Summary:	ASTERISK-12268: DTMF package's ssrc number wrong when partical bridge channel
Reporter:	gino (gino_he)	Labels:
Date Opened:	2008-06-27 04:44:37	Date Closed:	2011-06-07 14:02:37
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/RTP
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
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@myprovide [anthentication] auth => 8888:8888@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
Comments:	By: Joshua C. Colp (jcolp) 2008-07-02 21:36:01 Please provide an rtp debug and console output with DTMF logging (dtmf to console in logger.conf) with this. By: Leif Madsen (lmadsen) 2008-12-04 15:42:45.000-0600 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!