tag=as1cc96db7 To: Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 102 INVITE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- We're at 66.165.164.45 port 17188 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 81.201.84.27:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK55a6bdc9;rport;received=81.201.84.27 From: "17868667762" ;tag=as1cc96db7 To: ;tag=as4ae3ba28 Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 102 INVITE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 269 v=0 o=root 25483 25483 IN IP4 66.165.164.45 s=session c=IN IP4 66.165.164.45 t=0 0 m=audio 17188 RTP/AVP 0 8 122 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:122 t38/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- atpserver104*CLI> <-- SIP read from 81.201.84.27:5060: ACK sip:16477245778@66.165.164.45 SIP/2.0 Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK1ba829c6;rport From: "17868667762" ;tag=as1cc96db7 To: ;tag=as4ae3ba28 Contact: Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 102 ACK User-Agent: Vox Callcontrol Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 81.201.84.27, port 5060 T.38 UDPTL is at port 66.165.164.45:17188 13 headers, 15 lines Reliably Transmitting (NAT) to 81.201.84.27:5060: INVITE sip:17868667762@81.201.84.27 SIP/2.0 Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK460e5419;rport From: ;tag=as4ae3ba28 To: "17868667762" ;tag=as1cc96db7 Contact: Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 102 INVITE User-Agent: CallWeaver Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-callweaver-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 346 v=0 o=root 25483 25484 IN IP4 66.165.164.45 s=session c=IN IP4 66.165.164.45 t=0 0 m=image 17188 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPFEC --- May 24 11:35:18 NOTICE[3062512560]: channel.c:1651 opbx_read: Dropping incompatible voice frame on SIP/81.201.84.27-b6703008 of format slin since our native format has changed to alaw atpserver104*CLI> <-- SIP read from 81.201.84.27:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK460e5419;received=66.165.164.45;rport=5060 From: ;tag=as4ae3ba28 To: "17868667762" ;tag=as1cc96db7 Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 102 INVITE User-Agent: Vox Callcontrol Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- (12 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 81.201.84.27, port 5060 Transmitting (NAT) to 81.201.84.27:5060: ACK sip:17868667762@81.201.84.27 SIP/2.0 Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK460e5419;rport From: ;tag=as4ae3ba28 To: "17868667762" ;tag=as1cc96db7 Contact: Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 102 ACK User-Agent: CallWeaver Max-Forwards: 70 Content-Length: 0 --- atpserver104*CLI> <-- SIP read from 81.201.84.27:5060: BYE sip:16477245778@66.165.164.45 SIP/2.0 Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK530a0398;rport From: "17868667762" ;tag=as1cc96db7 To: ;tag=as4ae3ba28 Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 103 BYE User-Agent: Vox Callcontrol Max-Forwards: 70 Content-Length: 0 --- (9 headers 0 lines) --- Sending to 81.201.84.27 : 5060 (NAT) Transmitting (NAT) to 81.201.84.27:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK530a0398;received=81.201.84.27;rport=5060 From: "17868667762" ;tag=as1cc96db7 To: ;tag=as4ae3ba28 Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27 CSeq: 103 BYE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 -CallWeaver-HangupCause: Bearer capability not available --- Destroying call '1bf9c0e031362984645b01ec75a99ade@81.201.84.27' atpserver104*CLI> exit