[asterisk-bugs] [Asterisk 0017867]: chanspy cuts the voice between caller and callee if Local channel is used

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Aug 16 14:43:49 CDT 2010


The following issue requires your FEEDBACK. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17867 
====================================================================== 
Reported By:                chris0602
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17867
Category:                   Applications/app_chanspy
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.11 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-08-16 04:00 CDT
Last Modified:              2010-08-16 14:43 CDT
====================================================================== 
Summary:                    chanspy cuts the voice between caller and callee if
Local channel is used
Description: 
I'm making a SIP call from a SIP softphone through a SIP GSM gateway.
Because of 6 seconds silence during the connenction through the GSM
gateway I'm playing tones. Thats works fine until I try to spy the channel.
Then the caller can't hear the callee, the callee can't hear the caller but
the spier hears both sides. If the spier stays on the channel and the
caller establishes a new connection all works fine.



[out-gsmgate]
EXTEN => xxx, 1, Answer()
EXTEN => xxx, n, Playtones(waitforring)
EXTEN => xxx, n, dial(SIP/gsmgate&Local/s at no-op,60)

[no-op]
EXTEN => s, 1, Hangup()
====================================================================== 

---------------------------------------------------------------------- 
 (0125991) lmadsen (administrator) - 2010-08-16 14:43
 https://issues.asterisk.org/view.php?id=17867#c125991 
---------------------------------------------------------------------- 
Not enough information provided. Please include console output, channel
configuration information and the entire dialplan required to reproduce
this issue.

We'll also need SIP debug information. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-16 14:43 lmadsen        Note Added: 0125991                          
2010-08-16 14:43 lmadsen        Status                   new => feedback     
======================================================================




More information about the asterisk-bugs mailing list