[asterisk-users] SIP 482 Loop detected

[jonas kellens]

In sip.conf :

[general]

canreinvite=no                  ; Asterisk by default tries to redirect
the
                                ; RTP media stream (audio) to go
directly from
                                ; the caller to the callee.  Some
devices do not
                                ; support this (especially if one of
them is behind a NAT).
                                ; The default setting is YES. If you
have all clients
                                ; behind a NAT, or for some other reason
wants Asterisk to
                                ; stay in the audio path, you may want
to turn this off.

[3starsnet]
type=peer
host=85.119.188.3
username=username
secret=password
fromuser=username
dtmfmode=rfc2833
canreinvite=no
insecure=port,invite
qualify=yes
nat=yes
disallow=all
allow=gsm
allow=alaw
amaflags=billing
accountcode=3starsnet-out




On Tue, 2009-06-23 at 08:50 -0400, Steve Totaro wrote:

> 
> 
> 
> On Tue, Jun 23, 2009 at 8:09 AM, jonas kellens
> <jonas.kellens at telenet.be> wrote:
> 
>             -- Executing [0473775006 at intern:1]
>         NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new
>         stack
>             -- Executing [0473775006 at intern:2]
>         Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in
>         new stack
>             -- Called 3starsnet/0473775006
>             -- Got SIP response 482 "Loop Detected" back from
>         85.119.188.3
>             -- Now forwarding SIP/twinkle-088e6ea8 to
>         'Local/0473775006 at default' (thanks to SIP/3starsnet-088e2ca0)
>         [Jun 23 07:29:43] NOTICE[15896]: chan_local.c:504 local_call:
>         No such extension/context 0473775006 at default while calling
>         Local channel
>         [Jun 23 07:29:43] NOTICE[15896]: app_dial.c:561
>         wait_for_answer: Failed to dial on local channel for call
>         forward to 'Local'
>         
>         
>         Twinkle is my softphone, who is registered with my
>         Asterisk-server.
>         
>         85.119.188.3 is the IP-address of the SIP-server of my
>         VoIP-provider.
>         
>         SIP/3starsnet is the SIP-channel that corresponds with a peer
>         defined in sip.conf to place calls to my SIP-provider
>         3StarsNet.
>         
>         So twinkle calls my Asterisk-server, which calls my
>         SIP-provider...
>         
>         What is this "Loop detected" ??
>         
>         Jonas.
>         
>         
>         
> 
> I have seen this with canreinvite=yes set when it should not be. 
> 
> 
> 
> -- 
> Thanks,
> Steve Totaro 
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
> 
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