[asterisk-bugs] [Asterisk 0013034]: 183 response although progressinband=never

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13034 
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Reported By:                klaus3000
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13034
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.21 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-07-09 07:03 CDT
Last Modified:              2009-02-20 02:35 CST
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Summary:                    183 response although progressinband=never
Description: 
Hi!

Scenario with Asterisk 1.4.21.1:

SIP Client ----> chan_sip:Dial(zap):chan_zap ---> ISDN

very simple dialplan:
[fromklaus]
exten => _1X.,1,NoOp(1... SIP: Outgoing Call: Asterisk->HiCom)
exten => _1X.,n,Dial(Zap/g1/${EXTEN:1})

Immediately after sending the SETUP message, Asterisk responds with 183
Session Progress. Thus, the SIP client is waiting for inband audio, but
there is no inband audio available. progressinband=never

sip.conf:

[klaus]
type=peer
username=klaus
host=dynamic
context=fromklaus
canreinvite=no
progressinband=never


actually I tried all progressinband settings without any difference
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---------------------------------------------------------------------- 
 (0100431) oej (manager) - 2009-02-20 02:35
 http://bugs.digium.com/view.php?id=13034#c100431 
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After a lot of discussion, I've decided to create a branch called
"no-premature-183" with an option to disable this behaviour in chan_sip. We
won't send any frames until we actually have proceeding or alerting from
the other end. I don't know why chan_sip does that, but doesn't like
breaking it.

This could also be an option in zaptel/dahdi so that no frames are sent
INTO your pbx, but that's for someone else to patch and it would propably
be a better solution. My world is chan_sip :-)

We still want to figure out WHY this happens, what kind of frames
equipment send. 

See
http://lists.digium.com/pipermail/asterisk-commits/2009-February/031041.html 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-20 02:35 oej            Note Added: 0100431                          
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