[asterisk-users] SIP 482 Loop detected

[Steve Totaro]

On Tue, Jun 23, 2009 at 9:36 AM, jonas kellens <jonas.kellens at telenet.be>wrote:

>  Do you understand what is happening ?
>
>
>  -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-08de0490",
> "SIP/3starsnet/0473775006") in new stack
>     -- Called 3starsnet/0473775006
>     -- SIP/3starsnet-08d70ea8 is making progress passing it to
> SIP/twinkle-08de0490
>     -- Got SIP response 500 "Service Unavailable" back from 85.119.188.3
>     -- SIP/3starsnet-08d70ea8 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
>   == Auto fallthrough, channel 'SIP/twinkle-08de0490' status is
> 'CONGESTION'
>
> My setup :
>
> 3starsNet (85.119.188.3) -- (78.21.63.86)Endian Firewall(192.168.2.1) --
> (192.168.2.2) Asterisk (192.168.1.248) -- (192.168.1.16) twinkle
>
> asterisk*CLI> sip show registry
> Host                            Username       Refresh State
> Reg.Time
> 85.119.188.3:5060    092779077          105 Registered           Tue, 23
> Jun 2009 15:28:14
>
> asterisk*CLI> sip show peers
> Name/username                 Host                Dyn Nat ACL Port
> Status
> twinkle/twinkle                    192.168.1.16     D
> 5060     OK (11 ms)
> grandstream/grandstream  192.168.1.13    D                5060     OK (23
> ms)
> 3starsnet/092779077          85.119.188.3         N           5060     OK
> (21 ms)
>
>
> I don't understand what this sentence means :
> SIP/3starsnet-08d70ea8 is making progress passing it to
> SIP/twinkle-08de0490
>
>
I have seen "Service Unavailable" when trying to set caller ID to a value
that wasn't mine or provisioned to my account.

Setting it to your BTN or a DID through them should be an easy test.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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