GOOD NEWS The Migration Begins

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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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ariban

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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?
 

wardmundy

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When do We get this?
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.

 

wardmundy

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

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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.
 
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wardmundy

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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?
Will let you know in the morning. :biggrinjester:

10/17 7:50am UPDATE: Still up and running 12 hours later.
 
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wardmundy

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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.
 

krzykat

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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.
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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wardmundy

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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
 

wardmundy

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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
 

HT Greedy

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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
 

wardmundy

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

shetu

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I got these error.
Code:
asteriskinfo page not open from WebGui
 Whoops \ Exception \ ErrorException (E_WARNING)
file_get_contents(http://[email protected]: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!"
 
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wardmundy

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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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