[asterisk-users] Asterisk Fax detection *11.7

[Jakob-Matthias Böttger]

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