[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
======================================================================
More information about the asterisk-bugs
mailing list