[Asterisk-Dev] Re: 482 Loop Detected problem

[Olle E. Johansson]

Doug Meredith wrote:
> Charles Huang <dev.charleshuang at gmail.com> wrote:
> 
> 
>>Hi, all
>>
>>Can anyone give me a hint how to disable "482 Loop Dection" from the
>>"chan_sip.c".
>>
>>Since I am using the Asterisk as a pure gateway, therefore when calls coming
> 
>>from my T1 PSTN and land on any of my SIP phones with Call forword to PSTN
> 
>>number, then it will give me the "482 Loop Detected" fail.
> 
> 
> This behavior has existed for a long time.  I would argue this is a
> bug.  Certainly it is from a SIP point of view.  I guess from a
> monolithic PBX point of view it isn't a bug.
It might be a bug. That's why I asked for more detailed information.

> 
> I haven't seen any sign that anyone is too interested in fixing this,
????? What?
I've been working hard on improving our support of SIP so that we can
avoid these kind of things to happen. There are still situations where
it happens and I am eager to find those. One of the fixes in 1.2 is a
much improved support of SIP tags, so if you turn on pedantic, chances
are very low that this will happen without a cause.

> but if you look in the code, there is a comment that says something
> along the lines of "SIP doesn't permit this".  This certainly isn't
> true, but maybe they meant the SIP channel, not the SIP standard.
Yes, it's completely wrong, but one of Marks very old comments that is
still in there for historical reasons.

> 
> It appears that there is a simple "if" statement in the code that does
> this check and reports the error.  You could try simply commenting out
> the check and see what happens.  Someone previously mentioned that
> they tried this and got a core dump, but it might still be worth
> looking at.

That is a very dangerous advice. Let's get a call log and analyze
what's really going on, what kind of packet the forwarding phone is sending.

Let's work on this together.

/O