PIONEERS VitalPBX 2.3.8

hawk#1

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VitalPBX seems to be here to stay. Might I suggest a new forum topic? Maybe PIAF6 ?
I think it should have a totally different forum topic name. Something like Incredibly Vital Installed Pbx 1 (IVIP1) or something else.
 
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wardmundy

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Coming Soon: New SMS Toolkit for VitalPBX with Skyetel, Vitelity, and VoIP.ms DIDs

  1. @help – Play a list of supported SMS commands
  2. @news – Retrieve latest news headlines from Yahoo
  3. @weather – Retrieve weather report by zip code
  4. @wolfram – Siri-like query to Wolfram Alpha
  5. @whitelist – Whitelist a new IP address in your firewall
  6. @disa – Use DISA calling from your smartphone with password
  7. @cf on – Enable call forwarding from PBX extension to smartphone
  8. @cf off – Disable call forwarding from PBX extension
  9. @asteridex – Use AsteriDex Voice Dialer to place a call from PBX
  10. @fail2ban+ XXXXX – Restart Fail2Ban with PIN: XXXXX
  11. @fail2ban- XXXXX – Stop Fail2Ban with PIN: XXXXX
  12. @firewall+ XXXXX – Restart FirewallD with PIN: XXXXX
  13. @firewall- XXXXX – Stop FirewallD with PIN: XXXXX
  14. @asterisk+ XXXXX – Restart Asterisk with PIN: XXXXX
  15. @pbxstatus – Returns status of Asterisk, Fail2Ban, and FirewallD
 
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krzykat

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Coming Soon: New SMS Toolkit for VitalPBX with Skyetel, Vitelity, and VoIP.ms DIDs

  1. @help – Play a list of supported SMS commands
  2. @news – Retrieve latest news headlines from Yahoo
  3. @weather – Retrieve weather report by zip code
  4. @wolfram – Siri-like query to Wolfram Alpha
  5. @whitelist – Whitelist a new IP address in your firewall
  6. @disa – Use DISA calling from your smartphone with password
  7. @cf on – Enable call forwarding from PBX extension to smartphone
  8. @cf off – Disable call forwarding from PBX extension
  9. @asteridex – Use AsteriDex Voice Dialer to place a call from PBX
  10. @fail2ban+ XXXXX – Restart Fail2Ban with PIN: XXXXX
  11. @fail2ban- XXXXX – Stop Fail2Ban with PIN: XXXXX
  12. @firewall+ XXXXX – Restart FirewallD with PIN: XXXXX
  13. @firewall- XXXXX – Stop FirewallD with PIN: XXXXX
  14. @asterisk+ XXXXX – Restart Asterisk with PIN: XXXXX
  15. @pbxstatus – Returns status of Asterisk, Fail2Ban, and FirewallD
No Brainer to have.... also - how about some optional preemptive outbound SMS - Cron jobs could check for:

1) High CPU
2) Low Disk Space
3)Trunk / Extension unreachable -would require programming on VitalPBX side to do it right with option of "SMS Notification Unreachable: <Number to Notify>"
 
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smarks

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How about setting up ssh keys and disabling passwords on root logins.
Accessing root accounts over the internet with passwords is just plain insecure.
I agree that moving the ssh port off of port 22 is a good idea but I can also tell you from experience the bad guys will eventually find your alternate port. I have had ssh on alternate ports for the last 20 years or so but in the last 2 or 3 years the bad guys are finding the alternate ssh ports much more frequently.

Regards,
Tom
You typically want to change SSH ports to reduce the noise from all the scanner bots hammering port 22. The goal is to decrease logging activity and network activity and therefore load. Less activity also means less chance of a compromise but that is not the primary goal imo. You're never going to be able to hide from all of them but anything is better than port 22.

On nodes hosting multiple VPS servers the load can get quite high from all the port 22 scanning activity.
 
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kenn10

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@wardmundy I am testing the outbound CID Superfecta for calling name ID on outbound calls. While the code does the lookup and inserts the called party name into the CDR and database, it does not update the phone display. Its apples and oranges, I know, but IncrediblePBX 16-15 with the outbound CID Superfect does update the display.

Looking at the logs of the outgoing call, I see the lookup against OpenCNAM and the insertion into the CDR and phonebook database so I know the lookup works. The phone is PJSIP and the Skyetel trunk is PJSIP. The display flashes and changes the 10-digit dialed number to a 1+original number so I know something in VitalPBX is pushing a display update to the phone. I just am not getting the name. Is this a VitalPBX limitation?
 

wardmundy

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@kenn10: See if anything in this thread helps.

I think this is typically a phone setting. On Yealinks for both inbound and outbound calls, it's set with Settings:Call Display:Call Information Display Method
 
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kenn10

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@kenn10: See if anything in this thread helps.

I think this is typically a phone setting. On Yealinks for both inbound and outbound calls, it's set with Settings:Call Display:Call Information Display Method
No. Its an Aastra 9143i which works perfectly with FreePBX. The main line of the phone is on IncrediblePBX 16-15 and another line on the VitalPBX. I suspect there are some system settings akin to the FreePBX Settings>Advanced Settings that probably are not adjustable through the VitalPBX GUI.

I'll be curious to see if this works for anyone else who has phones on VitalPBX. I also have a Fanvil X4 hooked up to VitalPBX with the same results.
 

kenn10

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@kenn10: You might want to open a ticket with VitalPBX on this one.
VitalPBX bug report 0000148 submitted. The issue with "s" showing in the display instead of the dialed number became an issue when upgrading software from 2.3.8-1 to 2.3.8-3. However, the outbound caller-id lookup does not work for me on 2.3.8-1 either so far as updating the phone display with the name of the party you called.
 
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krzykat

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It seems the only way to do the HA is with a floating IP that must be in the same datacenter. To me, that rather defeats one of the major points of failure. If the datacenter is under a DDOS attack, or power issue or network issue, the whole datacenter is down, so both parts of the HA are down. I'd love to have the ability to have one server in Atlanta and another in Miami datacenters for example - but is that possible?
 
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wardmundy

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From my vantage point, HA isn't as crucial as it used to be. Almost all major SIP phones now have a failiover option on the phone itself. If your voicemails are delivered by email, then failover to another site is as easy as keeping both servers up to date. There's no voicemail to worry about. When the primary server fails, all of your phones will reconnect to the secondary server in less than a minute. And providers such as Skyetel provide similar failover services for trunking.
 

dhoppy

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From my vantage point, HA isn't as crucial as it used to be. Almost all major SIP phones now have a failiover option on the phone itself. If your voicemails are delivered by email, then failover to another site is as easy as keeping both servers up to date. There's no voicemail to worry about. When the primary server fails, all of your phones will reconnect to the secondary server in less than a minute. And providers such as Skyetel provide similar failover services for trunking.
So, do you backup the primary and restore it to the secondary server once a week, for example, so that changes to voicemail greetings, follow me settings, etc. on the primary server are always up to date on the secondary server?
 
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