[asterisk-users] Everyone is busy/congested: IP Trunk
joakimsen at gmail.com
joakimsen at gmail.com
Mon Oct 29 16:54:02 CDT 2007
No:
register => abc:123 at xyz
[peer]
host=zzz
Its possible to make mistakes and typos you know. Maybe you can post
your config file and we can help you.
On 10/26/07, bilal ghayyad <bilmar_gh at yahoo.com> wrote:
> 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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