[asterisk-bugs] [Asterisk 0015475]: AGI Script dialparties.agi returned error: Broken Pipe

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15475 
====================================================================== 
Reported By:                Bateleur
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   15475
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           Addons 1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-07-09 09:35 CDT
Last Modified:              2009-07-09 09:35 CDT
====================================================================== 
Summary:                    AGI Script dialparties.agi returned error: Broken
Pipe
Description: 
Not sure if this is the correct category ...

Running Trixbox 2.8 with Asterisk 1.6.  It has been up and running for a
few weeks now and has been running fairly succesfully.  As of yesterday,
any internal extension dialed fails with the device displaying
"Declined"... it doesn't even route to voicemail.  We are thus unable to
reach any extension.  We are still able to dial externally via DAHDI or via
VOIP trunks.  We can dial queues directly and incomming calls are still
routed via IVR to queues.  The agents extensions within those queues
however never ring and as such we loose the call.

The only error I can find in the log files is the broken pipe error.  The
dialplan.agi error occurs every single time thus making it impossible to
dial an internal extention.  The recordingcheck error also occurs every
time but the results (I don't think) are as crippling as the inability to
dial.

The only change we made to our config on the day was to add a new IAX2
trunk for inbound VOIP calls.
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-09 09:35 Bateleur       New Issue                                    
2009-07-09 09:35 Bateleur       Asterisk Version          => Addons 1.6.0    
2009-07-09 09:35 Bateleur       Regression                => No              
2009-07-09 09:35 Bateleur       SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================