[asterisk-bugs] [Asterisk 0016594]: ISDN to SIP doesn't generate SIP 180 Ringing with Call Progress ISDN message
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jan 19 06:26:54 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16594
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Reported By: denisgalvao
Assigned To:
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Project: Asterisk
Issue ID: 16594
Category: Core/Channels
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: 1.6.1.12
JIRA: SWP-718
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: 2010-01-13 10:57 CST
Last Modified: 2010-01-19 06:26 CST
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Summary: ISDN to SIP doesn't generate SIP 180 Ringing with
Call Progress ISDN message
Description:
When I place a call trough a SIP UA and this call goes out through an ISDN
link, the Call Progress message from ISDN is not converted to a SIP 180
Ringing message to the UA.
P.S.: Im not sure about the category of this issue, BTW I believe it is a
channel related thing.
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(0116881) denisgalvao (reporter) - 2010-01-19 06:26
https://issues.asterisk.org/view.php?id=16594#c116881
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I(actually not me) have a business rule on my application that analyze all
180 Ringing messages to do an action based on that SIP message
Ok, this could be changed, I thnk the same, but the person behind this
application doesn't like the idea, so I start to learn about this
internetworking call flows and found that doc.
I don't want to change the Asterisk behaviour based on my needs, but based
on the best practices
.
Could we have some thoughts from an Asterisk developer here?
Issue History
Date Modified Username Field Change
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2010-01-19 06:26 denisgalvao Note Added: 0116881
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