[asterisk-users] SIP 482 Loop detected

[jonas kellens]

Thank you for your answer.

Could you explain why the call fails ?

Connected to Asterisk 1.4.25.1 currently running on asterisk (pid =
17936)
Verbosity is at least 25
Core debug is at least 5
    -- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-0a0567f8",
"conversation to GSM") in new stack
    -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-0a0567f8",
"SIP/3starsnet/0473775006") in new stack
    -- Called 3starsnet/0473775006
    -- SIP/3starsnet-0a05c038 is making progress passing it to
SIP/twinkle-0a0567f8
REGISTER attempt 1 to 092779077 at 85.119.188.3
REGISTER attempt 2 to 092779077 at 85.119.188.3
Really destroying SIP dialog
'1d9d81a23ac05e735663b37b750a640f at 192.168.2.2' Method: OPTIONS
    -- Got SIP response 500 "Service Unavailable" back from 85.119.188.3
    -- SIP/3starsnet-0a05c038 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/twinkle-0a0567f8' status is
'CONGESTION'
Really destroying SIP dialog
'340811e66bc43ba36fb5d507066fc1a7 at 192.168.2.2' Method: INVITE
Really destroying SIP dialog 'xfdsxekzwoxcohv at localhost' Method: ACK


Jonas.


On Wed, 2009-06-24 at 02:47 +1000, Rob Hillis wrote:

> jonas kellens wrote:
> > Do you understand what is happening ?
> > I don't understand what this sentence means :
> > SIP/3starsnet-08d70ea8 is making progress passing it to
> > SIP/twinkle-08de0490 
> Pretty simple really.  Your SIP trunk 3starsnet is making progress with
> the call and Asterisk is passing that message on to SIP/twinkle. 
> Entirely normal.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090623/505e2603/attachment.htm