Summary:	ASTERISK-03305: [feature] stayinmediapath=on/off command for SIP
Reporter:	unknown	Labels:
Date Opened:	2005-01-18 13:13:56.000-0600	Date Closed:	2011-06-07 14:05:22
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/NewFeature
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	The ability to turn off/on asterisk staying in the media path through a [general] configuration option in sip.conf like stayinmediapath=on or mediapath=on. Right now, it's not very intuitive when trying to keep Asterisk out of the media path. I know you can add T or t to the Dial() command, or try canrevinvite=yes - but this does not always seem to work either and are just workarounds. It would be nice if we had the ability to turn off/on Asterisk staying in the media path.
Comments:	By: () 2005-01-18 13:14:42.000-0600 Oops the title should read: [feature] stayinmediapath=on/off command for SIP By: nick (nick) 2005-01-18 13:16:22.000-0600 Is that *exactally* what canreinvite is for?  If there's a problem with it, trace it down and file a bug.  Let's not add new features to work around bugs! By: Brian West (bkw918) 2005-01-18 13:30:39.000-0600 yes thats what canreinvite is for :P bkw By: () 2005-01-18 13:38:44.000-0600 Thanks for the karma reduction :) Mantis should have a title editing ability, so we could avoid simple mistakes.. but I digress... Perhaps I need to file a new bug, or canreinvite needs to be more intuitive. I can tell you that canreinvite does not always work, so then we have another issue altogether :) By: Mark Spencer (markster) 2005-01-18 14:45:19.000-0600 Provide the config and a sip debug of canreinvite failing. By: mochouinard (mochouinard) 2005-01-18 16:31:17.000-0600 I would like to beable to set this flag depending of IP address. of the remote rtp stream.  Like I can't link a local intanet stream with a Internet stream... By: Brian West (bkw918) 2005-01-18 16:32:39.000-0600 The invite can only work if both ends can speak the same codec/ability.  If one side is speaking gsm only and one is speaking ulaw only asterisk will not allow a reinvite.  Also if you have any flags such as TtwW or anything where asterisk needs to watch the mediastream.. a reinvite is impossible. bkw By: Olle Johansson (oej) 2005-01-25 12:15:36.000-0600 Hmm, maybe we should check IP with the localnet settings and turn off canreinvite if one end is in the localnet and the other is not. By: Mark Spencer (markster) 2005-01-25 12:31:52.000-0600 Either we need a sip debug for why canreinvite=yes isn't working or we need to close this bug as a configuration issue. By: Olle Johansson (oej) 2005-01-30 12:26:22.000-0600 ---Configuration issue.