[asterisk-bugs] [Asterisk 0013036]: Caller gets dropped and engaged tone after two Zap channels bridged.

noreply at bugs.digium.com noreply at bugs.digium.com
Wed Jul 9 09:22:01 CDT 2008


The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=13036 
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Reported By:                ruffle
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13036
Category:                   Core/Channels
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21 
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:             07-09-2008 09:21 CDT
Last Modified:              07-09-2008 09:21 CDT
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Summary:                    Caller gets dropped and engaged tone after two Zap
channels bridged.
Description: 
I have an ISDN PBX (Network Alchemy Argent Office) on Span 2 of a Wildcard
TE205P with a British Telecom ISDN 30 on Span 1.  This configuration has
been working well for ages and nothing has changed on the ISDN PBX.

Using Asterisk 1.4.21.1 and Zaptel-1.4.11 when a call is made from the
ISDN PBX the caller hears ring tone, the called party answers and the
caller then hears the busy tone and the called party hears nothing. The
channels remain bridged for a couple of minutes. Here's the output of show
channels when it's in this state:

asterisk*CLI> show channels
Channel              Location             State   Application(Data)
Zap/2-1              (None)               Up      Bridged Call(Zap/46-1)
Zap/46-1             01780XXXX81 at alchemy: Up     
Dial(Zap/g1/01780XXXX81|9999)
2 active channels
1 active calls
asterisk*CLI>

This doesn't happen to all of the calls.... just most of them :-(

I have attached the verbose=9 logging output of a failed call and my
zaptel.conf (itself unchanged for ages).

I'm happy to turn on whatever debugging is needed, provide traces/output
and apply patches if needed.
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Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-09-08 09:22  ruffle         Asterisk Version          => 1.4.21          
07-09-08 09:22  ruffle         SVN Branch (only for SVN checkou => N/A          
  
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