Asterisk did not work if configure from freepbx

Asterisk Asterisk Support
1

I’ve install asterisk (latest version). configure trough *.conf and works fine.
Then i install freepbx, installation runs fine and i could configure trough web.
But what i don’t understand, when i add extension trough freepbx, i got message “Call failed: Service Unavailable” (using x-lite). which i dont get it if i add extension trough sip.conf and extension.conf.

Simple, i can make a phone call (using X-Lite) if i add extension trough extension.conf but it won’t work if i add trough freepbx.

Any ide where i have to check. ?? i try to search but its getting too long.

2

Here are the message i got from asterisk console.

– Registered SIP ‘239’ at 10.1.8.239 port 52438 expires 3600
– Executing Macro(“SIP/239-09d73608”, “exten-vm|novm|239”) in new stack
– Executing Macro(“SIP/239-09d73608”, “user-callerid”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0?report”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0?start”) in new stack
– Executing Set(“SIP/239-09d73608”, “REALCALLERIDNUM=239”) in new stack
– Executing NoOp(“SIP/239-09d73608”, “REALCALLERIDNUM is 239”) in new stack
– Executing Set(“SIP/239-09d73608”, “AMPUSER=239”) in new stack
– Executing Set(“SIP/239-09d73608”, “AMPUSERCIDNAME=239”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0?report”) in new stack
– Executing Set(“SIP/239-09d73608”, “CALLERID(all)=239 <239>”) in new stack
– Executing NoOp(“SIP/239-09d73608”, “Using CallerID “239” <239>”) in new stack
– Executing Set(“SIP/239-09d73608”, “FROMCONTEXT=exten-vm”) in new stack
– Executing Set(“SIP/239-09d73608”, “VMBOX=novm”) in new stack
– Executing Set(“SIP/239-09d73608”, “EXTTOCALL=239”) in new stack
– Executing Set(“SIP/239-09d73608”, “CFUEXT=”) in new stack
– Executing Set(“SIP/239-09d73608”, “RT=”) in new stack
– Executing Macro(“SIP/239-09d73608”, “record-enable|239|IN”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0 > 0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing AGI(“SIP/239-09d73608”, “recordingcheck|20060913-120342|1158123822.11”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/239-09d73608”, “No recording needed”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0?dolocaldial|1”) in new stack
– Executing Macro(“SIP/239-09d73608”, “dial||tr|239”) in new stack
– Executing AGI(“SIP/239-09d73608”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
– AGI Script dialparties.agi completed, returning 0
– Executing NoOp(“SIP/239-09d73608”, “Returned from dialparties with no extensions to call”) in new stack
– Executing NoOp(“SIP/239-09d73608”, "DIALSTATUS is ") in new stack
– Executing GosubIf(“SIP/239-09d73608”, “0?docfu|1”) in new stack
– Executing NoOp(“SIP/239-09d73608”, “Voicemail is novm”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “1?s-|1”) in new stack
– Goto (macro-exten-vm,s-,1)
– Executing Congestion(“SIP/239-09d73608”, “”) in new stack
== Spawn extension (macro-exten-vm, s-, 1) exited non-zero on ‘SIP/239-09d73608’ in macro ‘exten-vm’
== Spawn extension (macro-exten-vm, s-, 1) exited non-zero on ‘SIP/239-09d73608’

3

– Executing Set(“SIP/239-09d73608”, “FROMCONTEXT=exten-vm”) in new stack
– Executing Set(“SIP/239-09d73608”, “VMBOX=novm”) in new stack
– Executing Set(“SIP/239-09d73608”, “EXTTOCALL=239”) in new stack

I had this same issue a few times with my SPA942 phones when I first set them up, make sure you enable voicemail and directory services for the extension. I think this is related directly to FreePBX, however do not know if this is your problem fixer.

4

Doesn’t work for me. I use PC to PC in this case, both using x-lite.
It works properly if the users and extensions registered trough *.conf.

but if i add trough freepbx. it wont work at all.

I don’t use any hardware (zaptel and so on) but i don’t think this is an issue. still wonder why…

any clue??