[Asterisk-Users] wrong password on authentication for INVITE to '"asterisk"

[chawki hammoud]

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.  


--- Rich Adamson <radamson at routers.com> wrote:

> 
> > i tried to send calls through an asterisk box to a
> > voip provider the calls failed and here what i got
> :
> > 
> > *CLI> Sep 24 11:09:19 WARNING[23356]:
> chan_sip.c:6890
> > handle_response: Forbidden - wrong password on
> > authentication for INVITE to '"asterisk"
> > <sip:asterisk at 195.112.214.99:5070>;tag=as667cb0ae'
> >     -- SIP/call-3f73 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
> > 
> > but when i have tried to send calls using xlite
> > softphone it worked and the calls passed without
> any
> > problems.
> 
> You've made a hell of a lot of assumptions that we
> understand
> your configuration, and we don't.
> 
> What is 195.112.214.99 and 213.61.187.150?
> 
> Is your sip phone registered with asterisk? (what
> does sip show
> peers indicate?)
> 
> Is your sip phone or asterisk registering with your
> sip provider?
> (what does sip show registry indicate?)
> 
> Paste the appropriate sections of sip.conf and
> extensions.conf
> along with some clue what addresses and extensions
> are what.
> 
> 
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