[asterisk-bugs] [Asterisk 0014295]: SIP on hold problems

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14295 
====================================================================== 
Reported By:                klaus3000
Assigned To:                file
====================================================================== 
Project:                    Asterisk
Issue ID:                   14295
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.22 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-01-21 07:20 CST
Last Modified:              2009-01-23 12:11 CST
====================================================================== 
Summary:                    SIP on hold problems
Description: 
Hi!

I have found 2 problems when the SIP phone puts a call on hold (a SNOM
phone which uses a=sendonly).

1. Although only RTP should be activated it looks like RTCP is deactivated
too, as I got these error message on the console:

  RTCP SR transmission error, rtcp halted

2. When the call is on hold and I unplug the phone, I would think that the
session is terminated after "rtpholdtimeout". Instead the session is
termianted after rtptimeout.

see trace below: rtptimeout=15, rtpholdtimeout=25
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0013835 "RTCP SR transmission error, rtcp ...
====================================================================== 

---------------------------------------------------------------------- 
 (0098594) svnbot (reporter) - 2009-01-23 12:11
 http://bugs.digium.com/view.php?id=14295#c98594 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 170507

_U  branches/1.6.1/
U   branches/1.6.1/channels/chan_sip.c

------------------------------------------------------------------------
r170507 | file | 2009-01-23 12:11:09 -0600 (Fri, 23 Jan 2009) | 18 lines

Merged revisions 170505 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170504 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4
lines
    
    Use the on hold flag to see if the call is on hold or not. It is
possible that our address for them will still be valid even though they are
on hold.
    (closes issue http://bugs.digium.com/view.php?id=14295)
    Reported by: klaus3000
  ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=170507 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-23 12:11 svnbot         Checkin                                      
2009-01-23 12:11 svnbot         Note Added: 0098594                          
======================================================================