I have set up extension.conf and sip.con with default
parameter of UNIVOICE server, but Asterisk show this
message when I call a number:
Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899
create_addr: No such host: univoice,Ttr
Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/100-2331", "") in new
stack
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