GOOD NEWS The Migration Begins

Discussion in 'Today's Tech News & Events' started by wardmundy, Oct 10, 2018.

  1. wardmundy

    wardmundy Nerd Uno

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  2. David Foxworth

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    And:

    Running as user 'asterisk'
    Running under group 'asterisk'
    Connected to Asterisk 16.0.0 currently running on pbxcloud (pid = 1130)
    pbxcloud*CLI> pj
    pjproject pjsip
    pbxcloud*CLI> pjsip list registrations

    <Registration/ServerURI..............................> <Auth..........> <Status.......>
    ==========================================================================================

    gvsip1/sip:eek:bihai.sip.google.com gvsip1 Registered

    Objects found: 1
     
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  3. wardmundy

    wardmundy Nerd Uno

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    [​IMG]
     
    #3 wardmundy, Oct 11, 2018
    Last edited: Oct 11, 2018
    krzykat, ariban and markieb like this.
  4. ariban

    ariban Member

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    does version 16 fix the gvsip problem that everyone is having on 13, where it disconnects every few hours and only amportal restart fixes that?
    any update script to go from 13 to 16 on incredible pbx?
     
  5. wardmundy

    wardmundy Nerd Uno

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    For the short term, I wouldn't count on Asterisk 16 fixing more things than it breaks. Google has broken GVSIP today without any help. Details here.
     
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  6. shetu

    shetu Member

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    When do We get this?
     
  7. wardmundy

    wardmundy Nerd Uno

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    Still some work to do. Asterisk 16 and FreePBX 15 have only been available for less than a week. FreePBX 15 is still Alpha code, and that's being charitable.

    [​IMG]
     
  8. wardmundy

    wardmundy Nerd Uno

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    With a little luck, it looks like we may be ready for the pioneers tomorrow. :arabia:
     
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  9. wardmundy

    wardmundy Nerd Uno

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    Found a deal breaker with the new 16-15 build. Stay tuned.

    UPDATE: Builds after 4 pm EDT should be fine now. Some TTS apps are broken until a new version of FLITE is released. Pico TTS seems to be fine.
     
    #10 wardmundy, Oct 16, 2018 at 3:43 PM
    Last edited: Oct 16, 2018 at 5:34 PM
  10. wardmundy

    wardmundy Nerd Uno

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    Will let you know in the morning. :biggrinjester:

    10/17 7:50am UPDATE: Still up and running 12 hours later.
     
    #11 wardmundy, Oct 16, 2018 at 7:37 PM
    Last edited: Oct 17, 2018 at 7:51 AM
  11. wardmundy

    wardmundy Nerd Uno

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    10/17 UPDATE: Fresh build this morning fixed several bugs. Your best bet is to reinstall.

    1. Asterisk's app_macro is now a deprecated component, but FreePBX uses it for most dialing functions. It now has been added in menuselect.
    2. FLITE is broken at the moment, but we've added picotts for TTS support and reworked a number of Incredible apps. IBM TTS is even better.
    Review Implementing IBM TTS and Voice Recognition section of latest Issabel tutorial for IBM setup. Pieces already in place.
    3. ODBC support has been reworked for Asterisk 16 and FreePBX 15. Sample apps included:
    a. Dial 222 and enter 12345 for a sample employee lookup application using sample MySQL timeclock database.
    b. Dial 223 and enter 335 for a sample speeddialer using AsteriDex dial codes. This one gets you DEL for Delta Airlines.
     
  12. krzykat

    krzykat Guru

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    I can't believe that you even still have IGOR around any more. Why not just mod all the best of Incredible Apps to use a preferred TTS engine?
     
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  13. wardmundy

    wardmundy Nerd Uno

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    Incredible Fax works; however, fax detection appears to be broken so fax receipt requires a dedicated DID or Google Voice trunk. Before running the Incredible Fax installer, issue the following command:
    Code:
    yum -y install php-devel --enablerepo=remi,remi-php56,epel
     
  14. shetu

    shetu Member

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    Do I need re download this (http://incrediblepbx.com/incrediblepbx-16-15.tar.gz) or not?

    I got this error msg ->/var/lib/asterisk/bin/module_admin listonline > /dev/null 2>&1 (1)
     
    #15 shetu, Oct 17, 2018 at 12:58 PM
    Last edited: Oct 17, 2018 at 10:46 PM
  15. wardmundy

    wardmundy Nerd Uno

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    Yes. Redownload before new install.

    Added missing STUN server components this morning but you can manually add these to existing builds:
    Code:
    sed -i 's|;stunaddr = mystunserver.com|stunaddr = stun.l.google.com:19302|' /etc/asterisk/res_stun_monitor.conf
    sed -i 's|;stunrefresh = 30|stunrefresh = 30|' /etc/asterisk/res_stun_monitor.conf
    amportal restart
     
  16. HT Greedy

    HT Greedy New Member

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    Would this be the cause for GV call screening even though it is set to off. I had something odd happen today. I havent been able to receive inbound test calls from any of my telephone numbers but I received an inbound call from a friend. After receiving the call, GV still screens. Any Idea?

    Thanks

    H
     
  17. wardmundy

    wardmundy Nerd Uno

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    Not seeing that on our test machines. Call Screening is an option within the Google Voice Settings portal.
     
  18. shetu

    shetu Member

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    I got these error.
    Code:
    asteriskinfo page not open from WebGui
     Whoops \ Exception \ ErrorException (E_WARNING)
    file_get_contents(http://...@localhost:8088/ari/endpoints): failed to open stream: Connection refused
    Code:
    /usr/sbin/amportal: line 44: export: `16': not a valid identifier
    /var/lib/asterisk/bin/freepbx_engine: line 119: export: `16': not a valid identifier
    
    Code:
    fwconsole restart
    Asterisk is still running and we can't stop it!"
     
    #19 shetu, Oct 19, 2018 at 2:01 PM
    Last edited: Oct 19, 2018 at 2:22 PM
  19. wardmundy

    wardmundy Nerd Uno

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    @shetu: asteriskinfo appears to be broken. The '16' errors are harmless. The restart hangs seem to be a known problem is FreePBX 15 alpha. Only solution is to reboot after you've made several changes in the dialplan. My suggestion is to make all the changes you wish to make before clicking the red bar to reload the FreePBX dialplan. It won't hang the first time and usually not the second time. Third time gets into Buyer Beware territory.
     

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