Sorry, I missed the line showing the call had been answered.
On 22/01/2014 8:11 AM, Larry Moore wrote:
Hello, Perhaps you need to have directmedia=no set for the channel, the call doesn't appear to have been answered hence asterisk won't be able to hear any tones to determine for itself if the call is an incoming fax. Larry. On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote:Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --------------- Current Sessions : 0 Reserved Sessions : 0 Transmit Attempts : 0 Receive Attempts : 1 Completed FAXes : 1 Failed FAXes : 1 Digium G.711 Licensed Channels : 1 Max Concurrent : 0 Success : 0 Switched to T.38 : 0 Canceled : 0 No FAX : 0 Partial : 0 Negotiation Failed : 0 Train Failure : 0 Protocol Error : 0 IO Partial : 0 IO Fail : 0 Digium T.38 Licensed Channels : 1 Max Concurrent : 1 Success : 0 Canceled : 0 No FAX : 0 Partial : 0 Negotiation Failed : 0 Train Failure : 1 Protocol Error : 0 IO Partial : 0 IO Fail : 0 so that should be ok. The corresponding dialplan section starts with [from-sip] include => inbound [inbound] exten => _X.,1,Answer() exten => _X.,n,GotoIf(${BLACKLIST()}?black,1) exten => _X.,n,Ringing exten => _X.,n,Progress() exten => _X.,n,Wait(5) exten => _X.,n,Dial(SIP/123&SIP/456,30,oxX) ... exten => fax,1,NoOp(**** FAX DETECTED ****) exten => fax,n,Goto(fax-rx,receive,1) in the sip.conf i specified [general] sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes caninvite=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes ... [abcde] type=peer insecure=invite defaultuser=12345678912 fromuser=12345678912 fromdomain=abcde.ab secret=guess-what host=abcde.ab qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=12345678912 but all i can see if i try to send a testfax is == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912@from-sip:1] Answer("SIP/abcde-00000016", "") in new stack > 0x7fd11404cd00 -- Probation passed - setting RTP source address to 123.456.789.123:17108 -- Executing [12345678912@from-sip:2] GotoIf("SIP/abcde-00000016", "0?black,1") in new stack -- Executing [12345678912@from-sip:3] Ringing("SIP/abcde-00000016", "") in new stack -- Executing [12345678912@from-sip:4] Progress("SIP/abcde-00000016", "") in new stack -- Executing [12345678912@from-sip:5] Wait("SIP/abcde-00000016", "5") in new stack -- Executing [12345678912@from-sip:6] Dial("SIP/abcde-00000016", "SIP/123&SIP/456,30,oxX") in new stack == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 -- Called SIP/200 -- Called SIP/201 -- SIP/123-00000018 connected line has changed. Saving it until answer for SIP/abcde-00000016 -- SIP/456-00000017 connected line has changed. Saving it until answer for SIP/abcde-00000016 -- SIP/123-00000018 is ringing -- SIP/456-00000017 is ringing Any hints why thats not working? Best Regards Jakob
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