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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