[Serusers] loop call with asterisk

[Richard]

Hi Bogdan,

* keeps the state of a call. I haven't gone through the details of the code,
but it seems to use callid to identify the call. It is an option to use tag
for extra checking.

What I probably need is to treat any new SIP INVITE to * as a new call, just
like what cisco does. With a good dial plan, this new call will never be
sent back to ser. Even if it does, ser will treat this as a loop because I
use record route. So loop avoidance is not a big issue.

The problem is how to identify an INVITE retransmission and subsequent 200
OK, ACK, re-INVITE, CANCEL and BYE messages. How to find out if a message is
for which call?

Thanks,
Richard



> So, depends a lot of what *  consider a loop (how exactly does this) -
> because, as you said, all important hdrs are the same. Btw, is at least
> RURI changing between  outgoing and returning call ( on *) ?
> The worst solution will be insert a very simple B2BUA in the middle to
> change at least some tags. But is ugly :-(
> 
> regards,
> Bogdan
> 
> 
> Richard wrote:
> 
> >Hi,
> >
> >I am trying to use * as a PSTN gateway. When a call comes in from PSTN,
> it
> >is forwarded to ser. If for some reason (e.g. call being forwarded back
> to
> >PSTN), ser will just do a record route and send it back to *. When * gets
> >this call, it thinks it is a loop and drop the call. I know that this
> >question has been raised before. I'd like to see, just from sip
> respective,
> >what's the theoretical way to solve it? Since all major fields are the
> same,
> >e.g. fromuri, touri, cseq, callid, what is the right way to detect loop
> in
> >SIP in this case? Btw, cisco router doesn't have this problem.
> >
> >Thanks,
> >Richard
> >
> >
> >
> >
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> >
> >