[asterisk-bugs] [Asterisk 0018072]: RTPAUDIOQOSJITTERBRIDGED and other variables contains nothing if call is terminated from DAHDI

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Sep 29 11:39:26 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18072 
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Reported By:                sles
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18072
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.13 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-09-28 23:22 CDT
Last Modified:              2010-09-29 11:39 CDT
====================================================================== 
Summary:                    RTPAUDIOQOSJITTERBRIDGED and other variables
contains nothing if call is terminated from DAHDI
Description: 
Hello!

I have following in extensions.conf:
exten => h,n,NoOp(-- QoS stats RTPAUDIOQOSJITTERBRIDGED:
${RTPAUDIOQOSJITTERBRIDGED})                                               
                         
exten => h,n,NoOp(-- QoS stats RTPAUDIOQOSLOSSBRIDGED:
${RTPAUDIOQOSLOSSBRIDGED})                                                 
                           
exten => h,n,NoOp(-- QoS stats RTPAUDIOQOSRTTBRIDGED:
${RTPAUDIOQOSRTTBRIDGED})    


Than I do Dial from DAHDI (E1 PRI link) to SIP:

    -- Executing [6052 at default:4] Dial("DAHDI/28-1", "SIP/6052") in new
stack


Really, 6052 is ekiga, i.e. softphone.

Than if I do hangup in ekiga I get:

    -- Executing [h at default:2] NoOp("DAHDI/14-1", "-- QoS stats
RTPAUDIOQOSJITTERBRIDGED: rxjitter=0.003858;") in new stack
    -- Executing [h at default:3] NoOp("DAHDI/14-1", "-- QoS stats
RTPAUDIOQOSLOSSBRIDGED: lost=16;expected=126;") in new stack
    -- Executing [h at default:4] NoOp("DAHDI/14-1", "-- QoS stats
RTPAUDIOQOSRTTBRIDGED: Not available") in new st

but if do I hangup on my phone, connected to PBX, which is connected to
asterisk over DAHDI I get:

    -- Executing [h at default:2] NoOp("DAHDI/6-1", "-- QoS stats
RTPAUDIOQOSJITTERBRIDGED: ") in new stack
    -- Executing [h at default:3] NoOp("DAHDI/6-1", "-- QoS stats
RTPAUDIOQOSLOSSBRIDGED: ") in new stack
    -- Executing [h at default:4] NoOp("DAHDI/6-1", "-- QoS stats
RTPAUDIOQOSRTTBRIDGED: ") in new stack

Variables are empty. 


Looks like a bug.

Thank you!


====================================================================== 

---------------------------------------------------------------------- 
 (0127504) lmadsen (administrator) - 2010-09-29 11:39
 https://issues.asterisk.org/view.php?id=18072#c127504 
---------------------------------------------------------------------- 
I don't think this is a bug. I'm pretty sure this is working based on the
way it is implemented.

When the SIP channel hangs up, the hangup routine is executed by that
channel, which contains the RTCP data. When the DAHDI channel hangs up, the
hangup routine is executed by that channel, which obviously would have no
RTCP data. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-09-29 11:39 lmadsen        Note Added: 0127504                          
======================================================================




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