[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. 
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https://issues.asterisk.org/view.php?id=17611 
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Reported By:                maxnuv
Assigned To:                
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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:              
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Date Submitted:             2010-07-09 01:34 CDT
Last Modified:              2010-07-12 01:36 CDT
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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...
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 (0124472) maxnuv (reporter) - 2010-07-12 01:36
 https://issues.asterisk.org/view.php?id=17611#c124472 
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The SIP channel was first on alaw, then ulaw, then g729, then gsm, and i
never found error about the format... 

After some test i think the real problem is chan_oss "or" the answer and
the music_on_hold application, the sip channel i think is ok. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-07-12 01:36 maxnuv         Note Added: 0124472                          
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