[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk

[Girish Gopinath]

Zen,

>I am trying to confirm the command 'canreinvite=yes' in sip.conf
>using grandstream BT101/2s and snom100s. In either case, no description
>nor 'canreinvite=yes', media stream always go through *.
>
>Do I need another settings for confirming sip clients directly
>communicate each other?

Do you have a Dial statement that has "t" or "T" in it?
This will force the media stream to pass through Asterisk.

Regards, Girish

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