[Asterisk-bsd] Re: Sipura SPA 3000 Config...

[Ian Welch]

Thanks for the help Chris.

Still don't know why it didn't work however I did a factory reset. 
(ie: Restored the default settings and all seems to work well now.)

btw:  To factory reset the Sipura 3000 use the IVR interface by dialing.

* * * *  'To get into the IVR
73738# 'Hidden rest option
12#

On Mon, 21 Mar 2005 07:17:08 -0600, Chris Coleman <chrisc at daemonnews.org> wrote:
> Did you put a timeout in your dial line?  You get that error if you 
> misconfigure your dial line.
> 
> -Chris
> 
> On Mar 20, 2005, at 9:41 AM, Ian Welch wrote:
> 
> > This is basically what I had... And incoming calls via the sipura box
> > work fine.  Its just outbound calls.  The error I keep getting is.
> >
> >
> > pbx*CLI>
> >     -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back
> > from X.X.X.X
> >     -- Executing Dial("SIP/203-8bec", "Sip/55555555 at sipura-jack") in 
> > new stack
> >     -- Called 55555555 at sipura-jack
> >     -- Got SIP response 410 "Gone" back from X.X.X.X    --
> > SIP/sipura-jack-0f2b is circuit-busy
> >   == Everyone is busy/congested at this time
> > pbx*CLI>
> >
> >
> > Am I wrong in assuming there should be a register line in the sip.conf
> > config?  Is there no where that I need usernames, and passwords /
> > secrets?
> >
> > Thanks In Advance...
> >
> > On Fri, 18 Mar 2005 09:41:35 +0000, Chris Stenton <jacs at gnome.co.uk> 
> > wrote:
> >>
> >> in my case
> >>
> >> exten => _8.,1,Dial(SIP/${EXTEN:1}@homesipurabox)
> >>
> >> sip.conf
> >>
> >> [homesipurabox]
> >> nat=no
> >> ;reinvite=no
> >> ;canreinvite=no
> >> type=friend
> >> host=192.168.123.19
> >> disallow=all
> >> allow=alaw
> >> allow=ulaw
> >> dtmfmode=rfc2833
> >> port=5061
> >> context=home-in
> >> mailbox=203
> >>
> >> on the sipura under pstn line
> >>
> >> voip to pstn gateway enabled yes
> >> voip caller auth method none
> >> one stage dialing yes
> >> voip caller default dp none
> >>
> >> Chris
> >>
> >>
> >> On Fri, 2005-03-18 at 12:29 +1100, Ian Welch wrote:
> >>> Mornin',
> >>>
> >>> I am pretty new to the Asterisk setup and have it working pretty well
> >>> with 20 sip phones, 2 IAXy boxes and a 4 port digium card. (2 x FXS, 
> >>> 2
> >>> x FXO) I have even reused some of my old sipPhone box's to add my
> >>> house and the bosses place to the PBX.
> >>>
> >>> All is well and Asterisk on FreeBSD seems quite stable in using the
> >>> latest ports.  Anyways I just got a new Sipura SPA 3000.  Seems kinda
> >>> cool and works as any other extention would however I can't seem to
> >>> get it to work as a VOIP -> PSTN gateway.
> >>>
> >>> Don't suppose anyone would know the correct config so I can get
> >>> Asterisk to dial out via this PSTN line?
> >>>
> >>> I assume it is just a stuff up in my config.  Asterisk can't register
> >>> with the sipura spa.
> >>>
> >>> If any one knows how could they give me a register line for sip.conf
> >>> and where do I set this up in the Sipura Web interface?
> >>>
> >>> Any help would be appreciated.
> >>> _______________________________________________
> >>> Asterisk-BSD mailing list
> >>> Asterisk-BSD at lists.digium.com
> >>> http://lists.digium.com/mailman/listinfo/asterisk-bsd
> >> --
> >>
> >> _______________________________________________
> >> Asterisk-BSD mailing list
> >> Asterisk-BSD at lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-bsd
> >>
> > _______________________________________________
> > Asterisk-BSD mailing list
> > Asterisk-BSD at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-bsd
> >
> 
>