[asterisk-users] Digium FFA + Gafachi T38 outgoing issues

James Sharp james at fivecats.org
Thu Oct 6 15:37:12 CDT 2011


Hi, folks.

I'm having a heck of a time trying to get outgoing T38 faxing (I don't 
need inbound right now) working with FFA and Gafachi.  G711 faxing works 
(as well as can be expected over the internet), but I want the higher 
reliability of T38.

I'm running Asterisk 10-beta1.

When I drop my callfile in to make the call, I get this:

     -- Attempting call on SIP/18884732963 at gafachi1a for application 
SendFAX(/srv/httpd/htdocs/upload/scantest2.tiff,dz) (Retry 1)
   == Using UDPTL CoS mark 5
   == Using SIP RTP CoS mark 5
        > Channel SIP/gafachi1a-0000000a was answered.
        > Launching SendFAX(/srv/httpd/htdocs/upload/scantest2.tiff,dz) 
on SIP/gafachi1a-0000000a
     -- Channel 'SIP/gafachi1a-0000000a' sending FAX:
     --    /srv/httpd/htdocs/upload/scantest2.tiff
     -- Channel 'SIP/gafachi1a-0000000a' FAX session '6' started
     -- FAX handle 0: [ 000.000594 ], STAT_EVT_STRT_TX       st: IDLE 
       rt: IDLENSTX
     -- FAX handle 0: [ 000.001139 ], STAT_EVT_TX_HW_RDY     st: 
WT_TX_HW_RDY rt: TRDYNHTY
     -- FAX handle 0: [ 000.001724 ], P30EVN_SEND_STARTED
[Oct  6 04:21:36] ERROR[11616]: res_fax.c:1421 generic_fax_exec: channel 
'SIP/gafachi1a-0000000a' FAX session '6' failure, reason: 'fax session 
timed-out' (TIMEOUT)
[Oct  6 04:21:36] NOTICE[11616]: pbx_spool.c:373 attempt_thread: Call 
completed to SIP/18884732963 at gafachi1a

---- THIS PART HAPPENS ABOUT 15 SECONDS LATER ----

     -- FAX handle 0: [ 040.000211 ], STAT_EVT_T1_EXP        st: WT_DIS 
       rt: WDISNT1X
     -- FAX handle 0: [ 042.499953 ], STAT_EVT_HW_CLOSE      st: 
WT_HW_CLS    rt: WCLSNCLS
     -- FAX handle 0: [ 042.500083 ], STAT_SES_COMPLETE
     -- FAX handle 0: [ 042.500110 ], P30EVN_COMPLETE
     -- Channel 'SIP/gafachi1a-0000000a' FAX session '6' is complete, 
result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, 
resolution: 'unknown', transfer rate: '2400', remoteSID: ''


A tcpdump trace shows the initial invite, ringing, answering, some G711 
frames back and forth, the send-T38-invite-after-10-seconds reinvite (as 
specified by the Z option), then the far end sends a bunch of T38 
traffic until Asterisk times out and drops the call.

What also confuses me is this (and this may just be semantics or a true 
bug):

asterisk*CLI> fax show stats

FAX Statistics:
---------------

Current Sessions     : 0
Reserved Sessions    : 0
Transmit Attempts    : 8
Receive Attempts     : 0
Completed FAXes      : 7
Failed FAXes         : 7


How can I have 8 attempted transmits, 7 completed faxes, and 7 failed 
faxes? I know 1 transmit didn't go through because I tried to place one 
call while another was in progess and I only have one licensed channel.



Thanks,

James Sharp
james at fivecats.org



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