[asterisk-bugs] [Asterisk 0010934]: [patch] Option r with early files

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10934 
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Reported By:                gasparz
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10934
Category:                   Applications/NewFeature
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0-beta5 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             10-10-2007 07:22 CDT
Last Modified:              06-06-2008 05:31 CDT
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Summary:                    [patch] Option r with early files
Description: 
I was asked to develop a feature thar switches between the ringing tone and
early media. So when you dial asterisk starts generating ringing tone, but
when the called channels starts sendig RTP (like mobile carriers: “The
person you are trying to reach is unavailable") with or without answering
the channel, the asterisk would have to end generating the ringing tone and
send the media it receives.

The patch does this.
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---------------------------------------------------------------------- 
 klaus3000 - 06-06-08 05:31  
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@qwell: Hi! I often have similar problems when interconnecting PBXs using
Asterisk, e.g:

site1:
PSTN----->Asterisk1<----ISDN------>PBX1
             ^
             | SIP or
             | IAX
site2:       V
PSTN----->Asterisk2<----ISDN------>PBX2

Now an extension of PBX1 calls an extension of PBX2 with PSTN bypass using
SIP or IAX. The PBX2 sends CALL PROCEEDING and ALERTING to Asterisk2
without any Progress Indicator. But strangly Asterisk1 sends CALL
PROCEEDING and ALERTING to PBX1 with Progress Indicator and "inband
information available".

Thus, the user1 das not get ringback as PBX2 does not generate inband
audio. Probably this is a bug somewhere else in Asterisks code, but could
be manually fixed with 'r' option too, except that real inband audio is not
anymore possible. 

Issue History 
Date Modified   Username       Field                    Change               
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06-06-08 05:31  klaus3000      Note Added: 0087882                          
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