[asterisk-users] No ring sound when calling SIP extensions over Webrtc

[D'Arcy J.M. Cain]

On Wed, 9 Sep 2015 16:27:19 -0500
Carlos Chavez <cursor at telecomabmex.com> wrote:
> The file is full of definitions for many countries.  It specifically
> has one for Mexico but we usually use the same one as the USA.

I simplified mine:

[general]
country=us              ; default location


[us]
description = United States / North America
ringcadence = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/2000,0/4000
congestion = 480+620/250,0/250
callwaiting = 440/300,0/10000
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440


-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net