[asterisk-users] Call Termination Provider Madness

[Chris Nighswonger]

On Wed, Oct 3, 2012 at 4:37 AM, Michel Verbraak <michel at verbraak.org> wrote:
> Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP
> portrange your asterisk will use for RTP traffic. change the rtpstart and
> rtpend to your needs and set them open in your FW. Do not make the range too
> small each active call will normally take one RTP channel incoming and one
> RTP channel outgoing.
> I have mine set to for example: rtpstart=10000 and rtpend=10100. This should
> be enough for 100 simultanious calls.

Thanks to everyone for the help in this regard. Its amazing how much I
still do not know after nearly 30 years of wrestling with computers.
:-)

A lack of understanding about the nature of RTP led me to limit
traffic inbound from specific IPs which, of course, led to inbound
call weirdness.

At this point I only have ~40 extensions, so I took Michel's advise
and set my RTP range to 10000-10100. The default 10000 ports was a bit
more surface area than I want to expose.

Kind Regards,
Chris