All circuits busy now

Hello!

First: I’m not an expert with freepbx. I’m currently learning.
My co-worker gave me the credentials for our freepbx before he left the company and now our phones aren’t working anymore. I restarted the freepbx and the phones worked again, for like an hour. Then the same message pops up in the logfiles and all current calls are interrupted. After this, when I try to call a number(for example my mobile phone), my freepbx-phone just tells me “all circuits busy now”.

The logfile tells me this:

[2017-05-15 12:03:07] WARNING[7953][C-0000001e]: file.c:774 ast_openstream_full: File please-try-call-later does not exist in any format
[2017-05-15 12:03:07] WARNING[7953][C-0000001e]: file.c:1100 ast_streamfile: Unable to open please-try-call-later (format (alaw|ulaw)): No such file or directory
[2017-05-15 12:03:07] WARNING[7953][C-0000001e]: app_playback.c:494 playback_exec: Playback failed on PJSIP/4010-0000004b for all-circuits-busy-now&please-try-call-later, noanswer
[2017-05-15 12:03:51] WARNING[7978][C-0000001f]: file.c:774 ast_openstream_full: File please-try-call-later does not exist in any format
[2017-05-15 12:03:51] WARNING[7978][C-0000001f]: file.c:1100 ast_streamfile: Unable to open please-try-call-later (format (alaw|ulaw)): No such file or directory

Are these warnings/errors the reason why we can’t call anyone with out freepbx phones? Our accountant got pretty angry when this error interrupted his call with one of our clients. I would be SO thankful for every hint to solve this!

Daniel

What kind of install is this?
FreePBX distro or something else?

Does this also happen on internal extension to extension calls or only on trunks?

Would be good to provide a full log of a call.

Its FreePBX distro and this happens only to calls to/from outside of our company. Internal calls are not affected.

Which log do you mean? Where can I find it? (I have SSH and web-UI access)
I thought the log above would be enough since this is the exact message when the error occurs. Its what shows up and from that point, no further calls work(outgoing/incoming). When I restart the server, calls work again.

Daniel

If you’re in a jam, paid support is an option:

The log lines you included are not the cause of you call failures. What you are seeing at that point is the system attempting to play recordings to the caller and failing, it may or may not be related to your trunk failure. You are going to have to provide the Asterisk log lines for a failed call:
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs

1 Like

Your system isn’t out of disk space by chance is it??

No, currently 33% used, 67% free disk space.

@lgaetz: Haven’t found anything yet. When I try to “live-log” the server spams things like “executing” and “app_dial”. The problem didn’t occur since yesterday. I’m looking forward to make a call when the “circuits are busy”-message comes up again and I can watch the server writing logs about it.

Will post here as soon as I get info.

EDIT: Something happened again.(Not what this thread was initially about but anyway) All current calls got cancelled and I was able to get this from the log: (I hope I didn’t

[2017-05-17 12:39:15] WARNING[4487] res_pjsip_outbound_registration.c: Temporal response ‘503’ received from ‘sip:tel.t-online.de:5060’ on registration attempt to ‘sip:[email protected]:5060’, retrying in ‘260’
[2017-05-17 12:39:19] WARNING[4487] res_pjsip_outbound_registration.c: Temporal response ‘503’ received from ‘sip:tel.t-online.de:5060’ on registration attempt to ‘sip:[email protected]:5060’, retrying in ‘90’
[2017-05-17 12:39:19] WARNING[4487] res_pjsip_outbound_registration.c: Temporal response ‘503’ received from ‘sip:tel.t-online.de:5060’ on registration attempt to ‘sip:[email protected]:5060’, retrying in ‘265’

[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] bridge_channel.c: Channel PJSIP/AAAAA-0000001e left ‘simple_bridge’ basic-bridge
[2017-05-17 12:39:21] VERBOSE[4656][C-00000010] bridge_channel.c: Channel PJSIP/BBBB-00000023 left ‘simple_bridge’ basic-bridge
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] app_macro.c: Spawn extension (macro-dial, s, 19) exited non-zero on ‘PJSIP/AAAAA-0000001e’ in macro ‘dial’
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] pbx.c: Spawn extension (ext-group, 4444, 15) exited non-zero on ‘PJSIP/AAAAA-0000001e’
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/AAAAA-0000001e”, “hangupcall,”) in new stack
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/AAAAA-0000001e”, “1?theend”) in new stack
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] pbx.c: Goto (macro-hangupcall,s,3)
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/AAAAA-0000001e”, “0?Set(CDR(recordingfile)=)”) in new stack
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] pbx.c: Executing [[email protected]:4] Hangup(“PJSIP/AAAAA-0000001e”, “”) in new stack
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘PJSIP/AAAAA-0000001e’ in macro ‘hangupcall’
[2017-05-17 12:39:21] VERBOSE[4582][C-0000000f] pbx.c: Spawn extension (ext-group, h, 1) exited non-zero on ‘PJSIP/AAAAA-0000001e’
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/CCCC-0000002e”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:1] Set(“PJSIP/CCCC-0000002e”, “TOUCH_MONITOR=1495017570.46”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:2] Set(“PJSIP/CCCC-0000002e”, “AMPUSER=CCCC”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:3] GotoIf(“PJSIP/CCCC-0000002e”, “0?report”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:4] ExecIf(“PJSIP/CCCC-0000002e”, “1?Set(REALCALLERIDNUM=CCCC)”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:5] Set(“PJSIP/CCCC-0000002e”, “AMPUSER=CCCC”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:6] GotoIf(“PJSIP/CCCC-0000002e”, “0?limit”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:7] Set(“PJSIP/CCCC-0000002e”, “AMPUSERCIDNAME=FFFFFF”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:8] GotoIf(“PJSIP/CCCC-0000002e”, “0?report”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:9] Set(“PJSIP/CCCC-0000002e”, “AMPUSERCID=CCCC”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:10] Set(“PJSIP/CCCC-0000002e”, “__DIAL_OPTIONS=Ttr”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:11] Set(“PJSIP/CCCC-0000002e”, "CALLERID(all)=“FFFFFF” ") in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:12] GotoIf(“PJSIP/CCCC-0000002e”, “0?limit”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:13] ExecIf(“PJSIP/CCCC-0000002e”, “1?Set(GROUP(concurrency_limit)=CCCC)”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:14] ExecIf(“PJSIP/CCCC-0000002e”, “1?Set(CHANNEL(language)=de)”) in new stack
[2017-05-17 12:39:30] VERBOSE[5552][C-00000015] pbx.c: Executing [[email protected]:15] GotoIf(“PJSIP/CCCC-0000002e”, “1?continue”) in new stack

I guess the first 3 lines initiated this problem by not responding from our provider, but how could this happen to only 3 of our 8 phones? Only the 3 phones which got disconnected show up, the other 5 were still able to make calls (internal/external). The 3 phones which got disconnected show up as “rejected” unter -> asrerisk -r -> pjsip show registrations, the others are “registered”, how it should be.

Thanks in advance

EDIT 2:
All circuits are busy now, I started logging before doing a call and ended it after my call got rejected. This is what the log says:(I posted it to pastebin for better overview)
https://pastebin.com/1qkjQKCJ

Pjsip could not get the call to your provider…

That’s why the call got all circuits busy. A full sip debug might tell us more but it’s either network connectivity, firewall, internet router, Sip Alg on the router or the provider the sip connection to the provider failed.

Jason