[Asterisk-Users] How to prevent native bridging between SIP channels

[Wolf N. Paul]

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