[asterisk-bugs] [Asterisk 0014005]: Pickup() can't pickup calls to some SIP devices

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Dec 2 11:53:42 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14005 
====================================================================== 
Reported By:                ddl
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14005
Category:                   Applications/app_directed_pickup
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           Older 1.4 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-12-01 18:28 CST
Last Modified:              2008-12-02 11:53 CST
====================================================================== 
Summary:                    Pickup() can't pickup calls to some SIP devices
Description: 
I've observed that when a SIP device responds to an INVITE with a 183
without
ever sending a 180 the outgoing channel remains in the Down state rather
than
moving to Ring[ing].  This prevents Pickup() from picking up the call.  I
did
some extensive Google searching to try to understand the Pickup() failures
and
generally found a lot of reports where Pickup() seemed to work in some
cases
but not in others.  I wonder if this could be the root cause of some of
the
confusion.

By adding AST_STATE_DOWN to the test in app_directed_pickup.c I was able
to
get Pickup() to work with SIP devices (in this case the FXS port on a
Cisco
router) that behave as above; however, I have no idea what collateral
damage
this might cause.  It feels like there should be some other state for
call
progressing.

I am using 1.4.11 as provided in the FreeBSD 6.3 port.  I apologize if
this
has already been addressed in a newer version.  I checked 1.4.22 and the
code appeared the same.
====================================================================== 

---------------------------------------------------------------------- 
 (0095678) qwell (administrator) - 2008-12-02 11:53
 http://bugs.digium.com/view.php?id=14005#c95678 
---------------------------------------------------------------------- 
It's possible that it was fixed somewhere besides that code.

Please try the latest version. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-02 11:53 qwell          Note Added: 0095678                          
======================================================================




More information about the asterisk-bugs mailing list