Recently, making changes on any of my hosted servers causes all the extensions on that particular instance to show as unavailable in Asterisk Info>Peers. Basically happens as soon as I hit ‘Apply Changes’.
The extensions can still dial out but cannot receive calls.
A reboot of the system brings everything back up, no problem.
Not seen this before and is happening across FreePBX 14 and 15 on both asterisk versions 13 and 16.
I saw a similar thread but it was closed. Is anyone else experiencing this?
Settings >> Asterisk SIP Settings >> chan_pjsip >> Allow Transports Reload need to be set to “No”
Thankyou both, seems to have resolved the issue. New default in 15 I guess?
I just spun up a new 15 install, and it was defaulted to yes.
Interesting, This feels like something that’s slipped through the cracks while testing and has been pushed to stable by accident. Can anyone on dev team confirm?
The tooltip literally says “Enabling this is not recommended, and may lead to issues.”
just checked and this is reported to be fixed as of sip settings 184.108.40.206 / 220.127.116.11
it is reportedly fixed in sipsettings 18.104.22.168 / 22.214.171.124
this may not do much for those installing from ISO with the older version;
ill try to take a look at it but my hunch is if the old module is in place when oobe for firewall completes at initial login and the option to configure sip settings is taken that its making that choice for you and setting it to yes - that said i dont see the fix actually changing the value and if so anyone installing from the ISO will have that errant config and the update may be pointless until a new ISO is published
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