Broke my out going calls, All Circuits Busy

Your paste contains no SIP trace. Are you certain that you saw the
PJSIP Logging enabled
confirmation and then made the test call without reloading or restarting FreePBX?

Thanks for taking the time on this.

asterisk -r
pjsip set logging on

Here is the pastebin and the screen output.

https://pastebin.freepbx.org/view/19685420


<— Received SIP request (545 bytes) from UDP:34.210.91.112:5060 —>
OPTIONS sip:[email protected]:5060;line=gqhogmf SIP/2.0
Max-Forwards: 20
Record-Route: sip:34.210.91.112;lr
Via: SIP/2.0/UDP 34.210.91.112:5060;branch=z9hG4bKd1ea.4d5ce7f8d460982970fcd9c153ddac9c.0
Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK9697744
Route: sip:34.210.91.112:5060;lr;received=sip:173.160.51.229:5060
From: sip:ping@invalid;tag=uloc-5ec4be7d-18-c68d3a1-ffeb45d6-f6d3697d
To: sip:[email protected]:5060;line=gqhogmf
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0
Max-Forward: 10

<— Transmitting SIP response (1006 bytes) to UDP:34.210.91.112:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 34.210.91.112:5060;rport=5060;received=34.210.91.112;branch=z9hG4bKd1ea.4d5ce7f8d460982970fcd9c153ddac9
c.0
Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK9697744
Record-Route: sip:34.210.91.112;lr
Call-ID: [email protected]
From: sip:ping@invalid;tag=uloc-5ec4be7d-18-c68d3a1-ffeb45d6-f6d3697d
To: sip:[email protected];tag=z9hG4bKd1ea.4d5ce7f8d460982970fcd9c153ddac9c.0;line=gqhogmf
CSeq: 1 OPTIONS
Accept: application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summar
y, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, applicat
ion/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: FPBX-14.0.13.33(16.6.2)
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:192.168.1.247:5060 —>
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.244:5060;rport;branch=z9hG4bKPj4ab84db9-26e3-4e42-9461-2fd71f2e51a5
From: sip:[email protected];tag=5bf045d8-aa88-4f40-9e87-66902e1d63f8
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: c8e8c67c-e1bf-4808-92e1-3e9944f91d68
CSeq: 12463 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.13.33(16.6.2)
Content-Length: 0

<— Received SIP response (358 bytes) from UDP:192.168.1.247:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.244:5060;rport;branch=z9hG4bKPj4ab84db9-26e3-4e42-9461-2fd71f2e51a5
From: sip:[email protected];tag=5bf045d8-aa88-4f40-9e87-66902e1d63f8
To: sip:[email protected];tag=1270616766
Call-ID: c8e8c67c-e1bf-4808-92e1-3e9944f91d68
CSeq: 12463 OPTIONS
User-Agent: Yealink SIP-W52P 25.73.0.40
Content-Length: 0

<— Transmitting SIP request (479 bytes) to UDP:34.210.91.112:5060 —>
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 173.160.51.229:5060;rport;branch=z9hG4bKPje995caca-6d17-497d-ac45-0b26b382fb5e
From: sip:[email protected];tag=1597fb66-7a72-4082-bc28-20b652533918
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: 53217800-2cdf-4c53-8ea1-61ac34c498de
CSeq: 60141 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.13.33(16.6.2)
Content-Length: 0

<— Received SIP response (411 bytes) from UDP:34.210.91.112:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 173.160.51.229:5060;rport=5060;branch=z9hG4bKPje995caca-6d17-497d-ac45-0b26b382fb5e;received=173.160.51
.229
From: sip:[email protected];tag=1597fb66-7a72-4082-bc28-20b652533918
To: sip:[email protected];tag=bf8638324618dc61059d4c604476fea1.03becf33
Call-ID: 53217800-2cdf-4c53-8ea1-61ac34c498de
CSeq: 60141 OPTIONS
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:192.168.1.203:5060 —>
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.244:5060;rport;branch=z9hG4bKPjd608b2ef-49e9-4313-8aa2-c61dfb4b4269
From: sip:[email protected];tag=40151185-2011-45f3-9b7f-df876540f08d
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: 083d0797-f19a-4949-902b-752b63eade62
CSeq: 28631 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.13.33(16.6.2)
Content-Length: 0

<— Received SIP response (357 bytes) from UDP:192.168.1.203:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.244:5060;rport;branch=z9hG4bKPjd608b2ef-49e9-4313-8aa2-c61dfb4b4269
From: sip:[email protected];tag=40151185-2011-45f3-9b7f-df876540f08d
To: sip:[email protected];tag=3737165474
Call-ID: 083d0797-f19a-4949-902b-752b63eade62
CSeq: 28631 OPTIONS
User-Agent: Yealink SIP-T26P 6.73.0.50
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:192.168.1.195:5062 —>
OPTIONS sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.244:5060;rport;branch=z9hG4bKPj61efcbaf-2996-4cbe-8d33-716385b14c5e
From: sip:[email protected];tag=7a60bebc-3202-4a55-998e-13d932066232
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: 14e790c0-ab0f-4eb4-8a98-83997befca9a
CSeq: 44168 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.13.33(16.6.2)
Content-Length: 0

<— Received SIP response (356 bytes) from UDP:192.168.1.195:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.244:5060;rport;branch=z9hG4bKPj61efcbaf-2996-4cbe-8d33-716385b14c5e
From: sip:[email protected];tag=7a60bebc-3202-4a55-998e-13d932066232
To: sip:[email protected];tag=789669197
Call-ID: 14e790c0-ab0f-4eb4-8a98-83997befca9a
CSeq: 44168 OPTIONS
User-Agent: Yealink SIP-T26P 6.72.0.76
Content-Length: 0

I am not on-site so am using a remote phone at a different IP address than the PBX.
To be clear, Everything was working then we did maintenance on users and extension, and lost outbound. The PBX is NAT with it’s own outside IP address.

What you posted is just qualify requests to the Flowroute trunk and your extension. It does not show an attempted call. You want to post the outgoing INVITE and the resulting responses. After making the failing test call, you should find what you need in the regular Asterisk log.

This is head of the pastebin file I just created. I made a filed call, got the call id by grepping the destination phone number, then grepped the “C-0005f04a” from /var/log.asterisk/Full

Is this what you are expecting?

  1. [2020-05-29 09:45:03] VERBOSE[3529][C-0005f04a] pbx.c: Executing [3035555555@from-internal:1] Macro(“PJSIP/204-0005c924”, “user-callerid,LIMIT,EXTERNAL,”) in new stack

  2. [2020-05-29 09:45:03] VERBOSE[3529][C-0005f04a] pbx.c: Executing [s@macro-user-callerid:1] Set(“PJSIP/204-0005c924”, “TOUCH_MONITOR=1590767103.382226”) in new stack

  3. [2020-05-29 09:45:03] VERBOSE[3529][C-0005f04a] pbx.c: Executing [s@macro-user-callerid:2] Set(“PJSIP/204-0005c924”, “AMPUSER=204”) in new stack

  4. [2020-05-29 09:45:03] VERBOSE[3529][C-0005f04a] pbx.c: Executing [s@macro-user-callerid:3] Set(“PJSIP/204-0005c924”, “HOTDESCKCHAN=204-0005c924”) in new stack

  5. [2020-05-29 09:45:03] VERBOSE[3529][C-0005f04a] pbx.c: Executing [s@macro-user-callerid:4] Set(“PJSIP/204-0005c924”, “HOTDESKEXTEN=204”) in new stack

https://pastebin.freepbx.org/view/b9a8e557#L22

You somehow filtered the desired (SIP trace) info out of the log.

Please post the entire log starting with the first entry for 09:45:03 and ending with the last. (Note that the SIP entries don’t have timestamps.)

https://pastebin.freepbx.org/view/96a3957d

As you have asked about weird, it seems yes on PJSIP logger:

From: sip:ping@invalid;tag=uloc-5ec4be7d-18-c68d3a1-ffeb45d6-f6d3697d
To: sip:[email protected]:5060;line=gqhogmf

I think will be more easy if you can block ports or on configuration access control on endpoints
You can look to following url.
https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples

Line 383:
SIP/2.0 403 Destination Blacklist - [email protected]
says the destination was invalid. However, line 340
From: <sip:[email protected]>;tag=f08f03a6-bc26-4f78-abf9-b7768ba2e52b
is an invalid caller ID and I suspect that’s why Flowroute rejected the call.
Normally, the outbound caller ID would be the External CID value in the extension settings, if not present the Route CID in the Outbound Route, or if that’s also not present, the Trunk CID.

Which of those do you have set?

I just found that the Outbound Route had the CID set to F1 Systems. I just removed that and applied to no effect. Reboot?

Removed the CID from outbound route and restarted Asterisk. No change.

grepped the call id of this failed call
https://pastebin.freepbx.org/view/c3b7334f

I want to thank you folks for your help. This a lesson in getting ahead of your self.

What happened:
Trunk provider noted a burst of unauthorized calls.
Looking into the calls we changed extension and user configurations.
Users reported no out bound calls.
Called trunk provider, put on hold and never answered (short staff no doubt)
We start digging and posting. Didnt call provider back, portal showed no warnings.

A number of things have been pointed out about my config and these are being addressed. Thanks again.

An email from trunk provider had gone to spam and we did not see the notice that they had blocked US calling. Error 21 might indicated that your outbound trunk is blocked as mine was.

2 Likes

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.