[PLUG] SIP Providers with Asterisk/FreePBX

[Aaron Burt]

This reminds me, I've registered asterhack.org but haven't put up a site or
announced a meeting yet.  Please mail me offlist if you, dear reader, would
be interested in an open-source telephony group here in Portland.

On Tue, Aug 21, 2012 at 05:28:12PM -0700, Scott Bigelow wrote:
> Am I unreasonable in thinking that it is nexVortex's job to transcode this
> connection, and not burden us with supporting all codecs their customers
> might be using? Or should I be allowing more codecs on my trunk? Thanks!

You should be transcoding or allowing more codecs.  NexVortex is just a
broker, and they're most likely not even in the middle of the RTP stream.

Also, g.729 usually requires a license to use, so it's often disabled.

Uness bandwidth is significantly constrained, I usually use g.711a/u - it's
still the best-sounding codec(s), and universally supported.