[SOLVED] Cannot Make Outbound or Inbound Calls On FreePBX 2.10.0.4

Hello Everyone,

I have spent too much time on this so it’s time to post on the forum. When I make an outbound call I get the Asterisk recording: “Your call cannot be completed as dialed, please check the number and dial again”. When I make an inbound call from my cell phone to my VOIP number I get the AT&T operator recording that says: “The number you dialed is not a working number please check the number and dial again”. All of my Primary and Secondary Trunks are in the green and registered as are the Contact IP and Network IP. The Firewall Test has passed as well. I have routed my one DID to my extension (I’m using Twinkle which registers just fine). I set the Dialed Number Manipulation Rules using the Wizard (Lookup numbers for local trunk 7-digit dialing) which was successfully completed for both Primary and Secondary Trunks. I’m sure this is a simple fix but for the life of me I can’t figure it out. Here is the tail of the Asterisk’s log when I try to make a call from Twinkle:

[2012-04-04 12:07:24] VERBOSE[-1] chan_sip.c: – Unregistered SIP ‘101’
[2012-04-04 12:10:44] VERBOSE[-1] chan_sip.c: – Registered SIP ‘101’ at 192.168.1.112:5060
[2012-04-04 12:10:44] NOTICE[-1] chan_sip.c: Peer ‘101’ is now Reachable. (102ms / 2000ms)
[2012-04-04 12:11:05] VERBOSE[-1] netsock2.c: == Using SIP RTP TOS bits 184
[2012-04-04 12:11:05] VERBOSE[-1] netsock2.c: == Using SIP RTP CoS mark 5
[2012-04-04 12:11:05] VERBOSE[-1] pbx.c: – Executing [[email protected]:1] ResetCDR(“SIP/101-0000002a”, “”) in new stack
[2012-04-04 12:11:05] VERBOSE[-1] pbx.c: – Executing [[email protected]:2] NoCDR(“SIP/101-0000002a”, “”) in new stack
[2012-04-04 12:11:05] VERBOSE[-1] pbx.c: – Executing [[email protected]:3] Progress(“SIP/101-0000002a”, “”) in new stack
[2012-04-04 12:11:05] VERBOSE[-1] pbx.c: – Executing [[email protected]:4] Wait(“SIP/101-0000002a”, “1”) in new stack
[2012-04-04 12:11:06] VERBOSE[-1] pbx.c: – Executing [[email protected]:5] Progress(“SIP/101-0000002a”, “”) in new stack
[2012-04-04 12:11:06] VERBOSE[-1] pbx.c: – Executing [[email protected]:6] Playback(“SIP/101-0000002a”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2012-04-04 12:11:06] VERBOSE[-1] file.c: – <SIP/101-0000002a> Playing ‘silence/1.ulaw’ (language ‘en’)
[2012-04-04 12:11:07] VERBOSE[-1] file.c: – <SIP/101-0000002a> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2012-04-04 12:11:10] VERBOSE[-1] file.c: – <SIP/101-0000002a> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[2012-04-04 12:11:10] VERBOSE[-1] pbx.c: == Spawn extension (from-internal, 8122082585, 6) exited non-zero on ‘SIP/101-0000002a’
[2012-04-04 12:11:10] VERBOSE[-1] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/101-0000002a”, “”) in new stack
[2012-04-04 12:11:10] VERBOSE[-1] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/101-0000002a’

Any help is always appreciated!

Did you set up outbound route(s) ?

Yes the outbound route has been setup to the best of my ability. Is there a specific log file that I could post that would give a more detailed explanation of the problem? I would expect at least the inbound number to ring busy instead of the operator stating it’s been disconnected.

Great News! I am now able to receive incoming calls on my IP phone. After looking in my Extensions settigns I saw that it was: “Used as Destination by 2 Objects”. One was my new VOIP phone number and the other was just a forward slash (/). After deleting the forward slash object I was able to make inbound calls. I still cannot make outbound calls but we are making progress :slight_smile: