[asterisk-bugs] [Asterisk 0011836]: Create random audio failure in Asterisk

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11836 
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Reported By:                tootai
Assigned To:                crich
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Project:                    Asterisk
Issue ID:                   11836
Category:                   Channels/chan_misdn
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 99777 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             01-24-2008 11:41 CST
Last Modified:              05-03-2008 14:19 CDT
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Summary:                    Create random audio failure in Asterisk
Description: 
When incoming calls from ISDN -Etch with 2 BP410P- join conferences
(meetme) they make the audio disappear on ALL services (other conferences,
voicemails, ...) for all devices. Calls are still passing OK between
devices, which are SIP or IAX, including from/to ISDN.

If we do a "misdn show channels", sometimes a channel appear in the list
as marked down, knowing that there is no call related to this channel from
a "core show channels". We then do a "misdn port up <port #>" and thinks
are starting to work again.

If the above manipulation doesn't work, restarting Asterisk make
everything going well again, till next time.

We tested with mISDN from Digium (make B410P) as well as the 1.1.7.2 from
mISDN.org

-- 
Daniel
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---------------------------------------------------------------------- 
 jmls - 05-03-08 14:19  
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is this still a problem with the latest 1.4 svn ? Thanks. 

Issue History 
Date Modified   Username       Field                    Change               
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05-03-08 14:19  jmls           Note Added: 0086356                          
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