[asterisk-bugs] [Asterisk 0017256]: Call gets disconnected after approx 20 seconds of ringing
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Apr 28 08:15:26 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=17256
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Reported By: gblades_skymarket
Assigned To:
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Project: Asterisk
Issue ID: 17256
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.4.30
JIRA:
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: 2010-04-28 07:18 CDT
Last Modified: 2010-04-28 08:15 CDT
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Summary: Call gets disconnected after approx 20 seconds of
ringing
Description:
We have a system where calls come in over ISDN and then get routed out over
SIP.
The SIP carrier gives a SIP/183 response to indicate call progress but
after approx 20 seconds of ringing asterisk disconnects the call. I am not
specifying a duration to the dial command so it should not timeout.
We were running an older version of asterisk (1.4.22) and at that time we
manually modified sip.conf to make it consider a 183 to be the same as a
180 response which fixed this problem but it can cause other issues itself)
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(0121065) davidw (reporter) - 2010-04-28 08:15
https://issues.asterisk.org/view.php?id=17256#c121065
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-- Channel 0/31, span 1 got hangup request, cause 19
This was closed from the other end of the circuit switched connection,
with a NO ANSWER status. Moreover, as the trace shows that:
[Apr 28 12:10:08] DEBUG[22107]: chan_dahdi.c:5547 dahdi_indicate: Received
AST_CONTROL_PROGRESS on Zap/31-1
this has to be a dahdi problem, or remote network problem. To be clear.
Asterisk is not initiating the close; whatever is at the other end of the
ISDN is doing so.
I tend to suspect the remote network, making this a support question, not
one for this bug tracker.
Issue History
Date Modified Username Field Change
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2010-04-28 08:15 davidw Note Added: 0121065
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