[asterisk-users] SIP 482 Loop detected

[jonas kellens]

Calls succeed now because I have added in sip.conf :

[3starsnet]
type=peer
host=85.119.188.3
username=username
secret=****
fromuser=username
fromdomain=sip.3starsnet.com

What does this 'fromdomain'-parameter do ?? So I can understand why this
is so important.

Jonas.


On Tue, 2009-06-23 at 13:13 -0400, Steve Totaro wrote:

> 
> 
> 
> 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'
> 
> 
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