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

Asterisk Bug Tracker noreply at bugs.digium.com
Sun Mar 29 16:14:08 CDT 2009


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:                tilghman
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Project:                    Asterisk
Issue ID:                   13034
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
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:              2009-03-29 16:14 CDT
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Summary:                    [patch] 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
====================================================================== 

---------------------------------------------------------------------- 
 (0102361) klaus3000 (reporter) - 2009-03-29 16:14
 http://bugs.digium.com/view.php?id=13034#c102361 
---------------------------------------------------------------------- 
Take a look at the "CALL PROCEEDING" - if it contains PROGRESS IE with
"inband information available" then it is necessary to start early media
(183) and the ringback should be generated by the remote end (callee
switch) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-29 16:14 klaus3000      Note Added: 0102361                          
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