[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

[Face]

On Mon, Nov 19, 2012 at 3:51 PM, Joshua Colp <jcolp at digium.com> wrote:
> Face wrote:
>>
>> Hello,
>
>
> Hola,
>
>
>> After Upgrade to Asterisk 11.1.0-rc1 I keep getting
>>
>>    == Using SIP VIDEO TOS bits 136
>>    == Using SIP VIDEO CoS mark 6
>>    == Using SIP RTP TOS bits 184
>>    == Using SIP RTP CoS mark 5
>>      -- Executing [603 at DLPN_AlDimnaDialPlan:601]
>> Dial("SIP/601-00000002", "SIP/603") in new stack
>> [Nov 16 06:42:33] WARNING[15547][C-00000004]: app_dial.c:2433
>> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
>> Subscriber absent)
>>    == Everyone is busy/congested at this time (1:0/0/1)
>>      -- Auto fallthrough, channel 'SIP/601-00000002' status is
>> 'CHANUNAVAIL'
>>
>> and would not go to voicemail?
>
>
> Unfortunately without more information (dialplan involved, complete console
> output, sip show peer 603) it's impossible to fathom any potential reason
> why this is occurring. I suspect that's why nobody has responded to you
> until now. If you can provide that information I'm sure we can all help to
> determine if there really is an issue at work here!
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
> --
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Well, thanks for responding. I went back to 10.10.0 and things seem to
be working fine now!

-- 
Sincerely,
falazemi