[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:              2009-02-23 07:03 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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---------------------------------------------------------------------- 
 (0100535) klaus3000 (reporter) - 2009-02-23 07:03
 http://bugs.digium.com/view.php?id=13034#c100535 
---------------------------------------------------------------------- 
Using your patch the 183 is supressed, but I get another 100 trying after
CALL PROCEEDING (additional to the one just sent after the INVITE is
received)


< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 3/0x3) (Terminator)
< Message type: CALL PROCEEDING (2)
q931.c:3641 q931_receive: call 32771 on channel 1 enters state 3 (Outgoing
call  Proceeding)
Klaus: received AST_CONTROL_PROCEEDING on channel DAHDI/1-1
    -- DAHDI/1-1 is proceeding passing it to SIP/klaus-08863120

<--- Transmitting (NAT) to 83.136.33.3:46922 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.10.0.51:46922;branch=z9hG4bK-d8754z-b14d2667ea26c136-1---d8754z-;received=83.136.33.3;rport=46922
From: "klaus samuel"<sip:klaus at 83.136.32.165>;tag=31219905
To: "01505641636"<sip:01505641636 at 83.136.32.165>
Call-ID: MjUzMTljOTA5ZTAxYzVlZjcyMDNkNzE3ZGZjNTY5NmQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01505641636 at 83.136.32.165>
Content-Length: 0 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-23 07:03 klaus3000      Note Added: 0100535                          
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