[asterisk-bugs] [Asterisk 0015420]: [patch] No audio on calls from asterisk sip phones to nortel set until dtmf from sip phone

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15420 
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Reported By:                scottbmilne
Assigned To:                rmudgett
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Project:                    Asterisk
Issue ID:                   15420
Category:                   General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.25.1 
Regression:                 No 
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:             2009-06-29 13:58 CDT
Last Modified:              2009-07-09 18:56 CDT
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Summary:                    [patch] No audio on calls from asterisk sip phones
to nortel set until dtmf from sip phone
Description: 
When placing a call from an Asterick SIP phone (X-Lite) to Nortel phone set
(M3903), no voice will pass until any key is pressed on the SIP phone. SIP
to SIP calls - normal. SIP to external phone - normal. Nortel to SIP -
normal. All permutations of calls pass audio immediately upon answer except
an sip to nortel.

Asterisk is behind Nortel Option 11c via PRI
Telco -> PRI -> Nortel Option 11c -> PRI -> Asterisk

Very new to Asterisk and VOIP and don't know where to start...Any
suggestions would be GREATLY appreciated.

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Relationships       ID      Summary
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related to          0013034 [patch] 183 response although progressi...
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 (0107549) svnbot (reporter) - 2009-07-09 18:56
 https://issues.asterisk.org/view.php?id=15420#c107549 
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Repository: asterisk
Revision: 205731

U   branches/1.6.2/channels/chan_dahdi.c

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r205731 | rmudgett | 2009-07-09 18:56:21 -0500 (Thu, 09 Jul 2009) | 28
lines

Merged revisions 205728 via svn merge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21
lines
  
  No audio on calls from Asterisk to various ISDN devices until DTMF sent
by caller.
  
  Add missing clearing of the dialing flag when the ISDN call is
CONNECTED.
  (i.e. When libpri generates the event PRI_EVENT_ANSWER.)
  
  (closes issue https://issues.asterisk.org/view.php?id=15420)
  Reported by: scottbmilne
  Patches:
        bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
  Tested by: scottbmilne, alecdavis
  
  (closes issue https://issues.asterisk.org/view.php?id=15416)
  Reported by: avinoash
  
  (closes issue https://issues.asterisk.org/view.php?id=15389)
  Reported by: alecdavis
  
  This patch should also fix the following issue:
  (issue https://issues.asterisk.org/view.php?id=15205)
  Reported by: vinsik
........

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http://svn.digium.com/view/asterisk?view=rev&revision=205731 

Issue History 
Date Modified    Username       Field                    Change               
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2009-07-09 18:56 svnbot         Checkin                                      
2009-07-09 18:56 svnbot         Note Added: 0107549                          
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