I inherited a freepbx system at this site I work at. I have no idea how it works and am not really familiar with PBX in general. It has been a pretty easy to maintain system as it has not given many fits and generally you use the UI to add new phones and users etc. Not to bad.
Though last night at about 2am users noticed calls could no longer be made from phone to phone or off site, and calling the site it goes instantly to voice mail options (which it was supposed to only do during site closed hours). The going to voice mail options asks users to press a button to ring certain parts of the site, though the system ignores these button presses. I was looking at logs from the machine itself in /var/log/asterisk/full
and saw that sometime the normal logging of reports like:
[2021-05-30 03:31:56] NOTICE[7296] res_pjsip_exten_state.c: Extension state subscription failed: Extension 307 does not exist in context ‘from-internal’ or has no associated hint
[2021-05-30 03:31:56] NOTICE[21900] res_pjsip_exten_state.c: Extension state subscription failed: Extension 99213168 does not exist in context ‘from-internal’ or has no associated hint
[2021-05-30 03:31:56] NOTICE[2471] res_pjsip_exten_state.c: Extension state subscription failed: Extension 9924168 does not exist in context ‘from-internal’ or has no associated hint
[2021-05-30 03:32:01] VERBOSE[32155] res_pjsip_registrar.c: Added contact ‘sip:[email protected]:5060’ to AOR ‘102’ with expiration of 900 seconds
[2021-05-30 03:32:01] VERBOSE[10792] res_pjsip/pjsip_configuration.c: Contact 102/sip:[email protected]:5060 has been created
[2021-05-30 03:32:01] VERBOSE[10792] res_pjsip/pjsip_configuration.c: Contact 102/sip:[email protected]:5060 has been deleted
[2021-05-30 03:32:01] VERBOSE[6299] res_pjsip/pjsip_configuration.c: Contact 102/sip:[email protected]:5060 is now Reachable. RTT: 62.482 msec
[2021-05-30 03:32:01] NOTICE[14560] res_pjsip_exten_state.c: Extension state subscription failed: Extension 307 does not exist in context ‘from-internal’ or has no associated hint
[2021-05-30 03:32:01] NOTICE[13485] res_pjsip_exten_state.c: Extension state subscription failed: Extension 99213*102 does not exist in context ‘from-internal’ or has no associated hint
These kinds of lines above are in most of the /var/log/asterisk/full*
But sometime last night the full log started to show:
[2021-05-31 03:17:12] ERROR[2471] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:[email protected]:5060
[2021-05-31 03:17:12] ERROR[32155] res_pjsip.c: Unable to retrieve PJSIP transport ‘asterisk’
[2021-05-31 03:17:12] ERROR[32155] res_pjsip.c: Unable to retrieve PJSIP transport selector for endpoint 162
[2021-05-31 03:17:12] ERROR[32155] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:[email protected]:5060
[2021-05-31 03:17:13] NOTICE[14890] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #587)
[2021-05-31 03:17:13] NOTICE[14890] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #587)
[2021-05-31 03:17:14] WARNING[14890] chan_sip.c: Remote host can’t match request REGISTER to call ‘74f3c0850671784360f3b33a1904675b@[::1]’. Giving up.
[2021-05-31 03:17:14] WARNING[14890] chan_sip.c: Remote host can’t match request REGISTER to call ‘762a54e810916972749388ba74c4d116@[::1]’. Giving up.
[2021-05-31 03:17:14] ERROR[4272] res_pjsip.c: Unable to retrieve PJSIP transport ‘asterisk’
[2021-05-31 03:17:15] ERROR[26816] res_pjsip.c: Unable to retrieve PJSIP transport ‘asterisk’
[2021-05-31 03:17:15] ERROR[26816] res_pjsip.c: Unable to retrieve PJSIP transport selector for endpoint 163
I have verified our sipstation account is paid up, though the screen in freepbx UI under community->sipstation shows a red error that ‘the server is currently not responding, it is either unavailable for or access is being blocked. If server is unavailable please try again later’
Not sure which server, looked at sagomas uptime and nothing seems to be down. Not sure how to test or what server it is trying. I pinged trunk1 DOT freepbx DOT org and trunk2, it worked.
I see nothing else wrong, and not sure what to check next or do. Or how to even test. It is odd I can call site and the voice mail does answer, though pushing buttons and extensions does nothing and no one can call offsite. No changes were made to the server which has an uptime of 3 years +.
Do I want to just do a sudo shutdown -r now ? Anyway I can test from the ui to get an error report of why calls cannot go out and why our site doesn’t ring and goes right to voicemail (of which the menu options are meaningless) Any other logs to look in ? (not sure what I am even looking for)