[asterisk-dev] Stuck in Asterisk SIP response 482

[Kevin P. Fleming]

On 07/02/2011 10:36 AM, Matiss at Jekabsons.lv wrote:
> Hallo all!
>
> I am working here on big project to make large Asterisk PBX with
> WEBinterface user self registering to use it @my city as local
> comunication phones.
>
> 1. i have managed to install it on Ubuntu 11.04 server
> 2. i have managet it to get running
> 3. i even have managet to get it to call thru TRUNK to external numbers
>
> but then i have stucked. Sometimes, not allways there was an error that
> loop detected... or unable to create chan - reason unknown. I have
> started to delete anything that could be wrong. Now i have a really
> basic SIP.conf and Extensions.conf and permanent error when calling out:
>
> == Using SIP RTP CoS mark 5
> -- Called 010010
> -- Got SIP response 482 "Loop Detected" back from 0.0.0.0:5060
> -- SIP/010010-000007c8 is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
> -- Auto fallthrough, channel 'SIP/000000-000007c7' status is 'CONGESTION'
>
>
> P.S.
> user configuration it with MySQL backend.

This is not really a development question, and should probably be posted 
to the asterisk-users list instead... but as a starting point, you need 
to try to find out why you are trying to place a SIP call to IP address 
"0.0.0.0". That can't be what you really wanted to do :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org