[asterisk-users] REALY strange issue with making calls biside 2 phones

[Matiss Jekabsons]

Thats my issue, i hope someone could suggest something:

Phone A -> Phone B



== Using SIP RTP CoS mark 5

     -- Executing [000001 at 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 at 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 at default:2] Hangup("SIP/000001-00000078", "")  
in new stack

   == Spawn extension (default, 000000, 2) exited non-zero on  
'SIP/000001-00000078'



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Best regards
Matiss Jekabsons
Procerto Ltd.