FYI Asterisk 12 + FreePBX 12 Beta

wardmundy

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I'm liking what I see so far. Any reason why I cant upload and restore settings from a Green install?


You gotta get it up to FreePBX 12 first. Then it's worth a try. It's on my list, too.
 

jeff.h

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Oh and what about Travelin' Man? Any reason not to install v3 or v4?
 

jeff.h

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Oh no, the auto configure button is absent in the SIP Settings page :cryin:
 

darmock

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there is also an update to asterisk that I am working on so you all have the latest release will take a bit to get it out there but should not take long... Just an incremental version

Tom
 

rossiv

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Haven't tried 64 bit, but 32 bit is on par with lgaetz 's timing. 63 minutes EXACTLY. I would have thought that my 1Gbps connection would have helped, drat. :)
fzpvADp.jpg
 

wardmundy

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Old Chinese Uncle Wardy Proverb:

It's not the time it takes to put it in that matters.
What's important is what you do with it after that. :sorcerer:
 

ZeekWorld

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Hi,
I've tried installing PIAF Black on CentOS 64bit this morning and is now up and running.
The only think I am wondering,.. incrediblepbx has not been adapted to Asterisk 12 correct?
I am currently running PIAF Green with incredibblepbx and fax and I was trying to clone my settings to the Beta PIAF BLACK

When tried to installed incrediblepbx11 got the following message:
Image%20189.jpg


Here the screenshot of PIAF Black with CentOS 64 bit up and running.
Image%20190.jpg
 

wardmundy

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Incredible PBX 12 is not yet ready, but we're working on it...

staytuned.png
 

tm1000

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If you have been playing with 12/12 versions of FreePBX/Asterisk, then you may have noticed that registering a SIP extension is not quite as straight forward as it once was. In advanced settings, there is a drop down menu for "SIP Channel Driver" which allows you to choose the new chan_pjsip driver, the old chan_sip driver or both drivers. If both are selected, then FreePBX offers both extension types when creating a new extension. In my initial testing I never got a SIP registration to work, only PJSIP. This post over on the FreePBX forum explains why, each service is bound to a different port. By default in PIAF Black, PJSIP registrations are accepted on 5060 and SIP registrations are accepted on 5061. When configuring extensions to register via SIP, you must specify the port number.


I am open to ideas about how to make this easier and better. I realize it's confusing. My thought is to have the port number announced on the extensions page.

When upgrading from 2.11 to 12 the port numbers will remain the same, this only applies to new installs.
 

tm1000

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Unrelated, but is there any reason for the average user to not use PJSIP exclusively?

There are things that don't work as well in pjsip at the moment. SMS is one of them. However here is a prime example of using both. On my development machine I have my trunks registered using chan_sip while my sip phones are all using pjsip, Asterisk does all the passthru work, so when I make a call it goes pjsip -> chan_sip seamlessly. The same could be said in reverse. I'm sure Cisco phones will have an issue, but then again they might not.
 

jeff.h

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* Unfortunately the patch for Asterisk 12.3.2 does not seem to exist on the Pbx in a Flash servers *YET*! :oops:
 

lowno

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Installing black does not install freepbx 12? Just tried, no freepbx 12 rc. Is there a step I missed?

UPDATE
Never-mind, I need to start installing PIAF the NEW way.
 

wardmundy

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Very weird. If you click on the link in your browser (after it fails), it pops right up. Thanks. Still investigating away...
 

krzykat

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Very odd - clicking the link fails, refresh fails. Typing http:// in front of the url and trying again worked ??
 

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