asterisk users - Oct 2011 - Digium FFA + Gafachi T38 outgoing issues

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

Check firewall and NAT settings!

Monitoring sip and media flow from asterisk cli can help, use "sip set
debug
on", "rtp set debug on" and "udptl set debug on"

Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/


On Fri, Oct 7, 2011 at 1:37 AM, James Sharp <james at fivecats.org> wrote:
> 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
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> 
http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
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On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
> Check firewall and NAT settings!
>
> Monitoring sip and media flow from asterisk cli can help, use "sip set
> debug on", "rtp set debug on" and "udptl set debug
on"
>
No NAT involved and I shut off IPTables.  Still no luck.  Debug shows 
the SIP invite, RTP frames going in & out, the SIP reinvite, and then 
UDPTL frames coming in until timeout.

See the entire transaction at http://pastebin.ca/2087758

for which user/number sip reinvite is for? ooh! you are running a direct
application without any dialplan or user, may be that is the cause!  I think
you should first write fax dialplan, reload asterisk and test again with
originate but this time with extension not application.

Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/



On Sat, Oct 8, 2011 at 12:20 AM, James Sharp <james at fivecats.org>
wrote:
> On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
>
>> Check firewall and NAT settings!
>>
>> Monitoring sip and media flow from asterisk cli can help, use "sip
set
>> debug on", "rtp set debug on" and "udptl set debug
on"
>>
>>
> No NAT involved and I shut off IPTables.  Still no luck.  Debug shows the
> SIP invite, RTP frames going in & out, the SIP reinvite, and then UDPTL
> frames coming in until timeout.
>
> See the entire transaction at http://pastebin.ca/2087758
>
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> 
http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
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