[asterisk-bugs] [Asterisk 0019268]: pri early media won't ring

Asterisk Bug Tracker noreply at bugs.digium.com
Thu May 12 11:16:44 CDT 2011


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19268 
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Reported By:                tbsky
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19268
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.8 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-05-10 22:12 CDT
Last Modified:              2011-05-12 11:16 CDT
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Summary:                    pri early media won't ring
Description: 
hi:
   our asterisk connection like below:
   <sip or iax phone>---<asterisk>---<alcatel pbx>---<PSTN>

asterisk connect to alcatel pbx via E1 isdn-pri
 
   when sip phone (snom 300) or iax phone (zoiper) call PSTN, they can not
hear the ring tone and the beginning of the PSTN voice.
   I have checked issue 18868 and tried below combination with no luck.

   1.8.3.2 + sig_pri patch
   1.8 svn which already has sig_pri patched
   1.8.4 + libpri patch (another unofficial patch in issue 18868)
  
   I downgrade to 1.6.2.18 and I found everything works fine.
zoiper iax is ok and after setting "prematuremedia=no",snom sip phone is
fine too.
 
   I don't what other options to try. or what debug information to
provide. please tell me if necessary.
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---------------------------------------------------------------------- 
 (0134842) tbsky (reporter) - 2011-05-12 11:16
 https://issues.asterisk.org/view.php?id=19268#c134842 
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hi:
   I attach two logs. 1.6.2.18 is working and SVN-1.8 is not working(can
not hear ring and the beginning of remote voice).

   I have "prematuremedia=no" in sip.conf and use zoiper sip as client for
testing.
  
   hope the log helps!! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-12 11:16 tbsky          Note Added: 0134842                          
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