[asterisk-bugs] [Asterisk 0018798]: call drop aleatory and aparently not a irq issue

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Feb 11 13:27:10 CST 2011


The following issue has been SUBMITTED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18798 
====================================================================== 
Reported By:                agustina
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18798
Category:                   Channels/General
Reproducibility:            unable to reproduce
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           Older 1.6.2 - please test a newer version 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-02-11 13:27 CST
Last Modified:              2011-02-11 13:27 CST
====================================================================== 
Summary:                    call drop aleatory and aparently not a irq issue
Description: 
I have the following arquitecture:
Asterisk with 2 E1, both are PCI express, so there wouldn`t be IRQ
issues.

I have my Asterisk 1.6.2.13 conected to a switch and my softphones and
hardophones connected to another switch.

What happens is that calls drop suddenly ramdomly. We have some sound
quality issues but don`t see errors other than some:

channel.c: Dropping incompatible voice frame on SIP/XXXX-0000326f of
format alaw since our native format has changed to 0x4 (ulaw), 
but this messages happen at all times and not when the call drop happens.

We are suspecting some problem with SIP protocol.

We are ataching full and the part where the call hangups.










====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-11 13:27 agustina       New Issue                                    
2011-02-11 13:27 agustina       Asterisk Version          => Older 1.6.2 -
please test a newer version
2011-02-11 13:27 agustina       Regression                => No              
2011-02-11 13:27 agustina       SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================




More information about the asterisk-bugs mailing list