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on my sp 321 I'm using udponly transport for both Line 1 Server and under Sip Servers Server 1

1A2 Haven't thought about it in so many years! 25pr cable per phone seems like miles of cross connect wire on a large install

Do you have SIP ALG disabled in your router/firewall?

Mom & Dad retired in the mid 70s and moved to a rural part of the county. Applied for phone service, 3 months later they were able to be added to a 10 party line!

I use Ring Groups to accomplish this: Inbound route for each DID xxx xxx 1234, xxx xxx 2341, xxx xxx 3412 Each DID has a rg point the ivr option to the desired rg repeat for the other 2 DIDs rg setup rg description xxx xxx 1234 enter the cell phone # followed by # in the extension list...

I setup Magic Mirror for a client. 50" TV and a wireless raspberry pi 4 running MM strapped to the back.

Are you running a script to update the external ip in the asterisk sip settings?

Did the external IP change?

Used rEFIt (now forked to rEFInd) and Clover serval years ago when I was building Hackintosh systems.

In January I moved a small client from Ubuntu to Zorin. Since then they have dropped all MS products and haven't looked back.

I only use Ubiquiti routers for myself and clients

Incredible 2020.1 Centos 7 Asterisk 16.9.0 Incredible GUI 15.0.22.29 Skyetel V 2.0 regions down - hosed iptables - any suggestions on how safely delete and reenter?

@wardmundy said, "If you're seeing s@from-sip-external with incoming calls, then the PBX can't decipher the DID of the incoming call."

More of the log would probably show that the DID is not matching what is defined in the inbound route

Tried the 2 trunk Skyetel Regions 2.0 still red and Kenn's 1 trunk Regions 2.0 still red Router and PBX have Skyetel IPs whitelisted Restarted router and pbx between configuration changes

Should one configure a two trunk or @kenn10 1 trunk for both in and out

Did you install freepbx distro or an incredible pbx flavor? extension configured for sip or pjsip? sep.conf.xml adjust for the channel driver in use 5060 for sip. 5061 for pjsip <sipPort>5060</sipPort> ? <securedSipPort>5061</securedSipPort> ? also try changing...

Try something like this in your search engine SIP NAT traversal site: pbxinaflash.com/forum

make a backup of /etc/asterisk/odbc.conf remove or change the reference to 222 & 223

@GI-Tech try editing the advanced settings of an extension extensions --> advanced tab change MWI Subscription type to Unsolicited