[asterisk-bugs] [Asterisk 0012937]: DTMF package's ssrc number wrong when partical bridge channel

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Dec 4 15:42:45 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12937 
====================================================================== 
Reported By:                gino_he
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12937
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.20.1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-06-27 04:44 CDT
Last Modified:              2008-12-04 15:42 CST
====================================================================== 
Summary:                    DTMF package's ssrc number wrong when partical
bridge channel
Description: 
I using asterisk-1.4.20.1 as my pbx,with two extensions(2000,8888),my
asterisk register a sip provide by 8888.The provide can dial into pstn.
sip.conf
[general]
register => 8888:8888 at myprovide
[anthentication]
auth => 8888:8888 at 172.21.9.202
[8888]
username=8888
type=friend
secret=8888
host=172.21.9.202
context=test_context
canreinvite=no

[2000]
type=friend
secret=8888
host=dynamic
context=test_context
canreinvite=no

extension.conf
[test_context]
exten => _9.,1,dial(sip/8888/${EXTEN},,)


when I make a call from 2000 by pressing 910086,the call be setuped and I
can hear IVR,then I press some number key on the phone,but nothing
happend.during the test,I captured package. From the packe I found asterisk
bridge them as partical bridge that I don't know what is meaning.and
asterisk using the wrong ssrc in dtmf package.



if I allow asterisk to reinvite,this won't happen,but it is native
bridge.
If I disallow asterisk to reinvite, but change dial rule as

extension.conf
[test_context]
exten => _9.,1,dial(sip/8888/${EXTEN},,T)

asterisk bridge two channel in generic mode, but asterisk didn't forward
any dtmf package


Sorry for my poor english
====================================================================== 

---------------------------------------------------------------------- 
 (0095803) blitzrage (administrator) - 2008-12-04 15:42
 http://bugs.digium.com/view.php?id=12937#c95803 
---------------------------------------------------------------------- 
Suspended due to lack of activity. Please request a bug marshall on
#asterisk-bugs on the IRC network irc.freenode.net to reopen the issue
should you have the additional issue requested. Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-04 15:42 blitzrage      Note Added: 0095803                          
======================================================================




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