[asterisk-bugs] [Asterisk 0013034]: [patch] 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:                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 11:05 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
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---------------------------------------------------------------------- 
 (0102356) festr (reporter) - 2009-03-29 11:05
 http://bugs.digium.com/view.php?id=13034#c102356 
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hi, I've the same problem as reporter (the latest SVN version)

i've configured sip.conf progressinband=never

I'm receiving to asterisk
- SIP/SDP Request: INVITE sip:xx at yy, with session description
asterisk responds with 183 with SDP although there is
progressinabnd=never

This call (INVITE) is routed to Zap/g1 PRI euroISDN

on pri debug: 
Message type: CALL PROCEEDING (2)
(after 2 seconds)
Message type: ALERTING (1)

after received ALERTING, asterisk sends 180 RING

but asterisk is not generating alert inside RTP SDP although it sends RTP
(but silence) so the far end is not generating alert (although 180 RING is
sended by us) because of early 183 

I'm not sure that i've explain it clear. If any question, logs or so, tell
me. 

(I've made small hacks inside channel_sip.c to not send 183 with SDP in
any situation) and it works for me in this installation. But it should be
fixed somehow. and the question - asterisk does not can generate its own
alert when ALERT is received on the PRI? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-29 11:05 festr          Note Added: 0102356                          
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