Hello Phil, On Saturday, April 23, 2016, 11:11:29 PM, you wrote:
> Actually, this is now sorted. It turns out the latest recommended > configs on the A&A wiki had peer vs. user confusion. On correcting > this, all was well. I'm glad you found it. It look me a while to track down that problem when I had it. The one that was hardest for me to track down was a slight mis-match between the RTP ports in Asterisk and the corresponding ports open on a firewall, which resulted in about 1 in 10 calls having no audio! Doh! -- Best regards, Julian mailto:[email protected] -- _____________________________________________________________________ -- 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
