[Asterisk-Users] Zap channel(s), meetme and codecs/licences

[jerry at voiptower.com]

Hi Rich,

Quoting Rich Adamson <radamson at routers.com>:

> > What codec does the Zap channel use by default?
>
> None/all. Look in /usr/src/asterisk/configs/zapata.conf.sample and
> you won't find any reference to codecs.

I did better; I googled, searched the wiki, and the list archives.
I find this hint from "zap show channel #":
Default law: ulaw

I didn't know if this was just a documentation thing, or if it was
something that was configurable.

> > Finally ... if I have a 3way call going, between 1 g729 caller and two
> > other callers, do I need one or two available licences? (I'm guessing that
> > zap doesn't do g729, and am wondering if I have an FXO caller and a local
> > FXS person talking to a VoIP caller using g729, how it would work)
>
> Someone else might want to chime in here, but it seems to me (as a non-
> programmer) that internal handling of voice packets (within *) were
> primarily slinear or something like that. If two end points can communicate
> with the same codec, the voice data is simply passed through (no
> conversion). If one user is g729 only communicating with other users,
> that g729 user consumes one license instance to convert to whatever
> the other users might be using. Two g729 users and one g711 user, likely
> uses two g729 licenses.

This is my thinking too, but that is what I'm a bit unclear about.

> Without a better understanding of what devices you are truly trying to
> use, I don't believe anyone is going to be able to answer your questions
> relative to zap channels and codec selection/conversion.

I have a customer that wants to try using a local IAX provider for origination.
The provider is using g729, and right now I have some digium FXS ports (along
with an FXO for outbound access). We have some SIP hardphones, and they all
have the ability to do g729. We will need a licence to do voicemail unless we
convert files to g729 format. However, to get a phonecall to be handled by
the Zap phones, I wasn't sure if something was needed or not, and you've
cleared that up.

I asked about meetme internal format just so I could see if there was a
possibility to "work around" the need for converters. My worry is that we
might run out of licences during the test, and that Asterisk would simply
drop a call if it can't complete one leg of the call, leaving both the caller
and my customer frustrated. Ideally we could buy n+1 licences, but I am
trying to get some sort of a feel for what "n" will look like, and how
practical/viable that is on a much larger scale rollout.

Thanks, by the way... you are always very helpful, both in the several replies
to my messages, and many other messages replies to others.

J.