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

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13034 
====================================================================== 
Reported By:                klaus3000
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13034
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             07-09-2008 07:03 CDT
Last Modified:              07-09-2008 15:11 CDT
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 oej - 07-09-08 15:11  
---------------------------------------------------------------------- 
q931.c:3844 q931_receive: call 32770 on channel 1 enters state 2 (Overlap
sending)

This occurs before we decide that we have media from the ISDN that we want
to send to the SIP phone. Is there an AST_FRAME that triggers this? 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-09-08 15:11  oej            Note Added: 0089970                          
======================================================================