[Asterisk-Users] Asterisk 1.2beta2 and UIP200
Waldo Rubinstein
waldo at trianet.net
Mon Nov 7 09:15:19 MST 2005
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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