[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:                file
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Project:                    Asterisk
Issue ID:                   13034
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
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:              2009-02-03 14:51 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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---------------------------------------------------------------------- 
 (0099364) oej (manager) - 2009-02-03 14:51
 http://bugs.digium.com/view.php?id=13034#c99364 
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I have a full debug from a system where this happens. I see audio being
sent to the SIP device after the 183, but before the PRI sets the channel
to ALERTING. At alerting the SIP channel sends an 180 Ringing, which is
correct. Unfortunately there's no debug in SIP_WRITE() so I don't see the
types of frames that arrives, I only see ALAW audio frames being sent in
RTP debug, which  leads me to the conclusion that for some reason there's
audio coming in on the PRI side (with an old legacy PBX connected).

The audio comes after PRI debug says that call enters state 1 (call
initiated). The next PRI even seems to be CALL PROCEEDING, but at that
time, we've already sent a lot of RTP packets.

I am not allowed to apply and changes to this system and can't release the
log file for public access, as this is a production system. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-03 14:51 oej            Note Added: 0099364                          
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