We relay on 'failed' extensions after AMI ORIGINATE command.
When moving from Asterisk 1.8.22 to Asterisk 13.2, it has stopped to work.
I belive that it is due to a change in pbx.c = ast_pbx_outgoing_exten.
Thanks,
Valter
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On 05/24/2015 11:01 PM, Mehdi Shirazi wrote:
Hi
I want to load balance SIP calls between two(or more)
Asterisks with only DNS SRV. I used bidirectional sync
Unison to synchronize configuration files and internal database file
between two Asterisk boxes.
The problem is when a calls come to
