[Asterisk-Users] Asterisk 1.2beta2 and UIP200

Anton Krall akrall-lists at intruder.com.mx
Mon Nov 7 12:45:04 MST 2005


Some ATAs do not like the qualify, I have some MTA102 and that's the case
with those, if I enable qualify, the ata doesn't work with asterisk, if I
disable qualify, the ATA works without problems. 

|-----Original Message-----
|From: asterisk-users-bounces at lists.digium.com 
|[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of C F
|Sent: Monday, November 07, 2005 9:57 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200
|
|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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