[asterisk-users] Spandsp rxfax txtax fails no errors

[daveasterisk]

I'm using Slackware 11.
I unistalled the package that provides libtiff 3.8.....
and installed the most current 3.7.... for lib tiff.

I downloaded asterisk 1.4 beta3 and the 1.4 beta2 addons and untared them.
created a simlink:
ln -s asterisk-1.4.0-beta3 asterisk

I've compiled spandsp from as follows

    cd /usr/src
    wget http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.3pre24.tgz
    tar -xzvf spandsp-0.0.3pre24.tgz
    cd spandsp-0.0.3
    ./configure
    make
    make install

I downloaded the rxfax, txfax, and patch files as folows:

    cd /usr/src/asterisk/
    wget
http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/app_rxfax.c
    wget
http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/app_txfax.c
    wget
http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/asterisk.patch

I tried to apply the patch but it failed so I applied it by hand.

did

    ./configure
    make menuselect

selected the fax rxfax and txfax applications and saved/exited (x option).
next

    make
    make install
    make samples

no errors for any above

cleaned up the sip.conf and extensions.conf to only have one "trunk"
line defined and DID to the fax context in extensions.conf


I can successfully have a fax machine call the asterisk box, it answer
the call and start the rxfax application. Unfortunately it doesn't do
anything.
here is the console log of recieving a fax call:

    asterisk -vvvvvvvvvvvvvvr
    Connected to Asterisk SVN-branch-1.4-r46117M currently running on
saster (pid = 12516)
    Verbosity is at least 14
        -- Executing [(asteriskfaxnumber)@dids-inbound:1]
Goto("SIP/xx.xxx.xx.xx-0821ec78", "incomingfax|fax|1") in new stack
        -- Goto (incomingfax,fax,1)
        -- Executing [fax at incomingfax:1]
Set("SIP/xxx.xxx.xx.xx-0821ec78", "FAXFILE=/tmp/recievedfax.tif") in new
stack
        -- Executing [fax at incomingfax:2]
RxFAX("SIP/xxx.xxx.xx.xx-0821ec78", "/tmp/recievedfax.tif") in new stack
    saster*CLI>

and that is where it just sits. no further messages.
the target file does not exist and the directory should have acceptable
rights.

    ls -ld /tmp
    drwxrwxrwt 10 root root 568 2006-11-20 10:47 /tmp/

the fax machine just sits there sending the fax beeps for about 30
seconds before timing out. when i call from a regular phone all i get is
silence.

I also tried initiating a fax to the fax machine.
converted a pdf to /tmp/test.tif
chmod 666 /tmp/test.tif
created a fax.call and moved it into /var/spool/asterisk/outgoing/
that successfully initiated the call and txfax app.
however the txfax still did nothing
I tried a  loop back to the asterisk server with the basic same results.
here is the console log:

    -- Attempting call on SIP/sipfaxline/(asteriskfaxnumber) for
out_fax at outgoingfax:1 (Retry 1)
    -- Executing [(asteriskfaxnumber)@dids-inbound:1]
Goto("SIP/xxx.xxx.xx.xx-08237608", "incomingfax|fax|1") in new stack
    -- Goto (incomingfax,fax,1)
    -- Executing [fax at incomingfax:1] Set("SIP/xxx.xxx.xx.xx-08237608",
"FAXFILE=/tmp/recievedfax.tif") in new stack
    -- Executing [fax at incomingfax:2] RxFAX("SIP/xxx.xxx.xx.xx-08237608",
"/tmp/recievedfax.tif") in new stack
[Nov 20 14:00:57] NOTICE[19864]: pbx_spool.c:341 attempt_thread: Call
failed to go through, reason 0
    -- Attempting call on SIP/sipfaxline/(asteriskfaxnumber) for
out_fax at outgoingfax:1 (Retry 2)
    -- Executing [(asteriskfaxnumber)@dids-inbound:1]
Goto("SIP/xxx.xxx.xx.xx-0824ddf8", "incomingfax|fax|1") in new stack
    -- Goto (incomingfax,fax,1)
    -- Executing [fax at incomingfax:1] Set("SIP/xxx.xxx.xx.xx-0824ddf8",
"FAXFILE=/tmp/recievedfax.tif") in new stack
    -- Executing [fax at incomingfax:2] RxFAX("SIP/xxx.xxx.xx.xx-0824ddf8",
"/tmp/recievedfax.tif") in new stack
[Nov 20 14:01:13] NOTICE[19869]: pbx_spool.c:341 attempt_thread: Call
failed to go through, reason 0
saster*CLI>

the sip.conf file has the relivent setting for the connection being used:

    [sipfaxline]
    type=friend
    nat=never
    host=xxx.xxx.xx.xx
    disallow=all
    allow=ulaw
    dtmfmode=auto
    context=dids-inbound
    canreinvite=yes
    nat=never

I do not have any hardware phone or audio cards in the machine. I don't
think this should matter from what I've read to at least get some
results. Once I prove the concept I'll get some hardware for quality
purposes.

I had a real had time trying to find how to patch asterisk 1.4 for
spandsp until I finally ran across the files that I listed above that I
downloaded.
Hopefully this post will also help others that are having a hard time
just getting everything compiled, as I did!!!!

I've done quite a bit of reading of posts but this is my first time posting.
Thanks in advance for any ideas.

Dave