[Asterisk-Users] Taking asterisk out of the media path - SIP - howis it achieved

[Tulika Pradhan]

the following are necessary conditions for asterisk out of mediia path -
- the codecs on both endpoints of the call should be the same.
- the technology on both endpoints should be the same.
- canreinvite=yes
- no 't', 'T', L, conference, transfer flags in Dial command

tulika


>From: David John Walsh <davidjohnwalsh at gmail.com>
>Reply-To: David John Walsh <davidjohnwalsh at gmail.com>,Asterisk Users 
>Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
>To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>Subject: [Asterisk-Users] Taking asterisk out of the media path - SIP - 
>howis it achieved
>Date: Mon, 2 May 2005 21:41:49 +0100
>
>Hello
>
>How do you make asterisk stay out of the media stream?  i.e once I set
>a call up between two parties, even if asterisk fell over the call
>would continue (in the same way a HLR on a mobile network works)
>
>I understand that many features will be lost if I do this, but all
>that I need seems to be supported by the end user hardware.
>
>incidentally I have tried canreinvite=yes, doesn't seem to work.  I
>have also tried removing any flags in the dial() command
>
>Thank you for any information.
>
>David
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