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I just emailed you and Thank you

Sure, where do i send the info - I may need to send you some $$$ too :)

I guess I am going to have to destroy the server then rebuild it. I didn't make a back up :(

root@npusa:~# systemctl status freeswitch.service ● freeswitch.service - freeswitch Loaded: loaded (/lib/systemd/system/freeswitch.service; enabled) Active: failed (Result: start-limit) since Sat 2018-02-24 20:34:11 UTC; 13s ago Process: 710 ExecStart=/usr/bin/freeswitch -u freeswitch -g...

This is my switch.conf.xml file and I checked in settings to listen on 8021 <configuration name="switch.conf" description="Core Configuration"> <cli-keybindings> <key name="1" value="help"/> <key name="2" value="status"/> <key name="3" value="show channels"/>...

Freeswitch will not load Can not access fs_cli Connection to Event Socket failed. I have rebooted and no sucess Any help would greatly be appreciated

I also get this error on a fresh install a few lines up 2018-02-23 16:49:18.583712 [NOTICE] switch_loadable_module.c:483 Adding Chat interface 'api' 2018-02-23 16:49:18.585401 [ERR] mod_avmd.c:768 Couldn't load XML configuration! Loading default settings 2018-02-23 16:49:18.585448 [NOTICE]...

I have spun up 3 instances - Fresh Installs - of Fusionpbx and get the error message in the log viewer on every one of them. Is this something to be concerned with? It appears at the bottom of the log viewer on first install - The IP of the server is207.246.117.8...

I did the global sip trace on and made some calls and this was the results . . . (never saw any)

I tried dummydummy and the ani shows dummy. It still doesn't show the caller ID. I tried both ways.

see below

Yes I do. It still shows Freeswitch as the ANI - I left username and password blank on the gateway since VI uses ip Auth

Here is a paste from the log viewer of the failed calls I am calling DID 770-827-4079 from ext 101 @207.246.117.8 9ac0728c-972d-4cc5-b997-121563bb6fbc Dialplan: sofia/internal/101@andrewbyrd.netphoneusa.net Regex (FAIL) [user_record] ${user_exists}(false) =~ /^true$/ break=never...

I just did on your advice. Tried a few test calls and then asked Voip Innovations what they saw on their end. This is what they said Andrew, Still seeing Freeswitch as the ANI and G722 being used as the media codec, however, the RPID field is now showing as just the 10 digit number. Thank you

I checked with Voip Innovations this is what they said G722 is not an audio codec that we support for our services. We only support G711ulaw/alaw, G729 and RFC 2833. You will want to remove G722 as a media codec from your trunks I went into sip profiles and changed the outbound codec to PCMU...

I have a spa 508 and a spa525g2 If I select the preferred codec on the phone itself to g722, the call will take 5 - 20 seconds to initiate If I select 711u as the preferred the call goes through immediately Any ideas what I should do here? I would like to use 722 if possible

I changed it to a comma, saved, rebooted the server, and then did a test. I have Ext 101, 102 and 103 set up on 1 phone. I had a call in progress on line 1 and instead of rolling over to line 2 it sent me to voice mail

I charge my clients a flat rate of $24.99/per phone per month. I choose low usage clients. I have 75 clients currently on 75 different Freepbx instances. You can imagine my pbx maintenance nightmares That is why I am trying to make fusion work for me so I can have a MT solution.

Hi DigitalDaz - I appreciate you trying to help. I am not real clear on what you are saying to do here This is where I went if (r.dialplan_detail_tag == "condition") then if (r.dialplan_detail_type == "destination_number") then...

Why do you use a different gateway for each client? Does that mean you have 10 different providers?