[asterisk-users] SIP detects loop when forwarding to voicemail
Tomasz Zieleniewski
tzieleniewski at gmail.com
Thu Nov 29 12:46:16 CST 2007
Thank you!
It would be greate to have these feature set as a parameter in sip.conf
cheers
tomasz
On Nov 28, 2007 2:38 PM, Philipp Kempgen <philipp.kempgen at amooma.de> wrote:
> Tomasz Zieleniewski wrote:
> > How does asterisk detect the loop.
> > What are the criteria here.
> > What do I need to change in the SIP message so
> > that asterisk will not consider it looped??
>
> > On Nov 23, 2007 4:03 PM, Tomasz Zieleniewski <tzieleniewski at gmail.com>
> > wrote:
> >
> >> hi,
> >>
> >> I use asterisk as a gateway which forwards external calls from pstn to
> >> my internal sip network.
> >> all sip signaling is passed to sip proxy.
> >> I also use asterisk as a voicemail server.
> >> everything works well when calls are passed to asterisk from local
> >> network.
> >> but when calls are forwarded from asterisk to sip proxy and then sip
> >> proxy decides to pass it back to asterisk
> >> waorking as a voicemail server
> >> asterisk complains about the loop and returns 482 response.
> >> Can it be somehow reconfigured??
>
> See
> http://bugs.digium.com/view.php?id=7403
> and look for this code in chan_sip.c:
> ---cut---
> /* 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);
> sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
> return 0;
> }
> ---cut---
>
> There's no way to configure the loop detection but you could
> remove the code to disable loop detection.
>
> Grüße,
> Philipp Kempgen
>
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