Can't make outbound calls: 1/2 second ring tone then: "Your call cannot be completed..."

Hello all. I’m a USA user with: FreePBX 2.5.2.2 with Asterisk API 2.5.0.3 and the X-Lite softphone. I have a Callcentric account that enables me to make outbound and inbound calls using an ATA only but no oubound from the PBX. I can receive calls on the PBX and have setup an IVR, voice mail, etc and they work great. I’ve tried many different suggestions I’ve found here at www.freepbx.org and combinations of them. I’ve also read that I may need to open up some ports or look at “Port Forwarding” or Triggering" at the firewall.

Below is what happens when I try to make an outbound call. Also below are my settings with private characters substituted:

I dailed using a 1: 13335550000, 3335550000 and also using a 9: 913335550000. I dialed from my extension 31000.

When I start with a 9 and 1 (913335550000) it asks me for my “password followed by the pound key”. If I use my numerical Voice Mail password for the ext I dialed from, it says the callcentric number is not in service and so on or “Password incorrect…”, and loops as I make failed tries. If I don’t do anything then after some loops it says “Goodbeye” and hangs up. The only reason I tried my Voice Mail password here is because it is numeric.

I get this when I don’t use a 9 (13335550000):
Ring tone for 1/2 second then: Your call cannot be completed as dialed. Please check the number and dial again. (3 tones for incorrect # plays and all beyond this and loops) The person you are calling is unavailable. Please try again.
–this seems to give the best results so I captured a portion of the log detailing my last failed outbound call attempt below—

I get this when I don’t use a 9 or 1 (3335550000):
(3 tones for incorrect # plays and all beyond this and loops) The person you are calling is unavailable. Please try again.

Edit Route

Route Name: to-callcentric-outbound
Route Password: ###
PIN Set: None
Emergency Dialing: [] --none
Intra Company Route: [] --none
Music On Hold?: default
Dial Patterns: 9|.
Dial patterns wizards: (pick one)
Trunk Sequence: 0 SIP/ClCtrc2

Dialing Options

Asterisk Dial command options: tr
Asterisk Outbound Dial command options:

Edit SIP Trunk > In use by 1 route

Outbound Caller ID: 13335550000
Never Override CallerID: [] --none
Maximum Channels: 3
Disable Trunk: [] --none
Monitor Trunk Failures: [] --none [] --none Enable
Outgoing Dial Rules > Dial Rules: none
Dial Rules Wizards: (pick one)
Outbound Dial Prefix: 9
Outgoing Settings > Trunk Name: ClCtrc2
PEER Details:
disallow=all
allow=gsm&ulaw&g729&g723
host=callcentric.com
qualify=yes
username=1777#######
secret=###
type=peer
context=from-trunk
fromuser=1777#######
fromdomain=callcentric.com
nat=yes
insecure=port,invite
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas
dtmfmode=rfc2833

Incoming Settings > USER Context: 31000usr
USER Details:
type=user
secret=###
host=dynamic
qualify=no
session-timers=refuse
nat=yes
canreinvite=no
dtmfmode=rfc2833

Registration > Register String: 1777#######:###@callcentric.com/1777#######

Log file of outbound attempt of: 13335550000. I hung up before it looped after it didn’t work.

[Apr 17 11:21:33] VERBOSE[4034] logger.c: == Manager ‘admin’ logged off from 127.0.0.1
[Apr 17 11:24:18] VERBOSE[4041] logger.c: – Executing [[email protected]:1] ResetCDR(“SIP/31000-09c2bcb0”, “”) in new stack
[Apr 17 11:24:18] VERBOSE[4041] logger.c: – Executing [[email protected]nternal:2] NoCDR(“SIP/31000-09c2bcb0”, “”) in new stack
[Apr 17 11:24:18] VERBOSE[4041] logger.c: – Executing [[email protected]:3] Wait(“SIP/31000-09c2bcb0”, “1”) in new stack
[Apr 17 11:24:20] VERBOSE[4041] logger.c: – Executing [[email protected]:4] Playback(“SIP/31000-09c2bcb0”, “silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer”) in new stack
[Apr 17 11:24:20] VERBOSE[4041] logger.c: – <SIP/31000-09c2bcb0> Playing ‘silence/1’ (language ‘en’)
[Apr 17 11:24:21] VERBOSE[4041] logger.c: – <SIP/31000-09c2bcb0> Playing ‘cannot-complete-as-dialed’ (language ‘en’)
[Apr 17 11:24:24] VERBOSE[4041] logger.c: – <SIP/31000-09c2bcb0> Playing ‘check-number-dial-again’ (language ‘en’)
[Apr 17 11:24:26] VERBOSE[4041] logger.c: – Executing [[email protected]:5] Wait(“SIP/31000-09c2bcb0”, “1”) in new stack
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Executing [[email protected]:6] Congestion(“SIP/31000-09c2bcb0”, “20”) in new stack
[Apr 17 11:24:27] VERBOSE[4041] logger.c: == Spawn extension (from-internal, 13335550000, 6) exited non-zero on ‘SIP/31000-09c2bcb0’
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Executing [[email protected]:1] Macro(“SIP/31000-09c2bcb0”, “hangupcall”) in new stack
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Executing [[email protected]:1] GotoIf(“SIP/31000-09c2bcb0”, “1?skiprg”) in new stack
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Goto (macro-hangupcall,s,4)
[Apr 17 11:24:27] DEBUG[4041] app_macro.c: Executed application: GotoIf
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Executing [[email protected]:4] GotoIf(“SIP/31000-09c2bcb0”, “1?skipblkvm”) in new stack
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Goto (macro-hangupcall,s,7)
[Apr 17 11:24:27] DEBUG[4041] app_macro.c: Executed application: GotoIf
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Executing [[email protected]:7] GotoIf(“SIP/31000-09c2bcb0”, “1?theend”) in new stack
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Goto (macro-hangupcall,s,9)
[Apr 17 11:24:27] DEBUG[4041] app_macro.c: Executed application: GotoIf
[Apr 17 11:24:27] VERBOSE[4041] logger.c: – Executing [[email protected]:9] Hangup(“SIP/31000-09c2bcb0”, “”) in new stack
[Apr 17 11:24:27] VERBOSE[4041] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/31000-09c2bcb0’ in macro ‘hangupcall’
[Apr 17 11:24:27] VERBOSE[4041] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/31000-09c2bcb0’
[Apr 17 11:24:44] VERBOSE[4042] logger.c: == Parsing ‘/etc/asterisk/manager.conf’: [Apr 17 11:24:44] VERBOSE[4042] logger.c: Found
[Apr 17 11:24:44] VERBOSE[4042] logger.c: == Parsing ‘/etc/asterisk/manager_additional.conf’: [Apr 17 11:24:44] VERBOSE[4042] logger.c: Found
[Apr 17 11:24:44] VERBOSE[4042] logger.c: == Parsing ‘/etc/asterisk/manager_custom.conf’: [Apr 17 11:24:44] VERBOSE[4042] logger.c: Found
[Apr 17 11:24:44] WARNING[4042] config.c: Unknown directive ‘#permit=192.168.15.0/255.255.255.0’ at line 18 of /etc/asterisk/manager_custom.conf
[Apr 17 11:24:44] VERBOSE[4042] logger.c: == Manager ‘admin’ logged on from 127.0.0.1

Help appreciated. Let me know if I left something out that would help with troubleshooting.

In your trunk remove

Outbound Dial Prefix: 9

That is used if you want to add a prefix to your outbound calling number. For example, if your telco requires 10 digit dialing, then you could dial 7 digits and have the area code (and/or country code) automatically added.

Edit SIP Trunk

Outbound Dial Prefix: 9 <-- Removed this but it made no difference when dialing 13335550000, however…

I did further exploring on what you said about the “9” and also removed it in the route pattern and tried the actual target # in its place: 13335550000 and it worked! It required me to key in the numerical password and then it let me through. I’ve been stuck on this for several nights over the past few months and now that’s behind me. Thanks!

Edit Route

Route Name: to-callcentric-outbound
Route Password: ### <-- I used this numerical password
PIN Set: None
Emergency Dialing: [] --none
Intra Company Route: [] --none
Music On Hold?: default
Dial Patterns: 9|. <-- I removed “9|.” and added the target ph #: 13335550000
Dial patterns wizards: (pick one)
Trunk Sequence: 0 SIP/ClCtrc2

I came here to get help with the dial pattern for my own FreePBX install and saw the input from you guys and remembered how to setup dialplans in the past so that it wouldn’t be necessary to make a dialplan for every number that you’ll want to call. Instead of “13335550000” you could make a dial pattern of “1333XXXXXXX”. I just tested out a similar dialplan on my own install and the call went through flawless. I will go on to say that numbers provided before the “X’s” should match the type of numbers you’ll have to dial for your provider. I don’t have my PBX pointing towards a live server as of yet so I haven’t had a chance to play around with the dialplans to see what is necessary for codes requiring area codes. Hopefully, the more I play around with this in my test setup, the more that comes back to my memory.

This has nothing to do with dial patterns. You also don’t have to remember it. Hover your mouse over the little circle and all the variables are explained.