[asterisk-users] Everyone is busy/congested: IP Trunk

bilal ghayyad bilmar_gh at yahoo.com
Fri Oct 26 13:36:15 CDT 2007


Hi Pablo;

How the IP address will be wrong, and asterisk able to
do registeration on the destination?

If the IP address wrong, so I will not be able to
register on that IP address.

Regards
Bilal

> 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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