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@EgyptInternationalVoIP:1]
Dial("SIP/EgyptOeratorSIP-09f9bed0",
"SIP/9617565116@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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