Updating configs drops grandstream phones connecting via sip-tls

hi, i have a new install of freepbx 13 (Asterisk 13.14.0) and about 15 grandstream phones. the phones connect using pjsip/tls and srtp. the phones seem to be working good, but i have noticed that if I make a change (any change) on the pbx that requires me to hit the ‘apply config’ button, all the phones go into a “unavailable state” according to “pjsip show contacts”. at this point, the phones seem fine but the pbx has dropped them and calls go directly to voicemail. the only way I can recover is to reboot all the phones.

applying the config (asterisk re-load config) does NOT drop the sip udp (non tls) connections I have.

I would appreciate help from anyone who has experience with this

thanks for your time!

The FreePBX peer support forum is http://community.freepbx.org/ and this sounds like it is the result of how FreePBX works.

yes i did start there. they seemed to think that it was an asterisk issue…I put a bug report in and this was the response

Freepbx does not control what happens during a reload. If you hit reload and your extensions are disconnecting it would be an asterisk issue. Not a freepbx one.


fyi, i talked more to the freepbx people, and they say that they are doing a ‘core reload’…they feel that what happens during a core reload is an asterisk issue…kind of makes sense?

this issue is very repeatable, is there something I can provide to narrow down the issue?

thanks for your help!

Ideally you need to reproduce this using Asterisk CLI, or a simple AMI script, to request the reload.

Either way you need to set logging verbosity to at least five and provide a log showing that a reload, and not a restart, has happened.

If Asterisk is actually crashing, you will need to substitute a debug build of the same version of Asterisk, and provide the backtraces from the core dump.

David, I would be happy to do this…I should have it tonight