Unable to call Out on PRI circuits

I can make inbound calls to extensions, but I’m unable to make outgoing calls. We usually dial 9 for an outside line on our current system.
I have a new installation of FreePBX 2.11 stable-4.211.64-5 installed using the FreePBX Distro on a new server. This install is using 3 AT&T PRI circuits on a Digium digital quad card so I downloaded the DAHDi module and applied it to the server. I setup the first 3 spans in the DAHDi screen under connectivity.
Span Configs:
Framing/Coding: ESF/B8ZS
Channels 23/24 (T1)
Signalling: PRI-CPE
Swichtype: National ISDN 2 (default)
Synch/Cock Source: 0
Line Build Out 0db
Pridialplan: National
Prilocaldialplan: National
Priexclusive: blank
Receive Gain: 0.0
Transmit Gain: 0.0

Group: 69
Context: from-digital
Used Channels 24

I left the 4th Span Undefined.

I then setup a single trunk using Group 69 with Maximum channels set to 69
Dialed Number manipulation on the trunk
(prepend) +9 | NXXNXXXXXX
(prepend + prefix | match pattern
Dial Rules Wizards: none
Outbound Dial Prefix: blank

DAHDI Trunks: Group 69 Ascending

I then set an outbound route
Route Name: 9_outside
Route CID: our main AT&T number
Route Password: blank
Route Type: nothing checked
Music on Hold: Default
Time Group: Permanent Route
Route Position: No Change

Dial Patterns that will use this route: left with defaults (prepend)+prefix | match pattern /CallerID
(This is the way it currently setup in our current phone server so I figured it was using the trunk defaults)
Normal Congestion set under Optional Destination for Matched Routes

I then setup some extensions to test with and made some inbound calls with no problems, but outbound calls will not work.
I have to test any changes after hours by moving the PRI circuits to the new system and testing and then moving them back prior to business hours so it take some time to troubleshoot.
I know this is probably a configuration issue somewhere so I was wondering if anyone could point me to what I did wrong or didn’t do right?

can you show your pri debug, do the debug when your calling

Sorry it took a couple of days to schedule moving the circuits. I tried to pull the PRI log this morning but when I’m calling out it’s not activating the PRI circuit so nothing is getting written out in the log. When I call into the building which is working the PRI log is populated. I tried this several times with different file names.

I think I have something misconfigured further up stream. I’m getting a message when I call out saying “the call cannot be completed as dialed” so perhaps something is wrong with my trunk or outbound route, but I set them up the way they are setup in our current much older version of Freepbx? I have tried calling out using 9 which is the way it works now in our old system and then also just dialing the number and neither one works.
I was also looking at the other logs and it appears I get the following message when I’m trying to dial out. Not sure if this might mean something.
[2013-09-06 05:59:45] WARNING[7477][C-00000016]: channel.c:4816 ast_prod: Prodding channel ‘SIP/1502-00000016’ failed

Any feedback you could give me is appreciated.

You should see something farther up in the log. Before it indicates playing the invalid message.

^this. If the problem is not obvious please pastebin the log for us.

OK so I’m new to Linux, but I believe the information you are looking for is in the file full, for some reason my log from the gui isn’t working this morning but that’s another issue I can worry about later. I am pasting the output from two tries. the first with 9 in front of the number which I realized from the output the 9 wasn’t getting dropped so I did it a second time and just dialed my extension. I did a replace with 1234567890 to replace the actual number dialed. I do not currently have the PRI circuits connected since it is during business hours, but based on my previous research I think this is blowing up before it get’s to that point.

[2013-09-06 09:14:29] VERBOSE[9113][C-00000005] netsock2.c: == Using SIP RTP TOS bits 184
[2013-09-06 09:14:29] VERBOSE[9113][C-00000005] netsock2.c: == Using SIP RTP CoS mark 5
[2013-09-06 09:14:29] VERBOSE[9324][C-00000005] pbx.c: – Executing [[email protected]:1] ResetCDR(“SIP/1502-00000005”, “”) in new stack
[2013-09-06 09:14:29] VERBOSE[9324][C-00000005] pbx.c: – Executing [[email protected]:2] NoCDR(“SIP/1502-00000005”, “”) in new stack
[2013-09-06 09:14:29] VERBOSE[9324][C-00000005] pbx.c: – Executing [[email protected]:3] Progress(“SIP/1502-00000005”, “”) in new stack
[2013-09-06 09:14:29] VERBOSE[9324][C-00000005] pbx.c: – Executing [[email protected]:4] Wait(“SIP/1502-00000005”, “1”) in new stack
[2013-09-06 09:14:30] VERBOSE[9324][C-00000005] pbx.c: – Executing [[email protected]:5] Progress(“SIP/1502-00000005”, “”) in new stack
[2013-09-06 09:14:30] VERBOSE[9324][C-00000005] pbx.c: – Executing [[email protected]:6] Playback(“SIP/1502-00000005”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2013-09-06 09:14:30] VERBOSE[9324][C-00000005] file.c: – <SIP/1502-00000005> Playing ‘silence/1.ulaw’ (language ‘en’)
[2013-09-06 09:14:31] VERBOSE[9324][C-00000005] file.c: – <SIP/1502-00000005> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2013-09-06 09:14:34] VERBOSE[9324][C-00000005] file.c: – <SIP/1502-00000005> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[2013-09-06 09:14:36] VERBOSE[9324][C-00000005] pbx.c: – Executing [[email protected]:7] Wait(“SIP/1502-00000005”, “1”) in new stack
[2013-09-06 09:14:36] VERBOSE[9324][C-00000005] pbx.c: == Spawn extension (from-internal, 91234567890, 7) exited non-zero on ‘SIP/1502-00000005’
[2013-09-06 09:14:36] VERBOSE[9324][C-00000005] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/1502-00000005”, “”) in new stack
[2013-09-06 09:14:36] VERBOSE[9324][C-00000005] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1502-00000005’

[2013-09-06 09:17:16] VERBOSE[9113][C-00000006] netsock2.c: == Using SIP RTP TOS bits 184
[2013-09-06 09:17:16] VERBOSE[9113][C-00000006] netsock2.c: == Using SIP RTP CoS mark 5
[2013-09-06 09:17:16] VERBOSE[9349][C-00000006] pbx.c: – Executing [[email protected]:1] ResetCDR(“SIP/1502-00000006”, “”) in new stack
[2013-09-06 09:17:16] VERBOSE[9349][C-00000006] pbx.c: – Executing [[email protected]:2] NoCDR(“SIP/1502-00000006”, “”) in new stack
[2013-09-06 09:17:16] VERBOSE[9349][C-00000006] pbx.c: – Executing [[email protected]:3] Progress(“SIP/1502-00000006”, “”) in new stack
[2013-09-06 09:17:16] VERBOSE[9349][C-00000006] pbx.c: – Executing [[email protected]:4] Wait(“SIP/1502-00000006”, “1”) in new stack
[2013-09-06 09:17:17] VERBOSE[9349][C-00000006] pbx.c: – Executing [[email protected]:5] Progress(“SIP/1502-00000006”, “”) in new stack
[2013-09-06 09:17:17] VERBOSE[9349][C-00000006] pbx.c: – Executing [[email protected]:6] Playback(“SIP/1502-00000006”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2013-09-06 09:17:17] VERBOSE[9349][C-00000006] file.c: – <SIP/1502-00000006> Playing ‘silence/1.ulaw’ (language ‘en’)
[2013-09-06 09:17:18] VERBOSE[9349][C-00000006] file.c: – <SIP/1502-00000006> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2013-09-06 09:17:21] VERBOSE[9349][C-00000006] file.c: – <SIP/1502-00000006> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[2013-09-06 09:17:23] VERBOSE[9349][C-00000006] pbx.c: – Executing [[email protected]:7] Wait(“SIP/1502-00000006”, “1”) in new stack
[2013-09-06 09:17:24] VERBOSE[9349][C-00000006] pbx.c: == Spawn extension (from-internal, 1234567890, 7) exited non-zero on ‘SIP/1502-00000006’
[2013-09-06 09:17:24] VERBOSE[9349][C-00000006] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/1502-00000006”, “”) in new stack
[2013-09-06 09:17:24] VERBOSE[9349][C-00000006] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1502-00000006’

I don’t see an outbound route matched.

You can also watch on Cisco console with ‘debug ccsip calls’

If remoted in don’t forget ‘term mon’ and when done ‘undebug all’

Thanks for the heads up on that. For some reason the outbound route was not matching like you said. I’m not sure why, but I chose the dial patterns wizards local 7/10 digit call which added NXXXXXX AND NXXNXXXXXX and I put a 9 in the prefix section so I thought that my number which was equal to 4041234567 would work with the +9|NXXNXXXXXX dial pattern but for some reason it was not. I went ahead and did a +9|. which allows it to work but it currently let’s everything through if they dial 9. Any ideas on why it wasn’t working the other way? Not a super big thing right now because I can move ahead with getting everything else setup, but if we ever wanted to limit what numbers could be called it would be helpful to understand what I was doing wrong.

Thanks,