[asterisk-dev] SIP invite to localhost/loopback
Geoffrey Mina
geoffreymina at gmail.com
Wed Jan 13 21:26:10 CST 2010
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?
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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