I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax (.tiff) from the first asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an INVITE with audio G.711. Asterisk2 accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to Asterisk1. But, Asterisk1 responds with "488 not acceptable here". I double check t38pt_udptl = yes in my sip.conf. Why not Asterisk1 can not accept the Re-INVITE with T.38 SDP? What do I miss?
//////////////////////////////////////////////////////////////////// dev10*CLI> fax show version Fax For Asterisk Components: dev10*CLApplications: 1.6.1_1.0.11 dev10*CLDigium Fax T.38 Driver: 1.6.1_1.0.9 (optimized for i686_32) dev10*CLDigium Fax G.711 Driver: 1.6.1_1.0.9 (optimized for i686_32) ---------------------------------------------------------- .call file Channel: SIP/1...@outbound-calls MaxRetries: 3 WaitTime: 30 Set: LOCALSTATIONID=22222 Set: LOCALHEADERINFO=T38 fax Set: T38CALL=1 Set: T38TXDETECT=yes CallerID: 123456 Context: fax-tx Extension: send priority:1 ----------------------------------------------------------- sip.conf [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw t38pt_udptl = yes ------------------------------------------------------------ extensions.conf [fax-rx] exten => receive,1,NoOp(**** FAX RECEIVE ****) exten => receive,n,Set(GLOBAL(FAXCOUNT)=${GLOBAL(FAXCOUNT)}+1) exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)}) exten => receive,n,Set(FAXFILE=fax-${FAXCOUNT}-rx.tif) exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)}) exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)}) exten => receive,n,NoOp(**** SETTING FAXOPT ****) exten => receive,n,Set(FAXOPT(ecm)=yes) exten => receive,n,Set(FAXOPT(headerinfo)=MY FAXBACK RX) exten => receive,n,Set(FAXOPT(localstationid)=1234567890) exten => receive,n,Set(FAXOPT(maxrate)=14400) exten => receive,n,Set(FAXOPT(minrate)=2400) exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}) exten => receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}) exten => receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}) exten => receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)}) exten => receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)}) exten => receive,n,NoOp(**** RECEIVING FAX : ${FAXFILE} ****) exten => receive,n,ReceiveFAX(/home/sip/fax/${FAXFILE}) [fax-tx] exten => send,1,NoOp(**** SENDING FAX ****) exten => send,n,Wait(6) exten => send,n,Set(GLOBAL(FAXCOUNT)=1) ;exten => send,n,Set(GLOBAL(FAXCOUNT)= ${GLOBAL(FAXCOUNT)}+1) exten => send.,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)}) exten => send,n,Set(FAXFILE=test.tif) ; Set FAXOPTs exten => send,n,NoOp(**** SETTING FAXOPT ****) exten => send,n,Set(FAXOPT(ecm)=yes) exten => send,n,Set(FAXOPT(headerinfo)=Fax from ${GLOBAL(LASTFAXCALLERNAME)} at ${GLOBAL(LASTFAXCALLERNUM)} was received.) exten => send,n,Set(FAXOPT(localstationid)=1234567890) exten => send,n,Set(FAXOPT(maxrate)=14400) exten => send,n,Set(FAXOPT(minrate)=2400) ; Send the fax exten => send,n,NoOp(**** SENDING FAX : ${FAXFILE} ****) exten => send,n,SendFAX(/home/sip/fax/${FAXFILE},d) [default] exten => _X.,1,NoOp(**** FAX DETECTED ****) exten => _X.,n,Goto(fax-rx,receive,1) ---------------------------------------------------------- The SIP trace is # U 2009/07/15 22:30:11.588436 74.13.233.143:5060 -> 209.167.0.151:5060 INVITE sip:1...@209.167.0.151 SIP/2.0..Via: SIP/2.0/UDP 74.13.233.143:5060;branch=z9hG4bK092e48ce;rport..Max-Forwards: 70..From : "123456" <sip:123...@74.13.233.143>;tag=as74992a24..To: <sip:1...@209.167.0.151>..Contact: <sip:123...@74.13.233.143>..Call-I D: 422fd4375fe79a5977e891870f5cc...@74.13.233.143..cseq: 102 INVITE..User-Agent: Asterisk PBX 1.6.1.1..Date: Wed, 15 Jul 2009 2 2:30:11 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces, timer..Content-Type: appl ication/sdp..Content-Length: 265....v=0..o=root 1425900082 1425900082 IN IP4 74.13.233.143..s=Asterisk PBX 1.6.1.1..c=IN IP4 74 .13.233.143..t=0 0..m=audio 18452 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=si lenceSupp:off - - - -..a=ptime:20..a=sendrecv.. # U 2009/07/15 22:30:11.723006 209.167.0.151:5060 -> 74.13.233.143:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 74.13.233.143:5060;branch=z9hG4bK092e48ce;received=74.13.233.143;rport=5060..From: "123456 " <sip:123...@74.13.233.143>;tag=as74992a24..To: <sip:1...@209.167.0.151>..Call-ID: 422fd4375fe79a5977e891870f5cc...@74.13.233. 143..CSeq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Suppor ted: replaces, timer..Contact: <sip:1...@209.167.0.151>..Content-Length: 0.... # U 2009/07/15 22:30:11.730205 209.167.0.151:5060 -> 74.13.233.143:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 74.13.233.143:5060;branch=z9hG4bK092e48ce;received=74.13.233.143;rport=5060..From: "123456" <s ip:123...@74.13.233.143>;tag=as74992a24..To: <sip:1...@209.167.0.151>;tag=as3114c7a3..Call-ID: 422fd4375fe79a5977e891870f5cc05b @74.13.233.143..CSeq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOT IFY..Supported: replaces, timer..Contact: <sip:1...@209.167.0.151>..Content-Type: application/sdp..Content-Length: 265....v=0.. o=root 2128364626 2128364626 IN IP4 209.167.0.151..s=Asterisk PBX 1.6.1.1..c=IN IP4 209.167.0.151..t=0 0..m=audio 13848 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=send recv.. # U 2009/07/15 22:30:11.730460 74.13.233.143:5060 -> 209.167.0.151:5060 ACK sip:1...@209.167.0.151 SIP/2.0..Via: SIP/2.0/UDP 74.13.233.143:5060;branch=z9hG4bK271a4788;rport..Max-Forwards: 70..From: " 123456" <sip:123...@74.13.233.143>;tag=as74992a24..To: <sip:1...@209.167.0.151>;tag=as3114c7a3..Contact: <sip:123...@74.13.233. 143>..Call-ID: 422fd4375fe79a5977e891870f5cc...@74.13.233.143..cseq: 102 ACK..User-Agent: Asterisk PBX 1.6.1.1..Content-Length: 0.... # U 2009/07/15 22:30:12.140990 209.167.0.151:5060 -> 74.13.233.143:5060 INVITE sip:123...@74.13.233.143 SIP/2.0..Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK00000000;rport..Max-Forwards: 70..Fr om: <sip:1...@209.167.0.151>;tag=as3114c7a3..To: "123456" <sip:123...@74.13.233.143>;tag=as74992a24..Contact: <sip:1...@209.167 .0.151>..Call-ID: 422fd4375fe79a5977e891870f5cc...@74.13.233.143..cseq: 102 INVITE..User-Agent: Asterisk PBX 1.6.1.1..Allow: IN VITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces, timer..Content-Type: application/sdp..Content-L ength: 288....v=0..o=root 2128364626 2128364627 IN IP4 209.167.0.151..s=Asterisk PBX 1.6.1.1..c=IN IP4 209.167.0.151..t=0 0..m= image 4240 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:9600..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:400..a= T38FaxMaxDatagram:400..a=T38FaxUdpEC:t38UDPFEC.. # U 2009/07/15 22:30:12.141353 74.13.233.143:5060 -> 209.167.0.151:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK00000000;received=209.167.0.151;rport=5060..From: <sip:19 0...@209.167.0.151>;tag=as3114c7a3..To: "123456" <sip:123...@74.13.233.143>;tag=as74992a24..Call-ID: 422fd4375fe79a5977e891870f5c c...@74.13.233.143..cseq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces, timer..Contact: <sip:123...@74.13.233.143>..Content-Length: 0.... # U 2009/07/15 22:30:17.140683 74.13.233.143:5060 -> 209.167.0.151:5060 SIP/2.0 488 Not acceptable here..Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK00000000;received=209.167.0.151;rport=5060.. From: <sip:1...@209.167.0.151>;tag=as3114c7a3..To: "123456" <sip:123...@74.13.233.143>;tag=as74992a24..Call-ID: 422fd4375fe79a5 977e891870f5cc...@74.13.233.143..cseq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFE R, SUBSCRIBE, NOTIFY..Supported: replaces, timer..Content-Length: 0..X-Asterisk-HangupCause: Normal Clearing..X-Asterisk-Hangup CauseCode: 16.... # U 2009/07/15 22:30:17.186266 209.167.0.151:5060 -> 74.13.233.143:5060 ACK sip:123...@74.13.233.143 SIP/2.0..Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK00000000;rport..Max-Forwards: 70..From: <sip:1...@209.167.0.151>;tag=as3114c7a3..To: "123456" <sip:123...@74.13.233.143>;tag=as74992a24..Contact: <sip:1...@209.167.0. 151>..Call-ID: 422fd4375fe79a5977e891870f5cc...@74.13.233.143..cseq: 102 ACK..User-Agent: Asterisk PBX 1.6.1.1..Content-Length: 0.... # ------------------------------------------------------------ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users