[asterisk-dev] Call Forwarding Detail Needed

[Steve Totaro]

Not sure how to explain this any clearer.  It doesn't.

Arpit Mehta wrote:
> Hi,
>
> I think it should be there in Asterisk. I just want to know how 
> Asterisk implements this and where does it implement it.
>
> Regards
>
> Arpit
>
> On 4/1/07, * Steve Totaro* <stotaro at totarotechnologies.com 
> <mailto:stotaro at totarotechnologies.com>> wrote:
>
>     The scenario you describe is *extremely easy* to setup and test.  I am
>     sure, you will find that there is no loop detection and if you
>     watch the
>     console with any level of verbose, it will continue to scroll very
>     rapidly down the screen.
>
>     Maybe RDNIS could be used to back track one extra level of forwarding,
>     other than that, I do not believe there is any mechanism to
>     prevent this.
>
>     What is your fear or what are you thinking is a problem?  Do you see
>     this causing so much activity that it slows or crashes a server?  Why
>     not try it and see?
>
>     Thanks,
>     Steve
>
>     Arpit Mehta wrote:
>     > Hi,
>     >
>     > Thanks. That was useful. But it is more like calling myself and
>     hence
>     > making a loop.
>     >
>     > I was thinking of how Asterisk detects a loop where more than one
>     > participants are involved. Also whether for detection of the loop,
>     > Asterisk will require to store and receive information from the
>     > calling person that all these numbers have been call forwarded to
>     > since that will be needed to find out whether or not you are call
>     > forwarding to the same person.
>     >
>     > Thanks
>     >
>     > Regards
>     >
>     > Arpit
>     >
>     > On 4/1/07, *Leif Madsen* <leif.madsen at asteriskdocs.org
>     <mailto:leif.madsen at asteriskdocs.org>
>     > <mailto: leif.madsen at asteriskdocs.org
>     <mailto:leif.madsen at asteriskdocs.org>>> wrote:
>     >
>     >     On Sunday 01 April 2007 03:34:53 Arpit Mehta wrote:
>     >     > I just want to know if Asterisk handles a looping case and
>     where
>     >     does it
>     >     > handle that .
>     >     >
>     >     > A--> B --> C--> D --> B
>     >     >
>     >     > If A calls  B ,
>     >     >   B call fwds to C ,
>     >     >   C call fwds to D ,
>     >     >   D call fwds to B . Now there is a loop .
>     >     > How does asterisk prevent this loop? I just to know where
>     (as in
>     >     which
>     >     > module in the Asterisk source code) and how is this prevented
>     >     (that is if a
>     >     > data structure of all the numbers that it has call fwded to is
>     >     passed on to
>     >     > B,C,D so that it can detect a loop)?
>     >     >
>     >     > Thanks. I hope you understand my scenario. Any suggestions
>     are
>     >     welcome.
>     >
>     >     As far I might understand it, Asterisk would basically parse the
>     >     Via: headers
>     >     to determine where the call was coming from, and if it
>     detected a
>     >     loop,
>     >     *should* handle that (this does seem like a loop, and not a
>     spiral).
>     >
>     >     I'll have to defer this to the SIP experts and C experts, but
>     >     basically here
>     >     is the code that chan_sip.c (at line 13185 of my chan_sip.c
>     file)
>     >     that checks
>     >     for the loop:
>     >
>     >             /* Check if this is a loop */
>     >             if (ast_test_flag(&p->flags[0], SIP_OUTGOING) &&
>     p->owner &&
>     >     (p->owner->_state != AST_STATE_UP)) {
>     >                     /* This is a call to ourself.  Send ourselves an
>     >     error code
>     >     and stop
>     >                     processing immediately, as SIP really has no
>     good
>     >     mechanism
>     >     for
>     >                     being able to call yourself */
>     >                     /* If pedantic is on, we need to check the tags.
>     >     If they're
>     >     different, this is
>     >                     in fact a forked call through a SIP proxy
>     >     somewhere. */
>     >                     transmit_response(p, "482 Loop Detected", req);
>     >                     p->invitestate = INV_COMPLETED;
>     >                     sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
>     >                     return 0;
>     >             }
>     >
>     >     Leif Madsen.
>     >
>     >
>     >
>     >
>     > --
>     > Arpit Mehta
>     > Graduate Student
>     > Department of Computer Science
>     > Columbia University
>     >
>     > Tel: 1-646-387-5998
>     >
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>
>
> -- 
> Arpit Mehta
> Graduate Student
> Department of Computer Science
> Columbia University
>
> Tel: 1-646-387-5998
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