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  1. FreePBX
  2. FREEPBX-23631

When upgrading Asterisk SIP Settings breaks outbound calls for ATT trunk



    • Type: Bug
    • Status: Closed
    • Priority: Minor
    • Resolution: Fixed
    • Affects Version/s: 16.0.21
    • Fix Version/s: None
    • Component/s: Asterisk SIP Settings
    • Labels:
    • ToDo:
    • Asterisk Version:
    • Distro Version:
    • Distro:
      FreePBX Distro


      Had a system suddenly stop working for outbound calls. Getting an all circuits are busy now. Very basic trunk setup through ATT, no registration etc, I just point it at the internal gateway. SIP server, SIP Port, UDP and only Ulaw selected.

      Since I could not figure out what happened to current system, with paid for modules etc, I did a fresh install with only an extension, the ATT trunk and an outbound route. Outbound calls now worked again. I found the moment I upgrades the Asterisk SIP settings module from 16.0.19 to 16.0.23 outbound calls broke once again. I tried 16.0.22 and the same thing happened. 

      Logs don't give me much of a clue, just the fact the it can no longer connect.
      13999[2022-07-28 00:00:03] ERROR[2597] res_pjsip.c: Endpoint 'ATT': Could not create dialog to invalid URI 'ATT'. Is endpoint registered and reachable?
      14000[2022-07-28 00:00:03] ERROR[2597] chan_pjsip.c: Failed to create outgoing session to endpoint 'ATT-7758261113'
      14001[2022-07-28 00:00:03] WARNING[6237][C-00000005] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
      14002[2022-07-28 00:00:03] VERBOSE[6237][C-00000005] app_dial.c: No devices or endpoints to dial (technology/resource)
      14003[2022-07-28 00:00:03] VERBOSE[6237][C-00000005] pbx.c: Executing [s@macro-dialout-trunk:37] NoOp("PJSIP/7099-00000007", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3") in new stack
      14004[2022-07-28 00:00:03] VERBOSE[6237][C-00000005] pbx.c: Executing [s@macro-dialout-trunk:38] GotoIf("PJSIP/7099-00000007", "0?continue,1:s-CHANUNAVAIL,1") in new stack
      ATT said they never got any data when I try and call out so the system does not seem to be connecting. What's a bit odd is I can setup a voip.ms connection with a username and password and that seems to work.  Going to continue trying settings to see if I can find the underlying cause but so I could not see anything really change under Settings -> Sip Settings.

      Rolling back to .19 does not seem to fix it either. Once I upgrade the issue persists. 



        Gliffy Diagrams




              • Assignee:
                jphilip Philip Joseph
                jraddigan Koinonia Services
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                • Created:


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