[Asterisk-Users] Asterisk 1.2beta2 and UIP200
C F
shmaltz at gmail.com
Mon Nov 7 16:05:47 MST 2005
First off, if they are on the same network without any nat, then it is
not needed at all. Since this works well with pre 1.2b2 I would say
you should open up a but on the bug tracker at:
bugs.digium.com.
I did not yet update to 1.2bx so I have no way of confirming this.
Thank You.
On 11/7/05, Waldo Rubinstein <waldo at trianet.net> wrote:
> Ok.
>
> The keepalives work for other phones, but not the UIP200. I have a
> bunch of X-Lites, X-Pros, SPA-841s, and UIP200s. It works fine in all
> but the UIP200 (only in 1.2b2).
>
> As far as your questions:
>
> 1) They are on the same network and same netmask
> 2) They are not natted.
>
> Let me know what you find.
>
> Thanks,
> Waldo
>
> On Nov 7, 2005, at 4:58 PM, C F wrote:
>
> > 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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> >>>>>>>>>>>>
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> >>>>>>>>>>>>
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> >>
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