First of all, sorry for my bad english.
I was stuck in asterisk dial plan like:
I have one asterisk server.
I have 2 users 1001 and 1000
The lab request I need to configure when a user call another user with something like area code. I have to answer it and wait for users next option like press 1 for call, press 2 for voice mail.
Example: when user 1000 call user 1001 with 1231001 i need to forward the call to 1001 when 1000 presss 1
My dial plan config file look like this with opt 1, i’m not finish opt 2 yet because I stuck at option 1
it is the result form my asterisk cli:
Executing [1231000@normaluser:1] Answer("SIP/1001-00000064", "") in new stack
-- Executing [1231000@normaluser:2] Set("SIP/1001-00000064", "Mynumber=[b]1000[/b]") in new stack
-- Executing [1231000@normaluser:3] BackGround("SIP/1001-00000064", "en/vm-instructions") in new stack
-- <SIP/1001-00000064> Playing 'en/vm-instructions.gsm' (language 'en')
== Using SIP RTP CoS mark 5
-- Executing [1@normaluser:1] Dial("SIP/1001-00000065", [b]"SIP/,20,r"[/b]) in new stack
[Apr 6 16:39:02] WARNING[C-0000006a]: app_dial.c:2330 dial_exec_full: Dial argument takes format (technology/resource)
== Spawn extension (normaluser, 1, 1) exited non-zero on 'SIP/1001-00000065'
My problem is why when I call with dial dont have any number, please help me