> 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 [EMAIL PROTECTED]
> -- 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"
Sure looks like an authentication problem. If you are absolutely
positive you have no typos in the username/secret, then have you
tried the suggestions from
/usr/src/asterisk/configs/sip.conf.sample
that suggests:
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;usereqphone=yes ; This provider requires ";user=phone" on URI
;call-limit=5 ; permit only 5 simultaneous outgoing calls to
this peer
Looks like a couple more parameters might be needed.
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