[asterisk-dev] SIP invite to localhost/loopback

[Klaus Darilion]

Geoffrey Mina schrieb:
> 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?

IMO dialing via SIP to localhost is even more ugly than chaining local 
channels. Also, local channels will be removed (masqueraded) on 
"connect" if possible.

regards
klaus

> 
> 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
>>
>>
>