[asterisk-users] Outbound Asterisk calls default directmedia specifications

[Kevin P. Fleming]

On 07/04/2012 01:47 PM, sathiish kumar wrote:
> Thanks for the response.. I did change it in the [general] settings.My
> setup is something like I have a remote conference (not meetme) which
> will send reinvite to redirect the RTP flow to a different server to
> load balance.There are three clients who join in the conference and i
> can listen to two other clients speak from the third client but when i
> record the conversation my recording of one of the clients ends before
> the stipulated hangup time. I am guessing this is because one of the
> clients doesn't understand what to do with a reinvite.. Any
> suggestions.In the SIP.conf i have changed the directmedia option to no
> and also enabled the ignoresdpversion option.

The 'directmedia' option *only* controls whether Asterisk will attempt 
to drop itself out of the media path between two SIP endpoints. It has 
no effect on whether or not Asterisk will respond appropriately to a 
re-INVITE received *from* a SIP endpoint (to which Asterisk should 
always respond properly, unless the re-INVITE is malformed in some way 
or is unacceptable).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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