[asterisk-bugs] [Asterisk 0014849]: [patch] SendFax function not working as expected on > 1.6.0.7
Asterisk Bug Tracker
noreply at bugs.digium.com
Sat May 30 04:02:50 CDT 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=14849
======================================================================
Reported By: afosorio
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 14849
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: block
Priority: normal
Status: new
Target Version: 1.6.0.10
Asterisk Version: 1.6.0.7
Regression: Yes
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-04-07 15:51 CDT
Last Modified: 2009-05-30 04:02 CDT
======================================================================
Summary: [patch] 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.
======================================================================
----------------------------------------------------------------------
(0105783) andrew (reporter) - 2009-05-30 04:02
https://issues.asterisk.org/view.php?id=14849#c105783
----------------------------------------------------------------------
With the patch I get an error at the end of the fax, but it works. It might
just be the end of call hangup being reported as an error. I'm using a
Cisco AS5300 gateway and PRI (send and receive).
If T38 is not enabled correctly in sip.conf, but supported I will get the
following error: "Audio loop reports T38 switchover but t38state !=
T38_STATE_NEGOTIATED". It should use the audio codec rather than fail.
Issue History
Date Modified Username Field Change
======================================================================
2009-05-30 04:02 andrew Note Added: 0105783
======================================================================
More information about the asterisk-bugs
mailing list