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@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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