[asterisk-bugs] [Asterisk 0014807]: SendFax and T38 issues
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jun 16 17:23:01 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=14807
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Reported By: moliveras
Assigned To: file
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Project: Asterisk
Issue ID: 14807
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: minor
Priority: normal
Status: assigned
Asterisk Version: 1.6.2.0-beta1
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-04-01 16:06 CDT
Last Modified: 2009-06-16 17:23 CDT
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Summary: SendFax and T38 issues
Description:
I am trying to get SendFax to work using a call file.
I initially tried it with asterisk-1.6.0.3 (spandsp-0.0.5pre4).
When I sent a call to a local spa2102 TA, the far end sent a re-invite to
switch over to T38, and the call went through as T38.
asterisk1.6.0.3 --> SBC --> ast1.4 --> spa2102
When I sent the call out to a land line via a carrier (SIP trunk),
asterisk receives the T38 reinvite, however the 200 OK that it sends back
only has g711 as a codec choice, and not udptl t38. As a result, the fax
failes.
asterisk1.6.0.3 --> SBC --> SIP trunk
I then tried upgrading to asterisk-1.6.2.0-beta1 (spandsp-0.0.6pre7).
Now when I make the call to the land line over the SIP trunk the behavior
is the same, however when I call the spa2102 line, ast1.6.2.0-beta1 rejects
the SIP reinvite with a 488, and the fax completes inband.
Also, the 1.6.2.0-beta1 spits out these errors while the fax in in
progress:
[Apr 1 13:53:41] ERROR[10228]: channel.c:2539 __ast_read: ast_read()
called with no recorded file descriptor.
I will attach packet captures for these scenarios momentarily. I can
reproduce this behavior consistently and can provide whatever info/debug is
needed.
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(0106524) moliveras (reporter) - 2009-06-16 17:23
https://issues.asterisk.org/view.php?id=14807#c106524
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I spent time looking at this today, and believe that I found the problem.
When the t38 reInvite from SIP carrier A is sent to asterisk, it changes
the SDP, however it does not increment the SDP version ID. The captures
for the cases where it works do increment the session version. I also was
able to reproduce this by creating a sipp script based on the capture from
carrier A; as soon as I change the session version on the reInvite,
asterisk lists t38 in the 200 OK.
Since this is a problem with the SIP carrier and not asterisk, this ticket
can be closed.
Issue History
Date Modified Username Field Change
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2009-06-16 17:23 moliveras Note Added: 0106524
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