FreePBX 15.0.17.12 Current Asterisk Version: 13.38.1
This morning a client reported none of their phones were connected. These are all Digium D62 and D65 phones configured with EPM. Selecting “use old server” was successful in brining the phones up and connecting to the PBX.
I did a factory reset on a phone they weren’t using currently and it will not come up. Phone displays “Error handshaking with proxy. Choose a new …”
I’ve updated endpoint manager to the latest edge version. still no good. All of the extension in EPM are showing as yellow and needing an update. If I go into the template for digium phones and select. “Save, Rebuild Configs and update phones” it crashes.
for some reason login for provisioning isn’t working. I try to go to the provisioning address in a web browser using the username/password set in system admin > provisioning protocols. the ‘xxx’ and 'yyyyyy" below are purposely changed for obvious reasons but I’ve double checked the username and password from provisioning protocols…
in the past I’ve seen where it will actually list the files…
Are you saying that the IP address won’t work? I’ve tried with both http and https. I can understand where https won’t work without the certified FQDN, that makes sense now that you say it.
you’re link example worked (with http). however if I substitute the macaddr.cfg that the phone is looking for it fails
If you’re using DPMA for your phones, then they are not provisioned using http, they provision using SIP Message packets, which is what you Option 66 string shows.
I entered a ticket for the error at the top of the ticket but not for the “Migrate to DPMA” errror yet.
One thing that occurred to me. these phones are using chan_sip which is using port 5160. DPMA Managment specifies port 5060 (which in Freepbx is not assigned to chan_pjsip).
If I change the DPMA Management setting to port 5160 and the Option 66 accordingly, would this solve this issue? or is the port in DPMA Management in separate from the chan_sip port?
Hi @ashcortech DPMA allows phones to use the SIP port to communicate with asterisk to fetch the configuration so depends on your situation, you can use the port.