Hello, I've got my SIP trunk registered correctly (according to asterisk), but when I make a call TO my number from my cellphone, I get a message saying "The party you are trying to reach is not available" and asterisk -r shows this:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
When I make an outgoing call from my soft phone registered to my asterisk I get "Your call cannot be completed as dialled, please check the number & dial again." asterisk -r shows this:
== 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 [0800000000@from-internal:1] ResetCDR("SIP/001-00000016", "") in new stack
-- Executing [0800000000@from-internal:2] NoCDR("SIP/001-00000016", "") in new stack
-- Executing [0800000000@from-internal:3] Progress("SIP/001-00000016", "") in new stack
-- Executing [0800000000@from-internal:4] Wait("SIP/001-00000016", "1") in new stack
-- Executing [0800000000@from-internal:5] Progress("SIP/001-00000016", "") in new stack
-- Executing [0800000000@from-internal:6] Playback("SIP/001-00000016", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/001-00000016> Playing 'silence/1.gsm' (language 'en')
-- <SIP/001-00000016> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
-- <SIP/001-00000016> Playing 'check-number-dial-again.gsm' (language 'en')
-- Executing [0800000000@from-internal:7] Wait("SIP/001-00000016", "1") in new stack
-- Executing [0800000000@from-internal:8] Congestion("SIP/001-00000016", "20") in new stack
== Spawn extension (from-internal, 0800000000, 8) exited non-zero on 'SIP/001-00000016'
-- Executing [h@from-internal:1] Hangup("SIP/001-00000016", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/001-00000016'
Any ideas why this isn't working? My knowledge level with Asterisk/PBX in general is "knows enough to be dangerous" so step by step insturctions on how to nail down what's wrong with it would be appreciated.