SOLVED TLS/SRTP on Digium Phones? SRTP not working

https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phones+and+FreePBX#DigiumPhonesandFreePBX-DigiumPhonesLineOptions

Updates made there don’t push reconfiguration notices to the phones. You’d have to change, red button, then go send a reconfigure to the phone.

UGH!!! Smacking my head!

I must have read that 4 times… got it in my head for some reason that that was only an option in switchvox and not in FreePBX. Why? who knows but I was looking of the line option in the Digium Phones tool, not the extensions…

Malcolm THANK YOU!!! it’s always the simple things, sometimes just requires a second set of eyes.

Now I have two shields and calls go through.

Now to tackle the next issue… the phones don’t show up in the Digium Phones config as registered…

in the asterisk log I see the following ever few seconds…

Updating DPMA user ‘302’ uri=‘pjsip::41341;transport=TLS’ ua=‘Digium D62 2_8_5’
Updating DPMA user ‘303’ uri=‘pjsip::60237;transport=TLS’ ua=‘Digium D62 2_8_5’

josephoatpbx*CLI> digium_phones show sessions
---- Digium Phone Module Active Sessions ----
SessionID:4848677081758440980 SecondsAlive:648 SecondsLastActivity:633 URI:pjsip::60237;transport=tls Auth:Yes Inactive:No Configured: Yes MAC:000FD30C8707
SessionID:8560459041898512846 SecondsAlive:700 SecondsLastActivity:700 URI:pjsip::33344;transport=tls Auth:Yes Inactive:No Configured: Yes MAC:000FD30A8813

Marking this as resolved and starting a new topic for the PJSip Phone not showing up as registered in Digium Phone Module

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