[asterisk-dev] SIP invite to localhost/loopback

[Kaloyan Kovachev]

On Wed, 13 Jan 2010 22:26:10 -0500, Geoffrey Mina wrote
> I am not using Local because this action is being performed BY a Local
> channel.  I wasn't comfortable nesting local channels in this
> environment since that is largerly untested methodology... And this is
> a high profile platform.
> 
> I figured it would be a more 'stable' solution to simply dial SIP into
> localhost.  I suppose this is not possible in asterisk though, huh?
> 

You may try dialing to localhost via IAX

> On 1/12/10, Kirill 'Big K' Katsnelson <kkm at adaptiveai.com> wrote:
> > On 100111 1828, Geoffrey Mina wrote:
> >> It seems as though checking of the direction of the message combined
> >> with the totag and fromtag of the SIP INVITE, Asterisk should be able
> >> to deal with this scenario without raising a "482 Loop Detected".
> >
> > I *think* you can force Asterisk into tag compliance by enabling
> > pedantic SIP checking. Tag checks are intentionally lax without that
> > option having been set, which is the default. That's purely a
> > speculation, however.
> >
> > Practically, you should go Local/ instead, unless you have some very
> > specific application in mind.
> >
> >   -kkm
> >
> >
> 
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