[asterisk-bugs] [Asterisk 0017372]: [patch] [regression] Progress in band error (don't send RTP packets)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17372 
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Reported By:                tech_admin
Assigned To:                jpeeler
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Project:                    Asterisk
Issue ID:                   17372
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Target Version:             1.6.2.11
Asterisk Version:           1.6.2.7 
JIRA:                       SWP-1526 
Regression:                 Yes 
Reviewboard Link:           https://reviewboard.asterisk.org/r/705/ 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2010-05-21 09:31 CDT
Last Modified:              2010-07-12 12:26 CDT
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Summary:                    [patch] [regression] Progress in band error (don't
send RTP packets)
Description: 
Dear all,

on a updated redhat rhel5 (kernel: 2.6.18-194.3.1.el5xen), I installed the
last asterisk release (1.6.2.7) from the source code.

The problem is: SIP caller doesn't hear ringing tone.
We use progressinband=yes with ch (Switzerland) tones and voice is G729
encoded using TC400B Digium card.
After answer, all is ok: both side RTP packets go trough asterisk without
translation.
The Digium card work fine with others service like IVR, Queues, Voicemail
... all using G729.


======================================================================
Relationships       ID      Summary
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related to          0017185 [patch] [regression] Using Local channe...
has duplicate       0017602 No ringback tone generated
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 (0124514) svnbot (reporter) - 2010-07-12 12:26
 https://issues.asterisk.org/view.php?id=17372#c124514 
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Repository: asterisk
Revision: 275706

_U  branches/1.6.2/
U   branches/1.6.2/main/channel.c

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r275706 | jpeeler | 2010-07-12 12:26:09 -0500 (Mon, 12 Jul 2010) | 25
lines

Merged revisions 275682 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010) | 18
lines
  
  Merged revisions 275665 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) | 11
lines
    
    Change ast_write to not stop generator when called from ast_prod.
    
    For SIP channels configured with the progressinband option on, the
ringback was
    being immediately stopped. This problem was due to ast_prod being
moved for a
    deadlock fix in 259858. Prodding the channel after setting up the
generator
    triggered the check in ast_write to stop the generator. The fix here
should
    write the frame the same as was done before the call to ast_prod was
moved.
    
    (closes issue https://issues.asterisk.org/view.php?id=17372)
    Reported by: tech_admin
  ........
................

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http://svn.digium.com/view/asterisk?view=rev&revision=275706 

Issue History 
Date Modified    Username       Field                    Change               
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2010-07-12 12:26 svnbot         Checkin                                      
2010-07-12 12:26 svnbot         Note Added: 0124514                          
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