Hi,,

When using a SIP proxy to load balance calls how do you make it that a call
on an attended transfer reaches the same Asterisk box every time? I was
told that in later versions of Asterisk there is some "magic" to make it
work correctly when load balancing.

TIA.

Dovid
-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
      https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to