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).
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