Article provided by Edwin of Atcom technologies

In rare instances there are occasions where after defining call rules in the IP0x calls do not route where intended. This article provided by Edwin at Atcom details how to troubleshoot the problem.

First, you need to view all of the calling process in the asterisk CLI. To enter the Asterisk CLI, use the putty utility to connect to the IP0x and then run asterisk -vvvvvgrc. Next try to make a call the information similar to below is displayed.

IP04*CLI>
    -- Executing [6001@DLPN_DialPlan1:1] Macro("SIP/6005-01400004", "stdexten|6001|SIP/6001&IAX2/6001") in new stack   
          //6005 uses its dialplan "DLPN_DialPlan1" to call SIp extensions 6001
    -- Executing [s@macro-stdexten:1] Set("SIP/6005-01400004", "__DYNAMIC_FEATURES=") in new stack
[May 26 11:18:54] WARNING[29635]: ast_expr2.fl:407 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input:
= 1
^
[May 26 11:18:54] WARNING[29635]: ast_expr2.fl:411 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source.
    -- Executing [s@macro-stdexten:2] GotoIf("SIP/6005-01400004", "?5:3") in new stack
    -- Goto (macro-stdexten,s,3)
    -- Executing [s@macro-stdexten:3] Dial("SIP/6005-01400004", "SIP/6001&IAX2/6001|20|") in new stack
    -- Called 6001
[May 26 11:18:54] WARNING[29635]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
    -- Got SIP response 486 "Busy Here" back from 192.168.1.111
    -- SIP/6001-0157a9ec is busy
    //extension 6001 is on the phone and return a busy signal to asterisk.
  == Everyone is busy/congested at this time (2:1/0/1)
    -- Executing [s@macro-stdexten:4] Goto("SIP/6005-01400004", "s-BUSY|1") in new stack
    -- Goto (macro-stdexten,s-BUSY,1)
    since 6001 is busy so we go to the voicemail handle part. 
    -- Executing [s-BUSY@macro-stdexten:1] VoiceMail("SIP/6005-01400004", "6001|b") in new stack
    -- Playing 'vm-theperson' (language 'en')
    -- Playing 'digits/6' (language 'en')
    -- Playing 'digits/0' (language 'en')
    -- Playing 'digits/0' (language 'en')
  == Spawn extension (macro-stdexten, s-BUSY, 1) exited non-zero on 'SIP/6005-01400004' in macro 'stdexten'
  == Spawn extension (macro-stdexten, s-BUSY, 1) exited non-zero on 'SIP/6005-01400004'

 
the flow above demonstrates how the call routes in the call process. I've added some comments below to highlight related files in asterisk that affect the call flow.

file [b]/etc/asterisk/users.conf[/b]:
[6005]
context=DLPN_DialPlan1                    //means 6005 use dialrule:DLPN_DialPlan1, all calls from 6005 will follow this rule

file [b]/etc/asterisk/extensions.conf[/b]:
[DLPN_DialPlan1]                              // the dial rule DLPN_DialPlan1 also include the other dial rule in the same extensions file.
include=default
include=parkedcalls
include=conferences
include=ringgroups
include=voicemenus
include=queues
include=voicemailgroups
include=directory

so to trace the calls, you can check the file: extensions.conf, users.conf in gui-->option-->show advance option--> file editor and the debug info from the asterisk -vvvvvgrc
if you have difficullty to debug the info, please send below infomation to my mail address This e-mail address is being protected from spambots. You need JavaScript enabled to view it
1)extensions.conf
2)users.conf
3) debug info in asterisk -vvvgrc
4) the calling process you expect