[asterisk-bugs] [Asterisk 0017611]: Answer not working, maybe chan_oss problem, may be chan_sip problem.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17611 
====================================================================== 
Reported By:                maxnuv
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17611
Category:                   Channels/chan_oss
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.32 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-07-09 01:34 CDT
Last Modified:              2010-07-12 06:02 CDT
====================================================================== 
Summary:                    Answer not working, maybe chan_oss problem, may be
chan_sip problem.
Description: 
This is the way i do the test.
from console
"dial number at from-sip"

the extension number at from-sip make a call trhu SIP provider
(tested and working from SIP phone).

i answer the "called" phone, i hear nothing (normally i hear something),
this is the first problem.

i transfer the call to the *199 extension from console
"transfer *199 at from-sip"

the *199 extension is "Answer() MusicOnHold()".

I hear nothing.

if i use the *198 extension, all things ok

the *198 extension is "Answer() PlayBack(silence/1) MusicOnHold()"

also if i transfer to another SIP channel no audio, i need to do the
"PlayBack(silence/1)" after the Answer() to hear something.

This is with asterisk 1.4.32 but also with 1.4.26.2 and some other in the
middle.

I checked the SIP debug, i see NO difference between the two calls (the
one transferred to the *199 and the one transferred to the *198) so this is
why i think the problem is chan_oss or console related.

I think the "real" problem is the "first" answer, to the SIP channel, but
something wrong must be also in the Answer application.

The strange is that if i use PlayBack, Voicemail or other apps it works...
====================================================================== 

---------------------------------------------------------------------- 
 (0124483) pabelanger (manager) - 2010-07-12 06:02
 https://issues.asterisk.org/view.php?id=17611#c124483 
---------------------------------------------------------------------- 
As mentioned before, 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               
====================================================================== 
2010-07-12 06:02 pabelanger     Note Added: 0124483                          
======================================================================