[Asterisk-Users] Asterisk 1.2beta2 and UIP200
C F
shmaltz at gmail.com
Mon Nov 7 11:29:15 MST 2005
I guess that somewhere in your settings you have a qualify on, or that
1.2 has it on by default. Do the following:
cd /etc/asterisk
grep ".*qualify.*" ./*
and see the output, if the only line that has qualify is that
qualify=no, then this looks like a bug to me. Please report back.
On 11/7/05, Waldo Rubinstein <waldo at trianet.net> wrote:
> Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1.
> Very strange.
>
> Anyway, thanks.
>
> - Waldo
>
> On Nov 7, 2005, at 10:57 AM, C F wrote:
>
> > The unreachable is the problem. Try adding a qualify=no to that sip
> > entry.
> >
> > On 11/7/05, Waldo Rubinstein <waldo at trianet.net> wrote:
> >> Additionally:
> >>
> >> *CLI> sip show peer 100074
> >>
> >> * Name : 100074
> >> Secret : <Set>
> >> MD5Secret : <Not set>
> >> Context : qa
> >> Subscr.Cont. : <Not set>
> >> Language : en
> >> AMA flags : Unknown
> >> CallingPres : Presentation Allowed, Not Screened
> >> Callgroup :
> >> Pickupgroup :
> >> Mailbox : 211 at 100
> >> VM Extension : asterisk
> >> LastMsgsSent : 0
> >> Call limit : 0
> >> Dynamic : Yes
> >> Callerid : "Waldo Rubinstein" <211>
> >> Expire : 11077
> >> Insecure : no
> >> Nat : No
> >> ACL : No
> >> CanReinvite : No
> >> PromiscRedir : No
> >> User=Phone : No
> >> Trust RPID : No
> >> Send RPID : No
> >> DTMFmode : rfc2833
> >> LastMsg : 0
> >> ToHost :
> >> Addr->IP : 10.0.10.236 Port 5060
> >> Defaddr->IP : 0.0.0.0 Port 5060
> >> Def. Username: 100074
> >> SIP Options : (none)
> >> Codecs : 0x6 (gsm|ulaw)
> >> Codec Order : (ulaw,gsm)
> >> Status : UNREACHABLE
> >> Useragent : Uniden SIP Phone p2 Ver BS4.63
> >> Reg. Contact : sip:100074 at 10.0.10.236:5060
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 6, 2005, at 11:11 PM, C F wrote:
> >>
> >>> 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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> >>
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