[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
Mon Jan 18 14:45:59 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: 2010-01-18 14:45 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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(0116847) wimpy (reporter) - 2010-01-18 14:45
https://issues.asterisk.org/view.php?id=16594#c116847
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I haven't got a clue as to why someone would want to translate the ISDN
messages to anything else than their SIP equivalents. Atfter all they
translate very well.
So I don't think Asterisk is doing anything wrong here. It looks
perfectely reasonable to me.
What is it you're calling that's answering without ringing and why do you
need ringing?
Would option r to dial be a solution to your problem?
Issue History
Date Modified Username Field Change
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2010-01-18 14:45 wimpy Note Added: 0116847
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