Thats my issue, i hope someone could suggest something:
Phone A -> Phone B
== Using SIP RTP CoS mark 5
-- Executing [000001@default:1] Dial("SIP/000000-00000076",
"SIP/000001") in new stack
== Using SIP RTP CoS mark 5
-- Called 000001
-- SIP/000001-00000077 is ringing
-- SIP/000001-00000077 answered SIP/000000-00000076
-- Locally bridging SIP/000000-00000076 and SIP/000001-00000077
== Spawn extension (default, 000001, 1) exited non-zero on
'SIP/000000-00000076'
Phone B -> phone A
== Using SIP RTP CoS mark 5
-- Executing [000000@default:1] Dial("SIP/000001-00000078",
"SIP/000000") in new stack
[Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [000000@default:2] Hangup("SIP/000001-00000078", "")
in new stack
== Spawn extension (default, 000000, 2) exited non-zero on
'SIP/000001-00000078'
--
--
Best regards
Matiss Jekabsons
Procerto Ltd.
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