[Asterisk-biz] need consultant

[jonny hashem]

i have an asterisk box (195.112.214.99) with this
configuration:

                      sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=XXXXXXX
secret=XXXXXX
allow=all

                    extensions.conf 

[call]
exten => _00.,1,Dial,SIP/callshop/${EXTEN}

and when i try to send calls to the voip provider
(callshopcompany "213.61.187.150") i got these
messages:

*CLI> dial 0017046872001 at call
    -- Executing Dial("OSS/dsp",
"SIP/callshop/0017046872001") in new stack
    -- Called callshop/0017046872001
*CLI> Sep 24 14:16:45 WARNING[22295]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to '"asterisk"
<sip:asterisk at 195.112.214.99:5070>;tag=as4cda63c2'
    -- SIP/callshop-f613 is circuit-busy
  == Everyone is busy/congested at this time
    -- Got SIP response 481 "Call Leg Does Not Exist"
back from 213.61.187.150
Sep 24 14:16:58 WARNING[22295]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'call'
 << Hangup on console >>

but when ive tried it on xlite in the same
configuration to send calls to the same company it
worked and the calls passed without any problems.

so whats the problem here,why the call goes well using
xlite and fails using asterisk despite they have the
same configuration.  

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