[asterisk-bugs] [Asterisk 0018959]: Can't spy direct channel while spying in a group

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Apr 12 15:13:58 CDT 2011


The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=18959 
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Reported By:                jamicque
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18959
Category:                   Applications/app_chanspy
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.18-rc1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-03-11 09:37 CST
Last Modified:              2011-04-12 15:13 CDT
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Summary:                    Can't spy direct channel while spying in a group
Description: 
When I want to run chanspy defining direct channel I want to spy, and a
group to cycle through I hear nothing.
here is a sample dialplan:
exten => 556,1,Set(SPYGROUP=TEST)
exten => 556,n,ChanSpy(test002,g(TEST)d)

if I run it: like that:
exten => 556,n,ChanSpy(,g(TEST)d)
or that:
exten => 556,n,ChanSpy(test002,d)
it works ok. But defining the channel and the extension is no possible.
On the other hand using ExtenSpy works ok:
exten => 556,n,ExtenSpy(test002,g(TEST)d)


Beneath is the log from connection:
  == Using SIP RTP CoS mark 5
    -- Executing [12 at phones:1] Goto("SIP/test004-00000080",
"internal,12,1") in new stack
    -- Goto (internal,12,1)
    -- Executing [12 at internal:1] Set("SIP/test004-00000080",
"SPYGROUP=TEST") in new stack
    -- Executing [12 at internal:2] Dial("SIP/test004-00000080",
"SIP/test002,10") in new stack
  == Using SIP RTP CoS mark 5
    -- Called test002
    -- SIP/test002-00000081 is ringing
    -- SIP/test002-00000081 answered SIP/test004-00000080
    -- Packet2Packet bridging SIP/test004-00000080 and
SIP/test002-00000081
  == Using SIP RTP CoS mark 5
    -- Executing [556 at phones:1] Goto("SIP/test003-00000082",
"queue,556,1") in new stack
    -- Goto (queue,556,1)
    -- Executing [556 at queue:1] Set("SIP/test003-00000082",
"SPYGROUP=TEST") in new stack
    -- Executing [556 at queue:2] ChanSpy("SIP/test003-00000082",
"test002,g(TEST)d") in new stack
    -- <SIP/test003-00000082> Playing 'beep.gsm' (language 'en')

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---------------------------------------------------------------------- 
 (0133676) lmadsen (administrator) - 2011-04-12 15:13
 https://issues.asterisk.org/view.php?id=18959#c133676 
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Can you provide a SIP trace along with a pcap that shows the RTP flow? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-04-12 15:13 lmadsen        Note Added: 0133676                          
2011-04-12 15:13 lmadsen        Status                   new => feedback     
======================================================================




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