I (601) call one of my users (8862), after one minute I try to call him again and get "Unable to create channel of type 'SIP' "

sip show peers does not list him.

I cannot figure out why this happens and more important how I can fix it.



-- Executing Dial("SIP/601-0f22", "SIP/8862|60|tr") in new stack
Apr 21 15:30:47 NOTICE[14706]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing NoOp("SIP/601-0f22", "CONGESTION") in new stack
-- Executing VoiceMail("SIP/601-0f22", "u8862") in new stack
-- Playing '/var/spool/asterisk/voicemail/others/8862/unavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
== Spawn extension (VoIP_Phone, 8862, 3) exited non-zero on 'SIP/601-0f22'



bye

Ronald

_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to