[Asterisk-Users] Re: fail to make call

[unplug]

Hi ...
  In my configuration below, I use realtime architecture in our
system.  I have one device attached to each asterisk server.  There is
no record when I issue "sip show users or sip show registry in CLI.  I
wonder how can I know who is registered in asterisk.  What command is
it?

On 6/20/06, unplug <maillisting at gmail.com> wrote:
> Hi
>   I have the following configuration
>
>          |
> UA1 --|------ asterisk1 -----------------------+
> UA2 --|------ asterisk2 -----------------------+ DB
> UA3 --|------ asterisk3 -----------------------+
> UA4 --|------ asterisk4 -----------------------+
>          |
>
> All UA is located in the same area.  A seperated PC is used as a
> centralized DB for storing a common dial plan, user account and
> register infomration.
> UA1 can make call to UA2,UA3 and UA4.
> UA2 can make call to UA1, UA3 but not UA4.
> UA3 can make call to UA4 but not UA1, UA2
> UA4 failed to make call to all UA.
>
> From the CLI and log below, asterisk shows it can't create channel.
> As I expect, all UA should able to find each other.  However, some of
> them are failed to find others.  I have no idea why they can't find
> each other well.  Is it the configuration problem?  Anyone can help?
>
>     -- Executing Dial("SIP/871966629896-5373", "SIP/871966760539|15")
> Jun 20 17:42:16 NOTICE[23355]: app_dial.c:1040 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 3 - No route to destination)
>   == Everyone is busy/congested at this time (1:0/0/1)
>     -- Executing Hangup("SIP/871966629896-5373", "")
>
> NOTICE[28269] app_dial.c: Unable to create channel of type 'SIP'
> (cause 3 - No route to destination)
>