[Asterisk-Users] Asterisk 1.2beta2 and UIP200
Waldo Rubinstein
waldo at trianet.net
Mon Nov 7 12:37:37 MST 2005
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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