[asterisk-bugs] [Asterisk 0018399]: Call torn down upon connection when early media 183 used

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Nov 30 21:40:51 CST 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18399 
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Reported By:                eeman
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18399
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.1-rc1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-11-29 15:27 CST
Last Modified:              2010-11-30 21:40 CST
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Summary:                    Call torn down upon connection when early media 183
used
Description: 
Asterisk 1.8.1-rc1 & Asterisk 1.6.2.14
Centos 5.5

have scenario as such

Asterisk-1.8.1-rc1 -SIP-> Asterisk 1.6.2.14 -SIP-> Broadvox (Sonus
Softswitch)

When calling a TF number that uses early media for their IVR (example
1-800-626-2001); once the call gets connected and the 200 OK message is
received, my 1.8.1-rc1 box issues a BYE message with a HangupCauseCode of
0. I can reproduce this with several numbers that are using early media for
their IVR's. Just as soon as my call gets connected to a call-center's ACD
Queue I hear 1-2 seconds of the recording before the call is torn down. I
have tested this using a linksys SPA-2102 ATA, A Polycom IP501, as well as
a Digium FXS module and get identical results. 
====================================================================== 

---------------------------------------------------------------------- 
 (0129227) pabelanger (administrator) - 2010-11-30 21:40
 https://issues.asterisk.org/view.php?id=18399#c129227 
---------------------------------------------------------------------- 
Please attach a full debug log (see below)
---
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs
from an Asterisk machine for the purpose of helping bug marshals
troubleshoot an issue:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-30 21:40 pabelanger     Note Added: 0129227                          
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