[asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Larry Moore lmoore at omninet.net.au
Mon Feb 3 07:48:21 CST 2014

On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote:
> Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger:
>> Am 03.02.2014 12:56, schrieb Larry Moore:
>>> On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:
>>>> as He is describing it he should actually provide t.38. but i don't
>>>> know
>>>> why it is not working thus im now getting
>>>> Feb 3 12:32:55] WARNING[9942][C-00000004]: chan_sip.c:10353
>>>> process_sdp:
>>>> Failed to initialize UDPTL, declining image stream
>>>> [Feb 3 12:32:55] WARNING[9942][C-00000004]: chan_sip.c:10497
>>>> process_sdp: Insufficient information in SDP (c=)...
>>>> and then the fax session starts recording data
>>> In udptl.conf set use_even_ports=yes and then issue a reload.
>>> You can confirm the settings have been applied by performing udptl
>>> show config.
>>> Change the the t38 line to read as;
>>> t38pt_udptl=yes,redundancy,maxdatagram=400
>>> Reload sip and test.
>> after that i started udptl debug as well and now i'm getting lots of
>> UDPTL (SIP/sipcall.ch-00000007): packet to (seq
>> 152, len 11)
>> and in between
>> [Feb 3 13:18:30] WARNING[26313][C-00000007]: res_rtp_asterisk.c:3548
>> ast_rtp_read: RTP Read too short
>> and in the end
>> [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct:
>> Autodestruct on dialog
>> '24d15e0d-28df847a-9fae13c-7ace at sip.iforb.com~1o' with owner
>> SIP/sipcall.ch-00000007 in place (Method: BYE). Rescheduling
>> destruction for 10000 ms
>> [Feb 3 13:18:41] ERROR[26313][C-00000007]: res_fax.c:1535
>> generic_fax_exec: channel 'SIP/sipcall.ch-00000007' FAX session '7'
>> failure, reason: 'fax session timed-out' (TIMEOUT)
>> == Spawn extension (fax-rx, receive, 11) exited non-zero on
>> 'SIP/sipcall.ch-00000007'
>> Thx, Jakob
> may do i have to open more ports then udp 10000:20000 (RTP), udp
> 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS)

The T.38 connection will be attempted when ReceiveFax is called.

The port number to use should be in the SDP information, yes, allow udp 
ports 4000-4999 in and out. If your firewall can be so configured you 
could set it to allow traffic in and out based upon the user ID Asterisk 
is running as, assuming it is using a unique unprivileged id.

You may like to try the following to see if your SIP provider will 
initiate a T.38 re-invite.




In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a 
T.38 re-invite this will trigger the switch to the Fax extension.

If this proves successful you can work on removing the Wait() from your 
dialplan as Asterisk will remain in the audio path and should 
successfully switch to the fax extension if extension 200 or 201 answer 
a call that happens to be a fax.


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