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@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,
--
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