[asterisk-users] forward skinny call to SIP

[Jason Parker]

I would recommend using SIP instead of h323, if at all possible.

----- nik600 <nik600 at gmail.com> wrote:
> i am trying to download
> 
>  Open H.323 version v1.17.1, PWLib v1.9.0
> 
> but http://www.openh323.org/ seems to be down, can you suggest me an
> alternative link where to download them?
> 
> many thanks..
> 
> On 12/4/06, Pavel Jezek <pavel.jezek at i.cz> wrote:
> > 1.4 - make menuselect
> > 1.2 - make in channels/h323 (read readme.txt here)
> >
> >
> >
> > nik600 wrote:
> > > thanks
> > >
> > > can you explain me how to compile asterisk with h323 support? or
> is it
> > > biult in by default?
> > >
> > > On 12/4/06, Pavel Jezek <pavel.jezek at i.cz> wrote:
> > >> i callmanager add asterisk as h323 gateway and also add route
> pattern to
> > >> this gateway
> > >> compile asterisk with h323 support, it will build chan_h323.so,
> add
> > >> callmanager as friend in h323.conf
> > >> in callmanager v4 you can also use SIP trunk between callmanager
> and
> > >> asterisk, but keep in mind, that only g711 codec is supported on
> SIP
> > >> trunk.
> > >> PJ
> > >>
> > >>
> > >> nik600 wrote:
> > >> >> If you have skinny phone registered eg. in ci$co callmanager,
> you
> > >> should
> > >> >> make h323 trunk between asterisk and callmanager.
> > >> >> PJ
> > >> > yes, this is my scenario, sorry for my bad explanation...
> > >> >
> > >> > How can i do that?
> > >> >
> > >> >>
> > >> >>
> > >> >> nik600 wrote:
> > >> >> >
> > >> >> > But how can i do that if the skinny phone isn't registered
> to
> > >> >> Asterisk?
> > >> >> >
> > >> >> > Thanks in advance
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-- 
Jason Parker
Digium