[asterisk-users] Asterisk Fax detection *11.7

[Larry Moore]

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 at 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 at from-sip:2] GotoIf("SIP/abcde-00000016",
> "0?black,1") in new stack
> -- Executing [12345678912 at from-sip:3] Ringing("SIP/abcde-00000016", "")
> in new stack
> -- Executing [12345678912 at from-sip:4] Progress("SIP/abcde-00000016", "")
> in new stack
> -- Executing [12345678912 at from-sip:5] Wait("SIP/abcde-00000016", "5") in
> new stack
> -- Executing [12345678912 at 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
>
>