[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 05:41:37 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 05:41 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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 (0116805) msetim (reporter) - 2010-01-18 05:41
 https://issues.asterisk.org/view.php?id=16594#c116805 
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(IMHO) 

In the meantime, I ask myself why five authors A. Johnston, S. Donovan, R.
Sparks, C. Cunningham, K. Summers accepted that flow. I don't know in deep
about ISDN and SIP interworking however, the document "contains best
current practice examples of Session  Initiation Protocol (SIP) call flows
showing interworking with the Public Switched Telephone Network (PSTN)"
(PAGE 1).

I'm not saying that either wimpy or denisgalvao are wrong (your arguments
really make sense) but I'm bring a new perspective trying to understand why
they wrote that flow. I don't found a RFC talking about interworking
between SIP/ISDN. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-18 05:41 msetim         Note Added: 0116805                          
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