[asterisk-bugs] [Asterisk 0014849]: SendFax function not working as expected on > 1.6.0.7
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri May 1 06:17:41 CDT 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=14849
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Reported By: afosorio
Assigned To:
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Project: Asterisk
Issue ID: 14849
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: feature
Priority: normal
Status: new
Asterisk Version: 1.6.0.7
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-04-07 15:51 CDT
Last Modified: 2009-05-01 06:17 CDT
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Summary: SendFax function not working as expected on >
1.6.0.7
Description:
When trying to send a fax via callfile using T.38 in asterisk version
1.6.0.5 it works fine, but after upgrading to asterisk 1.6.0.7 or newer it
does not work. When receiving re-INVITE from media gateway (Cisco AS-5400)
Asterisk ends call answering 488 Not acceptable here. I am running Asterisk
under Debian kernel 2.6.18-6-686. I am attaching captures both for scenario
which works and scenario which does not work.
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(0104017) vrban (reporter) - 2009-05-01 06:17
http://bugs.digium.com/view.php?id=14849#c104017
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@all have you tried to SendFax from one asterisk to the ReceiveFAX of an
other asterisk? I have done it to encircle this issue, with curious
result:
*s = asterisk 1, SendFax
*r = asterisk 2, ReceiveFAX
1. *s send it's INVITE to *r. without T.38 in sdp, what correct, because
the receiver *r shall re-invite with T.38
2. *r now send it's re-invite with T.38 to s* (after: app_fax.c:469
transmit_audio: Fax tone detected. Requesting T38) and *s send a "100
Trying" to this T.38 re-invite.
3. Till now all is ok. But now the curious part begin:
Right after the "100 Trying" from *s the SendFax and ReceiveFAX between *s
and *r start to transmit the fax with audio T.30 (not T.38!), and before *s
send the "488 Not acceptable here", the whole fax is already and _complete_
transferred with T.30!?
4. After the fax is transfered, *s and *r send some "503 Unavailable" back
and forth, because the SIP transmission is also messed up then.
can someone reproduce this?
@gentian
i tested with 1.6.2-svn, but this behave should be identical with all
versions after ast_1.6.0.5
Issue History
Date Modified Username Field Change
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2009-05-01 06:17 vrban Note Added: 0104017
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