Hi, My SIP service provider terminates calls in meetme in my Asterisk PBX and am getting delay on those channels. I found following link to measure delay in meetme and to decrease it eventually. http://lists.digium.com/pipermail/asterisk-dev/2005-August/014958.html It says, enable USE_RTC for dahdi_dummy.
I have been using virtual server for hosting Asterisk and I had it disabled as per one had mentioned here to prevent the crashing which was happening earlier..(http://www.odindev.com/content/troubles-zaptel-centos-52-xen). Can you shed some light on the issue? --SM -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
