[asterisk-users] SIP detects loop when forwarding to voicemail

[Philipp Kempgen]

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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