Tilghman Lesher [EMAIL PROTECTED] wrote:
> But since we have granted Digium a license to relicense the code to
> other parties, under other licenses, we need Digium to provide notice
> that they have not granted such a license before we can reasonably
> take action to enforce our rights under the G
Amar P Patel [EMAIL PROTECTED] wrote:
> What voice over IP providers are recommended for use with Asterisk and
> Degium hardware? I have searched on Google and found Vonage. I want to
> setup a voice over IP phone line that rings into the Asterisk box.
>
VoIP providers will probably not care whet
Philip Trauring [EMAIL PROTECTED] wrote:
> I'm new to the dev list. I tried searching online about running
> Asterisk with uClinux and found very little. Is anyone successfully
> doing this? Is there some web page that will direct me on how to do
> this? I was very surprised that there was nothing
Jeremy McNamara [EMAIL PROTECTED] wrote:
> Matthew Boehm wrote:
> > Hey guys,
> > I'm getting ready to write a custom least cost routing app for our
> > company and wanted to get some pro/con feedback on programming this app
> > a) in PHP as an AGI or b) in C as a loadable .so module
> >
> Use
>
> This code is new and I would appreciate any feedback on the install
> process or operational problems.
>
Well done on the G.729 and G.723.1 codecs. I keep meaning to give them
a go, but never seem to get around to it. It's nice to know that
they're out there though. Keep up the good work.
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
> If the use of a G729 converter in batch mode for non-realtime is
> royalty-free, then this would be a way forward to live happily ever
> after with Asterisk in passthrough mode only.
>
If this usage of G.729 was royalty-free then it wo
Constantine Filin [EMAIL PROTECTED] wrote:
> > While I understand the concern of too many 'vendor' specific
> > implementations, I applaud the inclusion of 'query', 'insert', etc.
> > commands into the asterisk dialplan versus through an AGI. If the
> > general consensus is to move this into ast_d
Vassilis Konstantinou [EMAIL PROTECTED] wrote:
> It appears that the new channels/chan_zap.c
> has quite a few new lines of code and makes the original UK Caller ID
> patch (second patch for the chan_zap file) to fail.
>
> I tried to find my way through the code but it is getting a bit too
> compl
[EMAIL PROTECTED] wrote (in asterisk-cvs):
> Update of /usr/cvsroot/asterisk/channels
> In directory mongoose.digium.com:/tmp/cvs-serv22203/channels
>
> Modified Files:
> chan_zap.c
> Log Message:
> Allocate pseudo channel if it's not explicitly listed
>
>
> Index: chan_zap.c
> ==
Steven Sokol [EMAIL PROTECTED] wrote:
> How do you plan to use Asterisk? The GPL restricts you from selling the
> code, but many people have commercial products that use Asterisk.
>
The GPL doesn't restrict you from selling the code; you can charge
whatever you like. The GPL simply stipulates tha
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