[Asterisk-Users] Asterisk 1.2beta2 and UIP200

C F shmaltz at gmail.com
Mon Nov 7 14:58:03 MST 2005


If this is the case. then we now know what the problem is. The
keepalives from asterisk to the phones were not working in 1.2b2. The
question now is why?
Please work with this so that we can troubleshoot this to see if it's
a bug with 1.2b2 or not.
1. Is the UIP200 on the same subnet as asterisk?
2. if not, is the UIP200 or asterisk natted?

In the meantime I will try to see on my 1.0.9 install if it works or
not with UIP200 phones.
Thank You.

On 11/7/05, Waldo Rubinstein <waldo at trianet.net> wrote:
> I do have qualify=yes pretty much in all my sip entries. I just
> changed all the entries where I have a UIP200 to qualify=no and now
> they all work. The funny thing is that it worked with qualify=yes in
> 1.0.9 and 1.2b1
>
> Thanks,
> Waldo
>
> On Nov 7, 2005, at 1:29 PM, C F wrote:
>
> > 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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