Face | 16 Nov 05:45 2012
Picon

Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

Hello,

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?

--

-- 
Sincerely,

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
(Continue reading)

Joshua Colp | 19 Nov 13:51 2012

Re: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

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,

--

-- 
(Continue reading)

Face | 20 Nov 02:01 2012
Picon

Re: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

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
(Continue reading)

Joshua Colp | 20 Nov 14:46 2012

Re: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

Face wrote:
>
> Well, thanks for responding. I went back to 10.10.0 and things seem to
> be working fine now!

This is certainly good to know but I'd like to know why upgrading to 11 
did not seem to work for you. This is the first case since it's been out 
where it doesn't appear to have been smooth. Would you be willing to 
provide the information I asked about from a running 11 instance?

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

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Face | 21 Nov 07:24 2012
Picon

Re: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

I upgrading to 11 because I want to use the "MessageSend" command from the AMI, ver 10 dose not have "MessageSend" In the list of commands. Unfortunately I remove  ver 11 and I dont think I can provide the information you asked.


On Tue, Nov 20, 2012 at 4:46 PM, Joshua Colp <jcolp <at> digium.com> wrote:
Face wrote:

Well, thanks for responding. I went back to 10.10.0 and things seem to
be working fine now!

This is certainly good to know but I'd like to know why upgrading to 11 did not seem to work for you. This is the first case since it's been out where it doesn't appear to have been smooth. Would you be willing to provide the information I asked about from a running 11 instance?

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

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



--
Sincerely,
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Gmane