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Never used Oracle cloud, as I'm still bitter about Sun Microsystems, but isn't it like AWS where you have to open ports or attach an IP address to the instance? Is the server receiving HTTP packets? On AWS you just run the installer, open the necessary ports, update the IP address variables to...

It's handset dependant, the phone is supposed to decide what is a missed call and what is not. From packet capture you should really be seeing a CANCEL with a reason of 'Call Completed Elsewhere' in the event of simultaneous dialling. That SIP message is supposed to be handled by the phone to...

Think it's all done in PHPMailer. Nothing in Email Logs in the web interface?

1. Rename the domain in Advanced -> Domains. Ideally this will be the FQDN you will ultimately be accessing the service from...however the public IP is acceptable if you don't have a domain name, and don't care about HTTPS. 2. In Variables, update the external_sip_ip and external_rtp_ip to your...

Yeh, you certainly can reprovision without rebooting. However there's a ton of parameters that Yealink's use that require the reboot, so it would come with a whole host of caveats. If you're happy with that, you should just be able to edit the provision/index.lua script so it doesn't send a...

The device needs the user too, not sure the extension matters (but you need that so they can log into the portal). Personally, I disable the permissions as not a single one of my customers cared about the feature and it only resulted in the odd extension being missed during prep.

Have a check of /etc/fusionpbx/config.php (or whatever it is nowadays). See if the password is correct when you give it a test with psql.

They probably used the Yealink's built in Hot Desking method, which isn't the best really as it requires giving the SIP credentials to the end users. You can mitigate these issues with a Trusted ACL/VPN/Local Network only... but it just gives me the ick knowing the password for the account will...

It's optional now. Speaking of which, I think the documentation might need a bit of seeing to with this change. I might submit a pull request in a bit. God send for me really, hundreds of domains and not a single one ever really wanted to segment off their contacts to specific users. They all...

It should be fine I should imagine, We used to do the same thing on an older PBX.

I'm 90% sure. That database contains the active state of sessions and such, I think what's happening is your phone is only listening to one of the servers (I would have to guess that last one in the list) as they are the same domain for failover. Weird idea, but it might even work if the...

Can't verify it for all models, but Line 0 means whatever my active line is. (Left and right arrows to switch between them) For your HA setup, you are replicating the core database too right? By default it's stored as a file in /var/lib/freeswitch/db/ but can be put into a postgresql database...

You can also enable the agent support for BLF subscribes in the /etc/freeswitch/autoload_configs/lua.conf.xml by uncommenting the startup-script: <!-- FusionPBX: Support BLF for Call Center Agents --> <!--<param name="startup-script" value="blf_subscribe.lua agent"/>--> Restart...

Getting closer, have you got the keys set to the correct lines? I think you *can* do the Line 0 trick with BLFs, but I'm not sure.

As for the lua.conf.xml, that's perfectly normal unless you want the 'extended' uses of BLFs (i.e, monitor a call flow or a DND status). They're off by default as it does vary wildly from vendor to vendor.

Hmmm, that's interesting. If I had to guess: $yealink_rport That's off by default on Yealinks, but is almost mandatory in most setups. However it could be a SIP ALG thing.... Possibly: features.blf_led_mode = {$yealink_blf_led_mode} if you have change the default settings in the Yealink...

I believe it's a SIP INFO packet, with dialog+info XML inside it. It's very strange it works on some models but not others.

Normally a SIP Provider will be able to give you a list or prefix of IP addresses to whitelist, instead of opening it up to the entire internet.

Basically, it's a built in feature of mod_sofia for FreeSWITCH that doesn't appear to have anyway of disabling (if I recall, I haven't researched the source in a while). If the call is completed elsewhere, FreeSWITCH will send the reason 'Call Completed Elsewhere' and your phone will be...

You appear to be behind NAT, have you set your ext-sip-ip and ext-rtp-ip to be the public address you are presenting from? A 30 second termination means a packet is not being ACK'd and FreeSWITCH tears down a call after 30 seconds believing it to be invalid. You should be able to see this...