[Asterisk-Users] Dial command to connect two channels and bypass asterisk server

[Nitin Gupta]

thanks for the information Peter, its really helpful. Also I have one more
question - do you have any idea how many such simultaneous calls can an
asterisk server handle (say running od 2.6Ghz, 1GB Ram, fedora machine)?

Thanks,
Nitin


On 2/13/06, Peter Fern <pete at keypoint.com.au> wrote:
>
> You can enable this on a per-peer basis with:
>
> sip peers:
> canreinvite=yes
>
> iax peers:
> notransfer=no
>
> Check the iax.conf.sample and sip.conf.sample files for usage.
>
> Nitin Gupta wrote:
>
> > Hi I was wondering if its possible to make Dial command bridge two
> > channels and after bridging bypass asterisk, so that the voice doesn't
> > need to pass through my asterisk server.
> >
> > For e.g., I have a user dialed in and he verifies himself and then
> > dials an international extension, after the call connects I don't want
> > the call to pass through asterisk server anymore. Is there any command
> > already there for any particular channel type?
> >
> > Thanks,
> > Nitin
> >
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