[asterisk-bugs] [Asterisk 0017372]: Progress in band error (don't send RTP packets)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17372 
====================================================================== 
Reported By:                tech_admin
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17372
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.7 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-05-21 09:31 CDT
Last Modified:              2010-05-24 14:52 CDT
====================================================================== 
Summary:                    Progress in band error (don't send RTP packets)
Description: 
Dear all,

on a updated redhat rhel5 (kernel: 2.6.18-194.3.1.el5xen), I installed the
last asterisk release (1.6.2.7) from the source code.

The problem is: SIP caller doesn't hear ringing tone.
We use progressinband=yes with ch (Switzerland) tones and voice is G729
encoded using TC400B Digium card.
After answer, all is ok: both side RTP packets go trough asterisk without
translation.
The Digium card work fine with others service like IVR, Queues, Voicemail
... all using G729.


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

---------------------------------------------------------------------- 
 (0122330) pabelanger (manager) - 2010-05-24 14:52
 https://issues.asterisk.org/view.php?id=17372#c122330 
---------------------------------------------------------------------- 
We need a 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:

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt


Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-05-24 14:52 pabelanger     Note Added: 0122330                          
======================================================================