add
canreinvite=no
to the sip user definition blocks for the SIP provider and for the SIP ATA.
Regards,
Marc
Wolf N. Paul wrote:
Hello,
how can I prevent Asterisk from trying to create a native bridge between an incoming call from a SIP provider and an extension attached to a SIP ATA?
My Asterisk is behind a firewall, and the native bridge invariably fails.
Thanks in advance for any suggestion!
(I DID search the list archives for "native bridge" and found one similar query without any replies).
Regards,
Wolf Paul _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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