[asterisk-bugs] [Asterisk 0013572]: WaitForSilence() sometimes doesn't always wait when using SIP and a callfile

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Oct 1 11:12:03 CDT 2008


The following issue requires your FEEDBACK. 
====================================================================== 
http://bugs.digium.com/view.php?id=13572 
====================================================================== 
Reported By:                DennisD
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13572
Category:                   Applications/app_waitforsilence
Reproducibility:            sometimes
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0-rc6 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-09-27 15:38 CDT
Last Modified:              2008-10-01 11:12 CDT
====================================================================== 
Summary:                    WaitForSilence() sometimes doesn't always wait when
using SIP and a callfile
Description: 
WaitForSilence() seems to always work when using a provider over IAX2 and a
call file, but doesn't always work with SIP providers.

For some reason, if I use Future-Nine to send the call out over SIP, it
works, but it fails to wait if I send the call over Callcentric (SIP) or
voip.ms (SIP).

If I send it over rapidvox (IAX2) or voip.ms (IAX2), WaitForSilence()
waits.

If I call the extension directly from a SIP phone (not using a call file),
it waits.

I've gone through my sip.conf many times, trying to see what the
differences could be, but I can't find anything that could cause this.
====================================================================== 

---------------------------------------------------------------------- 
 (0093026) blitzrage (administrator) - 2008-10-01 11:12
 http://bugs.digium.com/view.php?id=13572#c93026 
---------------------------------------------------------------------- 
Can you provide us some additional debugging information?

Since the issues seem to always be with SIP, can you provide a SIP trace
and the SIP history of the call?

In addition, it might be useful to have a wireshark trace with the RTP
audio as well in order to determine if it has something to do with the
audio those providers are sending.

Can you provide traces for calls that work, and calls that do not work in
order to allow us to compare?

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-10-01 11:12 blitzrage      Note Added: 0093026                          
2008-10-01 11:12 blitzrage      Status                   new => feedback     
======================================================================




More information about the asterisk-bugs mailing list