[asterisk-users] Asterisk Fax detection *11.7

[Paul Belanger]

On Tue, Jan 21, 2014 at 5:51 AM, Jakob-Matthias Böttger <jakob at j-mb.de> 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
>
>
Don't expect T.30 over SIP to be reliable. If you need fax, you should
be using T.38. Your codec is likely the issue.

-- 
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger