VoIP Mailing List Archives

[jakob at j-mb.de]

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

[ldardini at gmail.com]

It is really more interesting the receiving part. Can you paste here?

Leandro



2014/1/21 Jakob-Matthias Böttger <jakob@j-mb.de (jakob@j-mb.de)>

[jakob at j-mb.de]

Hi

The log i've posted

== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [url=tel:%5B12345678912][12345678912[/url]@from-sip:1] Answer("SIP/abcde-00000016", "") in new stack

[ldardini at gmail.com]

I am not sure, but try to add a wait(2) as first command. When I want fax detection, I insert always a small delay for letting the fax detection routine to detect it.

Leandro



2014/1/21 Jakob-Matthias Böttger <jakob@j-mb.de (jakob@j-mb.de)>

[jakob at j-mb.de]

i already added a Progess() and Wait(5) and it still does not detect faxes.


Am 21.01.2014 16:53, schrieb Leandro Dardini:

[ldardini at gmail.com]

Please paste the actual code. First has to be the Wait and then any other thing.

Leandro



2014/1/21 Jakob-Matthias Böttger <jakob@j-mb.de (jakob@j-mb.de)>

[paul.belanger at polyb...]

On Tue, Jan 21, 2014 at 5:51 AM, Jakob-Matthias Böttger <jakob@j-mb.de> wrote:

[lmoore at omninet.net.au]

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:

[lmoore at omninet.net.au]

Sorry, I missed the line showing the call had been answered.

On 22/01/2014 8:11 AM, Larry Moore wrote:

[ra25 at atlas.cz]

[jakob at j-mb.de]

Hello,

now i added

directmedia=no
disallow=all
allow=ulaw
allow=alaw

and i changed the caninvite part to canreinvite.
Now the faxdetection is working well. But now, after the faxsession has
started, i'm getting

res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short

as error.

Regards Jakob

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