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

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Jul 9 06:29:33 CDT 2010


The following issue has been UPDATED. 
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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:                     closed
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:              
Resolution:                 suspended
Fixed in Version:           
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Date Submitted:             2010-07-09 01:34 CDT
Last Modified:              2010-07-09 06:29 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...
====================================================================== 

---------------------------------------------------------------------- 
 (0124378) pabelanger (manager) - 2010-07-09 06:29
 https://issues.asterisk.org/view.php?id=17611#c124378 
---------------------------------------------------------------------- 
This is a support request, not a bug.  Please use the support lists.  If
you still suspect this is a bug, enable debug logs and upload the
appropriate information.

---
Thanks for your comments. This does not appear to be a bug report and we
are closing it. We appreciate the difficulties you are facing, but it would
make more sense to raise your question in the support tracker,
http://www.asterisk.org/support 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-07-09 06:29 pabelanger     Note Added: 0124378                          
2010-07-09 06:29 pabelanger     Status                   new => closed       
2010-07-09 06:29 pabelanger     Resolution               open => suspended   
======================================================================




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