All circuits are busy


(United States) #1

Hello I’m having issues with outbound calls.
I get a message stating “All circuits are busy”
this is my first FreePBX install or ANY Phone server for that matter.
Any help would be greatly appreciated.

thank you in advance
CLI and SIP settings pasted below

type=peer
host=sip.digiumcloud.net
defaultuser=username
fromuser=username
secret=password
insecure=invite
trustrpid=yes
sendrpid=pai
context=from-digium-siptrunk
directmedia=no
disallow=all
allow=g722&ulaw&g729

Connected to Asterisk 16.6.2 currently running on freepbx (pid = 40539)

[Dec 31 11:14:27] == Using SIP RTP TOS bits 184

[Dec 31 11:14:27] == Using SIP RTP CoS mark 5

[Dec 31 11:14:27] > 0x7f5eec030320 – Strict RTP learning after remote address set to: 192.168.4.89:63912

[Dec 31 11:14:27] – Executing [916199547723@from-internal:1] ResetCDR("SIP/112-00000041", "") in new stack

[Dec 31 11:14:27] – Executing [916199547723@from-internal:2] NoCDR("SIP/112-00000041", "") in new stack

[Dec 31 11:14:27] – Executing [916199547723@from-internal:3] Progress("SIP/112-00000041", "") in new stack

[Dec 31 11:14:27] – Executing [916199547723@from-internal:4] Wait("SIP/112-00000041", "1") in new stack

[Dec 31 11:14:27] > 0x7f5eec030320 – Strict RTP switching to RTP target address 192.168.4.89:63912 as source

[Dec 31 11:14:28] – Executing [916199547723@from-internal:5] Playback("SIP/112-00000041", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack

[Dec 31 11:14:28] – <SIP/112-00000041> Playing ‘silence/1.ulaw’ (language ‘en’)

[Dec 31 11:14:29] – <SIP/112-00000041> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)

[Dec 31 11:14:31] – <SIP/112-00000041> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)

[Dec 31 11:14:32] > 0x7f5eec030320 – Strict RTP learning complete - Locking on source address 192.168.4.89:63912

[Dec 31 11:14:34] – Executing [916199547723@from-internal:6] Wait("SIP/112-00000041", "1") in new stack

[Dec 31 11:14:35] – Executing [916199547723@from-internal:7] Congestion("SIP/112-00000041", "20") in new stack

[2019-12-31 11:14:35] WARNING[56219][C-00000029]: channel.c:4864 ast_prod: Prodding channel ‘SIP/112-00000041’ failed

[Dec 31 11:14:35] == Spawn extension (from-internal, 916199547723, 7) exited non-zero on ‘SIP/112-00000041’

[Dec 31 11:14:35] – Executing [h@from-internal:1] Macro("SIP/112-00000041", "hangupcall") in new stack

[Dec 31 11:14:35] – Executing [s@macro-hangupcall:1] GotoIf("SIP/112-00000041", "1?theend") in new stack

[Dec 31 11:14:35] – Goto (macro-hangupcall,s,3)

[Dec 31 11:14:35] – Executing [s@macro-hangupcall:3] ExecIf("SIP/112-00000041", "0?Set(CDR(recordingfile)=)") in new stack

[Dec 31 11:14:35] – Executing [s@macro-hangupcall:4] NoOp("SIP/112-00000041", " montior file= ") in new stack

[Dec 31 11:14:35] – Executing [s@macro-hangupcall:5] GotoIf("SIP/112-00000041", "1?skipagi") in new stack

[Dec 31 11:14:35] – Goto (macro-hangupcall,s,7)

[Dec 31 11:14:35] – Executing [s@macro-hangupcall:7] Hangup("SIP/112-00000041", "") in new stack

[Dec 31 11:14:35] == Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/112-00000041’ in macro ‘hangupcall’

[Dec 31 11:14:35] == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/112-00000041’

freepbx*CLI>


(Lorne Gaetz) #2

You don’t have an outbound route set up with a pattern that matches 916199547723


(United States) #3

I’ve tried both with and without the 9 or 1 before with the same issue.
Dial plans below.
NXXXXXX
NXXNXXXXXX
1NXXNXXXXXX

Thank you for the quick reply btw.


(Itzik) #4

Setup the following dial patterns in your outbound route.

You can see it properly reflects in the database.

MariaDB [asterisk]> select * FROM outbound_route_patterns;
+----------+----------------------+--------------------+-----------+----------------+
| route_id | match_pattern_prefix | match_pattern_pass | match_cid | prepend_digits |
+----------+----------------------+--------------------+-----------+----------------+
|        1 |                      | 1NXXNXXXXXX        |           |                |
|        1 |                      | 911                |           |                |
|        1 |                      | 933                |           |                |
|        1 |                      | NXXNXXXXXX         |           | 1              |
|        1 |                      | NXXXXXX            |           | 1619           |
|        1 | 1                    | 911                |           |                |
|        1 | 9                    | 1NXXNXXXXXX        |           |                |
|        1 | 9                    | 911                |           |                |
|        1 | 9                    | NXXNXXXXXX         |           | 1              |
|        1 | 9                    | NXXXXXX            |           | 1619           |
|        1 | 91                   | 911                |           |                |
+----------+----------------------+--------------------+-----------+----------------+

Place another call and if it doesn’t work, please provide a log as outlined in this link:

https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII


(United States) #5

That didn’t work.
here are the log files.

Thanks again,

login as: root
root@192.168.4.254’s password:
Last login: Tue Dec 31 13:07:00 2019 from 192.168.4.89


| | | ___ | ___ \ \ / /
| |
_ __ ___ ___ | |
/ /| |
/ / \ V /
| | | '__| / _ \ / _ | __/ | ___ \ /
| | | | | __/| __/| | | |
/ // /^\
_| || _| _|_| ___/ / /

NOTICE! You have 5 notifications! Please log into the UI to see them!
Current Network Configuration
±----------±------------------±--------------------------+
| Interface | MAC Address | IP Addresses |
±----------±------------------±--------------------------+
| eth0 | 38:63:BB:46:8C:10 | 184.188.54.248 |
| eth1 | 38:63:BB:46:8C:11 | 192.168.4.254 |
| | | fe80::3a63:bbff:fe46:8c11 |
| eth2 | 38:63:BB:46:8C:12 | |
| eth3 | 38:63:BB:46:8C:13 | |
±----------±------------------±--------------------------+

Please note most tasks should be handled through the GUI.
You can access the GUI by typing one of the above IPs in to your web browser.
For support please visit:
http://www.freepbx.org/support-and-professional-services

±-----------------------------------------------------------+
| Your system is currently up to date! |
| Your PBX is up to date. |
| Also 2 Uninstalled modules. |
±-----------------------------------------------------------+
[root@freepbx ~]# tail -f /var/log/asterisk/full
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] pbx.c: Executing [s@crm-hangup: 2] NoOp(“SIP/112-0000005d”, “HANGUP CAUSE: 34”) in new stack
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] pbx.c: Executing [s@crm-hangup: 3] ExecIf(“SIP/112-0000005d”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] pbx.c: Executing [s@crm-hangup: 4] NoOp(“SIP/112-0000005d”, “MASTER CHANNEL: 1577828508.108 = 1577828508.108”) i n new stack
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] pbx.c: Executing [s@crm-hangup: 5] GotoIf(“SIP/112-0000005d”, “0?return”) in new stack
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] pbx.c: Executing [s@crm-hangup: 6] Set(“SIP/112-0000005d”, “__CRM_HANGUP=1”) in new stack
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] pbx.c: Executing [s@crm-hangup: 7] AGI(“SIP/112-0000005d”, “agi://127.0.0.1/sangomacrm.agi”) in new stack
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] res_agi.c: <SIP/112-0000005d>AG I Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] pbx.c: Executing [s@crm-hangup: 8] Return(“SIP/112-0000005d”, “”) in new stack
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] app_stack.c: Spawn extension (f rom-internal, h, 1) exited non-zero on ‘SIP/112-0000005d’
[2019-12-31 13:41:52] VERBOSE[17753][C-0000003a] app_stack.c: SIP/112-0000005d I nternal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2019-12-31 13:43:25] VERBOSE[40591][C-0000003b] netsock2.c: Using SIP RTP TOS bits 184
[2019-12-31 13:43:25] VERBOSE[40591][C-0000003b] netsock2.c: Using SIP RTP CoS mark 5
[2019-12-31 13:43:25] VERBOSE[18292][C-0000003b] pbx.c: Executing [916199547723@from-internal:1] ResetCDR(“SIP/112-0000005f”, “”) in new stack
[2019-12-31 13:43:25] VERBOSE[18292][C-0000003b] pbx.c: Executing [916199547723@from-internal:2] NoCDR(“SIP/112-0000005f”, “”) in new stack
[2019-12-31 13:43:25] VERBOSE[18292][C-0000003b] pbx.c: Executing [916199547723@from-internal:3] Progress(“SIP/112-0000005f”, “”) in new stack
[2019-12-31 13:43:25] VERBOSE[18292][C-0000003b] pbx.c: Executing [916199547723@from-internal:4] Wait(“SIP/112-0000005f”, “1”) in new stack
[2019-12-31 13:43:26] VERBOSE[18292][C-0000003b] pbx.c: Executing [916199547723@from-internal:5] Playback(“SIP/112-0000005f”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2019-12-31 13:43:26] VERBOSE[18292][C-0000003b] file.c: <SIP/112-0000005f> Playing ‘silence/1.ulaw’ (language ‘en’)
[2019-12-31 13:43:27] VERBOSE[18292][C-0000003b] file.c: <SIP/112-0000005f> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2019-12-31 13:43:29] VERBOSE[18292][C-0000003b] file.c: <SIP/112-0000005f> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[2019-12-31 13:43:32] VERBOSE[18292][C-0000003b] pbx.c: Executing [916199547723@from-internal:6] Wait(“SIP/112-0000005f”, “1”) in new stack
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Executing [916199547723@from-internal:7] Congestion(“SIP/112-0000005f”, “20”) in new stack
[2019-12-31 13:43:33] WARNING[18292][C-0000003b] channel.c: Prodding channel ‘SIP/112-0000005f’ failed
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Spawn extension (from-internal, 916199547723, 7) exited non-zero on ‘SIP/112-0000005f’
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Executing [h@from-internal:1] Macro(“SIP/112-0000005f”, “hangupcall”) in new stack
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/112-0000005f”, “1?theend”) in new stack
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/112-0000005f”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/112-0000005f”, " montior file= ") in new stack
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“SIP/112-0000005f”, “1?skipagi”) in new stack
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“SIP/112-0000005f”, “”) in new stack
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/112-0000005f’ in macro ‘hangupcall’
[2019-12-31 13:43:33] VERBOSE[18292][C-0000003b] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/112-0000005f’


(Itzik) #6

If you choose posting the raw log over pastebin at least use the formatter tool to help us read it.

Did you apply config after making changes?

Can you please post screenshots of your outbound route?


(United States) #7

I apologize for the paste, as I said before I’m new to this.
Yes I did select apply config after making the changes.

here are the dial patterns in my outbound route.


(United States) #8


(United States) #9

Thank you everyone for your help.
turns out the problem was me.
there was a space in my password.
Outbound is working just fine now.

Happy New Year!!

Maurice


(system) closed #10

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