[Asterisk-Dev] SIP REINVITE

[Olle E. Johansson]

BJ Weschke wrote:
>  Server A (IP 192.168.1.1) 
>  Server B (IP(s) 192.168.1.2 [actual] 192.168.1.3 [vip])
>  Server C (IP(s) 192.168.1.4)
> 
>  All servers are Asterisk installs. All servers have SIP canreinvite=yes.
> 
>  Server A calls Server B on his VIP. The call sets up fine, but the
> 3rd of 4th step in the dial plan is to then transfer that call on to
> Server C. Server B dials server C and then begins to attempt a native
> bridge between Server A and C. Server A responds back with "SIP/2.0
> 482 Loop Detected" assumably because the man in the middle has
> different terminating/originating IP addresses and has sent an
> improper invite back to A to start the briding process.
Can you send me a packet trace of this?

>  Does ANTHM's patch from a few weeks back to chan_sip fix this
> problem, or is this still a "live" issue? If it is patched, who needs
> the patch in the scenario above? Just server B? or Servers A and C
> too?
I haven't seen the loop detected issue, but understand where it's coming
from. Anthm's patch is more to be seen as a proof-of-concept than
something you want to use. I'm trying to continue the work based on his
patch, but it will require a lot of changes to chan_sip.

I'm glad to see another person wanting to transfer calls from Asterisk
to another SIP domain - I just had a question from a core developer on
the theme "why would anyone want to do that?"... So I needed your mail
to prove that's it is not only me and my customers that need that function.

Digging into how chan_sip handles transfers I'm amazed that it work with
anything... ;-)

/Olle