[asterisk-bugs] [Asterisk 0016756]: app_fax doesn't receive fax with T.38

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Feb 5 04:31:02 CST 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16756 
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Reported By:                viniciusfontes
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16756
Category:                   Applications/app_fax
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                       SWP-845 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-02-02 13:56 CST
Last Modified:              2010-02-05 04:31 CST
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Summary:                    app_fax doesn't receive fax with T.38
Description: 
When I try to receive a fax from a provider that supports T.38 using
ReceiveFAX() provided by app_fax.so, the fax transmission ends prematurely
with an error.

What happens (according to Kevin Fleming) is this:

"In this call, Asterisk sent a re-INVITE to T.38 mode
from audio mode, the provider accepted it, and then Asterisk
acknowledged it. Immediately afterwards, Asterisk sent a re-INVITE
*back* to audio mode, which the provider accepted (and included T.38
capabilities in their response). Because of this, the FAX reception
process failed since the T.38 session was destroyed."

For reference, here is a link to the original mailing list thread: 
http://lists.digium.com/pipermail/asterisk-users/2010-February/244136.html
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---------------------------------------------------------------------- 
 (0117772) viniciusfontes (reporter) - 2010-02-05 04:31
 https://issues.asterisk.org/view.php?id=16756#c117772 
---------------------------------------------------------------------- 
Still having very low bitrates even with maxdatagramsize=180. I attached a
CLI session with "sip set debug peer voxip on". My relevant sip.conf
section is as follows:

[voxip]
username=5421047000
nat=no
type=peer
secret=5421047000
port=5060
canreinvite=no
insecure=port,invite
host=10.150.65.16
fromuser=5421047000
fromdomain=gvt.com.br
dtmfmode=rfc2833
context=entrada-e1
disallow=all
allow=g729,alaw
qualify=no
;t38pt_udptl=yes
t38pt_udptl = yes,maxdatagram=180


I might be wrong, but I didn't see Asterisk trying to negotiate the
maxdatagram size to 180 not even once. It just tries to negotiate it as
1400, which the provider doesn't accept and falls back to 612. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-02-05 04:31 viniciusfontes Note Added: 0117772                          
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