[asterisk-bugs] [Asterisk 0016750]: Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jun 8 18:40:05 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16750
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Reported By: iskatel
Assigned To:
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Project: Asterisk
Issue ID: 16750
Category: Applications/app_fax
Reproducibility: always
Severity: major
Priority: normal
Status: acknowledged
Asterisk Version: 1.6.2.1
JIRA: SWP-839
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 01:32 CST
Last Modified: 2010-06-08 18:40 CDT
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Summary: Audio loop reports T38 switchover but t38state !=
T38_STATE_NEGOTIATED
Description:
SendFax not working with t38
Path of call
Asterisk(1.1.1.1) -> Cisco AS5300(2.2.2.2) -> PSTN
app_fax.c was applied from
http://svnview.digium.com/svn/asterisk/branches/1.6.2?view=revision&revision=225871
After re-INVITE has been received from Cisco, asterisk sends 200 OK, then
ERROR appears and asterisk breaks the call. ReceiveFax works.
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(0123154) alexr1 (reporter) - 2010-06-08 18:40
https://issues.asterisk.org/view.php?id=16750#c123154
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Doesn't seem to be entirely related, but while trying to troubleshoot my
problem (above), I changed the codecs on the trunk between the servers to
just "ulaw".
This morning I noticed normal faxes through the T38 Gateway -> AST1.4 ->
AST 1.6 (and vice versa) were no longer working! I added "alaw" back in to
both servers SIP.conf and then it began working. Without the "alaw" it
still went into T38 switchover, but had errors like "Disconnected after
permitted number of retries". Could this be related? When I see "Settling
with this capability (nothing)" (for my original issue), it makes me think
its codec related...
Issue History
Date Modified Username Field Change
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2010-06-08 18:40 alexr1 Note Added: 0123154
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