[Asterisk-biz] need consultant

[Mark Edwards]

Seeing as you have posted this to -biz I am assuming you want to pay for
this to get fixed.

Post this message to -users and you might be able to get this fixed for
free.

that'll be $100 please.

Mark.

On 9/25/05, jonny hashem <jonnyhashem at yahoo.com> wrote:
>
> i have an asterisk box (195.112.214.99 <http://195.112.214.99>) with this
> configuration:
>
> sip.conf
> [callshop]
> type=peer
> host=sip.callshopcompany.com <http://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 <http://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 <http://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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>



--
regards,

Mark P. Edwards
FWD: 667917
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