On Tue, Oct 24, 2006 at 07:28:57AM -0500, Kevin P. Fleming wrote:
> Whatever we do, we will not offer formats to the second endpoint that
> were not chosen during negotiation with the first endpoint. We don't
> want the call to be setup using direct media and a format which we
> cannot support shou
On Mon, Oct 23, 2006 at 05:47:56PM -0500, Kevin P. Fleming wrote:
> > (However, I have no idea what these devices would do if you completed the
> > call, and then they had to use the 'WIBBLE' codec to send audio...)
>
> I don't think that my proposal is practical; instead, what I am going to
> do
On Mon, Oct 23, 2006 at 02:37:39PM +0100, Brian Candler wrote:
> Here's one I knocked together in Perl:
> http://pobox.com/~b.candler/software/testsiperror
>
> The fake codec it offers is "WIBBLE/8000".
>
> And the results I get when pointing it at various SIP
On Tue, Oct 17, 2006 at 11:13:48AM +0100, Alexandre Almeida wrote:
> [xxyy]
> username=123456
> type=peer
> secret=1423241
> port=5070
> outboundproxy=my.proxy.com
> host=gateway.server.com
>
>
> when dialing a number like this 00123456789:
>
> asterisk make this,
>
> SIP:[EMAIL PROTECTED]:5070
On Tue, Oct 17, 2006 at 08:54:50AM +0100, Brian Candler wrote:
> Sorry, I missed the notes, as I wasn't emailed when they were added. (Mantis
> did this when notes were added to 8072
Mistype, I meant 8078. Anyway, that's a report I opened in the same way
AFAIK, and whenever someon
On Mon, Oct 16, 2006 at 03:34:10PM +0200, Johansson Olle E wrote:
> Or, set up the call as normal - one call leg to Asterisk, transcode
> and another
> call leg and media stream to the other device.
>
> That should be the case here.
But the point is - how do you know when you need to do that?
On Mon, Oct 16, 2006 at 03:11:15PM +0100, Brian Candler wrote:
> - if you get a 606 then INVITE again, this time listing all the codecs you
> are prepared to transcode for,
... and in particular, at this point don't offer G729 if you don't have a
G729 licence available ...
>
On Mon, Oct 16, 2006 at 02:11:09PM +0100, Brian Candler wrote:
> * Make the INVITE to the second phone only offer codecs which the first
> phone offered. Then if the call is rejected due to no compatible codec
> (606?), try again using the normal two-leg setup, with SDP pointing at the
&
On Mon, Oct 16, 2006 at 01:28:13PM +0200, Johansson Olle E wrote:
> >I've tested it now, and in my SVN build, which is about 10 days out
> >of date,
> >it appears to be broken.
> >
> >Full description posted at http://bugs.digium.com/view.php?id=8152
> >
> >In summary: the two endpoints think the
On Thu, Oct 05, 2006 at 03:47:22AM -0400, Jeremy McNamara wrote:
> I suggest you purchase the Asterisk book. You seem to be missing quite a
> few major core concepts of how Asterisk functions.
>
>
> http://www.oreilly.com/catalog/asterisk/
Or just download the whole book for free:
http://www.as
On Tue, Sep 26, 2006 at 05:02:56PM -0500, Jason Parker wrote:
> Let's look at other apps (say, apache) for an example of something
> similar. What happens if you throw thousands of requests at apache per
> second? It's going to die. And why shouldn't it? Stuff like this, in my
> opinion, is bes
On Mon, Sep 04, 2006 at 07:18:33PM +0500, Anton wrote:
> Does anyone know is possible to pass DIALUP modem calls
> over, through asterisk? Nothing extra needed I think, say
> asterisk is in-between of 2 telcos, connected over E1 from
> both sides. Modem calls is already DATA, so will it work,
>
Looking at SVN trunk source for asterisk, a couple of minor points:
(1) doc/mp3.txt says:
Please use mpg123 0.59r. Using mpg123 pre0.59s can/may/will result
in crashes and/or unreliable playback.
Running "make mpg123" in the Asterisk source directory will
download and make the w
On Tue, Aug 22, 2006 at 04:53:44PM +0100, Julian Lyndon-Smith wrote:
> libtasn1 is a library for Abstract Syntax Notation One (ASN.1) and
> Distinguish Encoding Rules (DER) manipulation.
>
> Whatever that means :)
http://www.itu.int/ITU-T/studygroups/com17/languages/X.680-0207.pdf
http://www.itu
On Fri, Jul 28, 2006 at 12:49:14PM +0200, Olivier Krief wrote:
> In many cases, I noticed (though I didn't experienced) data modems could
> simply be replaced by serial-to-ethernet converters plus dialup routers.
> What do you think of that ?
That won't work for me:
(1) EPOS terminals may have m
On Fri, Jul 28, 2006 at 10:27:58AM +0100, Brian Candler wrote:
> Does anyone know if there's a standard mechanism for carrying data modem
> calls over a VoIP-type network? I know there's T.38 for fax, but I have not
> been able to find something equivalent for generic data mode
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