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

Zen Kato zenkato at pis.bekkoame.ne.jp
Wed Mar 3 15:53:58 MST 2004


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?


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