Summary:	ASTERISK-04008: progressinband=yes does not work
Reporter:	adomjan (adomjan)	Labels:
Date Opened:	2005-04-28 06:11:40	Date Closed:	2011-06-07 14:10:13
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/CodecHandling
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) callstrace ( 1) dialplan ( 2) sip-trace-2 ( 3) sip-progressinband.patch
Description:	Progressinband does not work whith sip - sip. It's very bad when I call an unreachable mobile via sip telco connection. The mobile provider send inband unreachable message, I just hear ringing.... The telco send rtp traffic this time asterisk does not to phone. I checked with tcpdump. It's the same behavior when I make call with 2 grandstream phone. The asterisk v1 stable (CVS-v1-0-04/19/05-15:52:39) is ok.
Comments:	By: Kevin P. Fleming (kpfleming) 2005-04-28 11:28:31 As the bug guidelines request, please post a snippet of your dialplan showing how you are originating the call, and a _complete_ trace of the call in progress. That includes 'set verbose 255', 'set debug 255' and 'sip debug'. By: ewieling (ewieling) 2005-04-28 15:56:05 You're not using the "r" option to Dial, are you? By: adomjan (adomjan) 2005-04-29 02:37:44 I'm not using r option... I uploaded the strace, and a part of dialplan. By: Kevin P. Fleming (kpfleming) 2005-04-29 10:53:22 OK, that trace shows no "183 Session Progress" messages at all. That means they cannot send you inband progress, because they haven't told us to expect it. By: adomjan (adomjan) 2005-04-29 12:38:48 thanks, the another problem is when I make call between my 2 sip phones, asterisk does not send inband ringing tone. I would like to use it because the grandstream phones generate only us ringig tone. Another * running cvs v1-0 send inband ringing. By: Kevin P. Fleming (kpfleming) 2005-04-29 12:49:42 Again, we cannot help you with the problem with only "asterisk does not send ringing tone". We need to see the portion of the dialplan that's involved and a trace of a failing call. Otherwise we are just shooting questions at you one at a time until we happen upon the correct one. By: adomjan (adomjan) 2005-05-02 01:40:29 Ok, I uploaded sip-trace-2 from dialplan: exten => 1234,1,Dial(SIP/15800100) exten => 1234,2,Hangup By: adomjan (adomjan) 2005-05-09 11:19:44 I really need inband progress... My only dirty way was I modified if (!ast_test_flag(p, SIP_PROGRESS_SENT) ||     (ast_test_flag(p, SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) condition to always false near 2144 line. I dont have time to find where is proplem in chan_sip, but I would like to say it's still exists. By: adomjan (adomjan) 2005-05-10 06:28:50 I made a small modification (sip-progressinband.patch) and seems ok now. By: Michael Jerris (mikej) 2005-05-10 07:56:11 Is this really right?  It looks to me like this turns progress on when progress is not sent, and off if it is? By: adomjan (adomjan) 2005-05-10 08:19:56 But now it works... May the bug is somewhere else. I did not say apply the patch. By: Olle Johansson (oej) 2005-05-13 13:03:10 I use early media quite a lot in SIP and it works in asterisk. Whether or not the UA can here it is up to the UA. By: Kevin P. Fleming (kpfleming) 2005-05-14 21:09:55 I cannot see there that is actually anything wrong with Asterisk's behavior here. You are requesting that early audio be turned on without receiving indication from the peer that it should be, which is definitely not correct behavior according to every accepted SIP implementation. If you have peers that are sending you audio without sending you "183 Session Progress" first, then those peers are broken.