Hi Rich, Quoting Rich Adamson <[EMAIL PROTECTED]>:
> > 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. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
