[asterisk-users] Everyone is busy/congested: IP Trunk
Pablo Allietti
pablo at lacnic.net
Fri Oct 26 09:51:33 CDT 2007
On Fri, Oct 26, 2007 at 06:55:12AM -0700, bilal ghayyad wrote:
> Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route to destination)
i think you have the wrong ip information
>
> I established an SIP IP Trunk between Asterisk and
> another softswitch (asterisk registered on the
> softswitch successfully) and I saw this on the
> softswitch.
>
> >From firefly softphone, I was need to do a call to be
> via this softswitch (ofcourse, the softphone will send
> for asterisk and asterisk should route to the
> softswitch based on the extensions.conf
> configurations.
>
> But, always I receive this message (and the call does
> not even reach to the softswitch, it is not sended
> from Asterisk to the softswitch):
>
> Executing [9617565116 at EgyptInternationalVoIP:1]
> Dial("SIP/EgyptOeratorSIP-09f9bed0",
> "SIP/9617565116 at EgyptAlooNet") is new stack
>
> Unable to create channel of type SIP (cause 3 - No
> route to destination)
>
> Everyone is busy/congested at this time (1:0/0/1)
>
> Anyone faced that?
>
> Is it related to a paramater that control number of
> allowed channels per IP trunk? Maybe I have such
> parameters is 0 ? I do not know even if there is such
> parameter.
>
> At the softswitch, I do not see even any attempt
> (nothing related to the dialed number), so why
> Asterisk does not send the called number to the
> softswitch and why asterisk assume there is not
> available channel?
>
> The softphone codec is g729a and the softswitch
> support such codec. Also, if it is a codec matter,
> then call should be send to the softswitch, and the
> softswitch will gives an error related to the codec
> missmatch.
>
> Any help?
>
> Regards
> Bilal Ghayad
>
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.-
Pablo Allietti
E-mail: pablo at lacnic.net | LACNIC
Phone : +598 2 6042222 | http://LACNIC.NET
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