vicidial.org

hello
i have implemented AMD in my dialer implementation its working but
its not upto the mark
i am using SIP trunk can someone guide be any specific setting for SIP
trunks so that i can achive best possible results

i basicaly want to know some brief on what parameter in AMD() i shall play around on SIP trunks

Topic moved to Support:

Please post support questions in the proper Forum.

codec?

Asterisk version?

astguiclient version?

loadavg while calling?

Concurrent calling maximum channels?