Cannot make outbound call

Using FreePBX 12.0.1rc5, when I try to make my first outbound call, it fails. The output from the CLI is:

- Added contact 'sip:[email protected]:5060' to AOR '201' with expiration of 60 seconds
- Executing [[email protected]:1] ResetCDR("PJSIP/201-00000002", "") in new stack
- Executing [[email protected]:2] NoCDR("PJSIP/201-00000002", "") in new stack
- Executing [[email protected]:3] Progress("PJSIP/201-00000002", "") in new stack
- Executing [[email protected]:4] Wait("PJSIP/201-00000002", "1") in new stack
- Executing [[email protected]:5] Progress("PJSIP/201-00000002", "") in new stack
- Executing [[email protected]:6] Playback("PJSIP/201-00000002", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
- <PJSIP/201-00000002> Playing 'silence/1.ulaw' (language 'en')
- <PJSIP/201-00000002> Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
- <PJSIP/201-00000002> Playing 'check-number-dial-again.ulaw' (language 'en')
- Executing [[email protected]:7] Wait("PJSIP/201-00000002", "1") in new stack
- Executing [[email protected]:8] Congestion("PJSIP/201-00000002", "20") in new stack
  == Spawn extension (from-internal, 9995551212, 8) exited non-zero on 'PJSIP/201-00000002'
- Executing [[email protected]:1] Hangup("PJSIP/201-00000002", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/201-00000002'

I’m a total newbie with Asterisk. The provider is Flowroute. What have I done wrong?

Thanks Lars

Doesnt look like you have any outbound routes or trunks setup. Or maybe just the outbound routes.

Thanks tm, you’re right. I set one up, dialed out, got a quick busy signal and got the following output:

Connected to Asterisk 12.4.0 currently running on localhost (pid = 2113)
-- Executing [[email protected]:1] ResetCDR("PJSIP/201-00000003", "") in new stack
-- Executing [[email protected]:2] NoCDR("PJSIP/201-00000003", "") in new stack
-- Executing [[email protected]:3] Progress("PJSIP/201-00000003", "") in new stack
-- Executing [[email protected]:4] Wait("PJSIP/201-00000003", "1") in new stack
-- Executing [[email protected]:5] Progress("PJSIP/201-00000003", "") in new stack
-- Executing [[email protected]:6] Playback("PJSIP/201-00000003", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <PJSIP/201-00000003> Playing 'silence/1.ulaw' (language 'en')
-- <PJSIP/201-00000003> Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
-- <PJSIP/201-00000003> Playing 'check-number-dial-again.ulaw' (language 'en')
-- Executing [[email protected]:7] Wait("PJSIP/201-00000003", "1") in new stack
-- Executing [[email protected]:8] Congestion("PJSIP/201-00000003", "20") in new stack
  == Spawn extension (from-internal, 9995551212, 8) exited non-zero on 'PJSIP/201-00000003'
-- Executing [[email protected]:1] Hangup("PJSIP/201-00000003", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/201-00000003'

What can I check now?

Dial plan is a possibility, your dial plan needs to be setup to route the call to the trunk.