Broke my out going calls, All Circuits Busy

I was adding and removing users and extensions and now lost outbound calling. This has not been my day.

https://pastebin.freepbx.org/view/8128d018#L208

Well you should check PJSIP/[email protected]
You should check what’s going on with F1System (outbound route settings).

[2020-05-28 13:42:00] VERBOSE[12135][C-0005f01b] app_dial.c: Called PJSIP/[email protected]
[2020-05-28 13:42:01] VERBOSE[12135][C-0005f01b] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[

Agreed, that much is clear. I take it you did not see anything else weird in there. Didnt change routes or dial patterns. SIP provider has not turned off the tap, checked that first.

Maybe delete the route and recreate it.

[2020-05-28 13:42:00] VERBOSE[12135][C-0005f01b] pbx.c: Executing [[email protected]:23] Set("PJSIP/204-0005c8cb", "TRUNKOUTCID=303-464-0011") in new stack
This got used as the caller ID to send, because there was apparently no CID set for the extension or the outbound route.

Your provider will probably reject a number containing dashes. Also, they might require 13034640011.

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Well no by the way it’s quite hard to check, because FreePBX log invoke many subroutines to each call.
I’ve see in other posts that your system has been compromised, have you any protection? As SBC or fail2band.

f2b is running and has not objected to anything. I will look at the extension CID, normally they do not have anything set so just use the default value when created.

There is no default. If no Outbound CID is set for the extension, the Route CID is used. If that’s also not present, the Trunk CID. If that is malformed, the provider may reject the call.

If that’s not your issue, at the Asterisk command prompt, type
pjsip set logger on
and make a failing test call.
Post a new log, which will not include the SIP trace.

Fail2ban filtering setup is difficult and takes time to work, you need to check with Asterisk log and detect compromised messages and test with rules.
At this instance better to close in router all incoming port in order to stop who is compromised your system.

https://pastebin.freepbx.org/view/1a255104

There is no logger details in that paste. Please update it with that information.

Your first pastebin shows no CALLERID(all) matches. That’s a problem.

[2020-05-28 13:42:00] VERBOSE[12135][C-0005f01b] pbx.c: Executing [[email protected]:31] ExecIf("PJSIP/204-0005c8cb", "0?Set(CALLERID(all)=)") in new stack
[2020-05-28 13:42:00] VERBOSE[12135][C-0005f01b] pbx.c: Executing [[email protected]:32] ExecIf("PJSIP/204-0005c8cb", "0?Set(CALLERID(all)=)") in new stack
[2020-05-28 13:42:00] VERBOSE[12135][C-0005f01b] pbx.c: Executing [[email protected]:33] ExecIf("PJSIP/204-0005c8cb", "0?Set(CALLERID(all)=204)") in new stack

Your second paste shows the CALLERID(all) getting set with the extension number being the final value.

[2020-05-28 16:27:04] VERBOSE[7951][C-0005f040] pbx.c: Executing [[email protected]:30] ExecIf("PJSIP/204-0005c904", "1?Set(CALLERID(all)=303-464-0011)") in new stack
[2020-05-28 16:27:04] VERBOSE[7951][C-0005f040] pbx.c: Executing [[email protected]:31] ExecIf("PJSIP/204-0005c904", "1?Set(CALLERID(all)=204)") in new stack

This is a failed call with pjsip logging on

https://pastebin.freepbx.org/view/d74d10be

You must be doing something wrong. Because there should be actual SIP messages displayed here. Try this again by starting from the system CLI and then do:

asterisk -r
pjsip set logger on

Do not have any verbose output. We don’t see to see that at this point. We need to see actual SIP messages.

you called 1303931911 but F1Systems repied with a cause 21

Cause No. 21 - call rejected.
This cause indicates that the equipment sending this cause does not wish to accept this call. although it could have accepted the call because the equipment sending this cause is neither busy nor incompatible. This cause may also be generated by the network, indicating that the call was cleared due to a supplementary service constraint. The diagnostic field may contain additional information about the supplementary service and reason for rejection.strong text

Only resource, call F1Systems

Im not understanding. F1systems is the PBX name. Calling any number including the main phone number from any of the extensions gets All Circuits Busy.

Where is the PJSIP logger output?

When you type
pjsip set logger on
you should see a response
PJSIP Logging enabled
If you see nothing or something different, provide details.
Then, without doing anything in the FreePBX GUI or on the console, make your failing test call. The Asterisk log should now include the SIP trace.

Does not the first line show you calling ‘Yourself’ then ?

  1. [2020-05-28 17:43:30] VERBOSE[19731][C-0005f044] app_dial.c: Called PJSIP/[email protected]

  2. [2020-05-28 17:43:30] VERBOSE[19731][C-0005f044] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)

  3. [2020-05-28 17:43:30] VERBOSE[19731][C-0005f044] pbx.c: Executing [[email protected]:35] NoOp(“PJSIP/204-0005c914”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21”) in new stack

I was gathering this was a client PBX that stopped being able to call the F1Systems PBX after changes where made.

Either way , [email protected] is replying with a 21

Outgoing calls should not be using the local service.

https://pastebin.freepbx.org/view/d74d10be