[Asterisk-Users] Asterisk 1.2beta2 and UIP200

[C F]

can you post the sip.conf for that uip200?

On 11/6/05, Waldo Rubinstein <waldo at trianet.net> wrote:
> When I dial the extension, I get this:
>
>      -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20") in new
> stack
>    == Everyone is busy/congested at this time (1:0/0/1)
>
>
> When I do a sip show peer 100074, everything it shows matches the
> results of executing the same sip show peer on * 1.0.9 and 1.2b1,
> except:
>
>    Status       : UNREACHABLE
>
> However, I can make any type of calls from them phone. I can ping the
> phone from the * server. It's just that * 1.2b2 can't reach it, for
> some reason.
>
> Thanks,
> Waldo
>
> On Nov 6, 2005, at 1:37 PM, C F wrote:
>
> > Whats the exact CLI output you are getting when calling that
> > extension?
> >
> > On 11/6/05, Waldo Rubinstein <waldo at trianet.net> wrote:
> >> Nope. It isn't active. I even factory reseted the phone but still the
> >> same. One more piece of information: it works just fine in 1.2b1. I
> >> beginning to think it could be a bug in 1.2b2.
> >>
> >> Any other ideas/suggestions?
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 5, 2005, at 9:10 PM, C F wrote:
> >>
> >>> You sure that the DND (Do Not Disturb) button is not active on the
> >>> UIP200?
> >>>
> >>> On 11/4/05, Waldo Rubinstein <waldo at trianet.net> wrote:
> >>>> I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro
> >>>> phones.
> >>>>
> >>>> All phones register fine with * and I can place outbound calls with
> >>>> no problem.
> >>>>
> >>>> I can call from the X-Pro to any other X-Pro. I can call from
> >>>> UIP200
> >>>> to any other X-Pro. However, the UIP200 cannot receive calls. Every
> >>>> time I call the UIP200, the CLI says Everyone is Busy/Congested and
> >>>> sends the call to voicemail.
> >>>>
> >>>> Everything is in the same network. I have in sip.conf
> >>>> localnet=10.0.10.0/24
> >>>>
> >>>> and in each UIP200 sip profile
> >>>> nat=never
> >>>>
> >>>> What's wrong?
> >>>>
> >>>> I have the same configuration in * 1.0.9 and it works just fine.
> >>>> Could the SIP protocol be broken in 1.2b2?
> >>>>
> >>>> Thanks,
> >>>> Waldo
> >>>>
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