[asterisk-bugs] [Asterisk 0014126]: Using dtmfmode=info AND canreinvite=yes (both in sip.conf) AND dynamic features (features.conf/Dial() with w or W flags)

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Dec 22 15:44:15 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14126 
====================================================================== 
Reported By:                malaiwah
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14126
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.23-rc3 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-12-22 15:22 CST
Last Modified:              2008-12-22 15:44 CST
====================================================================== 
Summary:                    Using dtmfmode=info AND canreinvite=yes (both in
sip.conf) AND dynamic features (features.conf/Dial() with w or W flags)
Description: 
If Asterisk is configured for SIP INFO and re-invites, if one wants to
enable dynamic features on a Dial application call, Asterisk will stay in
the audio path for all the conversation.

This behavior should be correct for in-band DTMF and RFC2833 but for SIP
INFO, it could still send a re-invite and get back the voice path IF a
dynamic feature is engaged (in my case, it is
http://bugs.digium.com/view.php?id=3 for quick recording of the
conversation).
====================================================================== 

---------------------------------------------------------------------- 
 (0096846) russell (administrator) - 2008-12-22 15:44
 http://bugs.digium.com/view.php?id=14126#c96846 
---------------------------------------------------------------------- 
I just want to note that your note about a built in limit for maximum
number of calls is bogus.  There is no such limit.

Can you please provide your configuration files, a SIP trace of the call
in question, as well as your console output when doing a test?  That will
help us reproduce the problem. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-22 15:44 russell        Note Added: 0096846                          
======================================================================




More information about the asterisk-bugs mailing list