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Type:
Bug
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Status: Closed
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Priority:
Minor
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Resolution: Fixed
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Affects Version/s: 16.0.21
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Fix Version/s: None
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Component/s: Asterisk SIP Settings
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Labels:None
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ToDo:
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Asterisk Version:16.25.0
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Distro Version:FreePBX 16.0.21.8
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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.