FreePBX12 (IncrediblePBX) and Asterisk 13 on Pogoplug Pro V3

I agree with your opinion in general. However, for using plain Asterisk (I never really need or like FreePBX or other GUI versions of Asterisk) with no more than 3-4 simultaneous calls, i.e., for home use, I think Pogo or Dockstar should do fine.



Edited 1 time(s). Last edit at 04/27/2017 11:11PM by twinclouds.

twinclouds Wrote:
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> I agree with your opinion in general. However, fo
> r using plain Asterisk (I never really need or lik
> e FreePBX or other GUI versions of Asterisk) with
> no more than 3-4 simultaneous calls, i.e., for hom
> e use, I think Pogo or Docstar should do fine.

As I said, pogo should be OK as "light" PBX server, but not cheerful at all in office environment. this is why I had been using PC till satisfied with the performance of orange pi.

Did you set IVR for incoming call? my pogo didn't work well when IVR starting.



Edited 1 time(s). Last edit at 04/26/2017 06:23PM by asterix.

asterix Wrote:
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> tried debian8+freepbx 13 on startdock and pogoplug
> e02, just works, not recommended.
>
> you can try raspberry pi or orange pi pi, freepbx
> works well on both.
>
Sorry, what is startdock? It is a hardware like pogoplug?

Sorry. It should be Dockstar. :-)

habibie Wrote:
-------------------------------------------------------
> asterix Wrote:
> --------------------------------------------------
> -----
> > tried debian8+freepbx 13 on startdock and pogopl
> ug
> > e02, just works, not recommended.
> >
> > you can try raspberry pi or orange pi pi, freepb
> x
> > works well on both.
> >
> Sorry, what is startdock? It is a hardware like po
> goplug?

Sorry my fault. It's dockstar, didn't use it many years, so forgot the right name"dockstar". :)

twinclouds Wrote:
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> I agree with your opinion in general. However, for using plain Asterisk (I never really need or like FreePBX or other GUI versions of Asterisk) with no more than 3-4 simultaneous calls, i.e., for home use, I think Pogo or Dockstar should do fine.
>
I can confirm this. I had used a Seagate Dockstar and am using a Seagate GoFLEX Home (both have the same Marvel CPU clocked 1.2GHz) to handle all my PBX routings using a plain FreeSWITCH.

AFAICT, a device like Seagate Dockstar, GoFLEX NET/Home, any Pogoplug lines should work just fine as a plain (no GUI) PBX system and can easily handle 10+ concurrent calls sans transcodings.

habibie Wrote:
-------------------------------------------------------
> twinclouds Wrote:
> --------------------------------------------------
> -----
> > I agree with your opinion in general. However,
> for using plain Asterisk (I never really need or l
> ike FreePBX or other GUI versions of Asterisk) wit
> h no more than 3-4 simultaneous calls, i.e., for h
> ome use, I think Pogo or Dockstar should do fine.
> >
> I can confirm this. I had used a Seagate Dockstar
> and am using a Seagate GoFLEX Home (both have the
> same Marvel CPU clocked 1.2GHz) to handle all my P
> BX routings using a plain [url=http://freeswitch.o
> rg]FreeSWITCH[/url].
>
> AFAICT, a device like Seagate Dockstar, GoFLEX NET
> /Home, any Pogoplug lines should work just fine as
> a plain (no GUI) PBX system and can easily handle
> 10+ concurrent calls sans transcodings.

did you try IVR for incoming calls?

I made this process immensely more tolerable by using "screen" to put it in the background to prevent it from being interupted. "apt-get install screen'

Steve Wrote:
-------------------------------------------------------
> I made this process immensely more tolerable by us
> ing "screen" to put it in the background to preven
> t it from being interupted. "apt-get install scre
> en'


Nice description of Screen here.

=========
-= Cloud 9 =-

Steve Wrote:
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> I made this process immensely more tolerable by using "screen" to put it in the background to prevent it from being interupted. "apt-get install scre en'
>
Exactly. That's one of many reasons I use screen utility.

I don't remember if using the following approach will achieve the same thing. My preference choice of shell to do this is tcsh shell. You can use any shells of your choice.

1. ssh into another Linux desktop computer, issue something like this (make >& output.txt &) to compile a long program by piping the compilation output to a file and putting the process into a background. If you use a shell other than csh and/or tcsh, you will need to find out how it does the pipe.
2. Then, use tail -f output.txt to see the compilation status.
3. The above two steps can also be combined into a single command line, i.e. (a) make >& output.txt & tail -f output.txt, (b) touch output.txt;make >& output.txt & tail -f output.txt, or (c) make >& output.txt & sleep 1;tail -f output.txt. The (a) approach assumes the existence of output.txt. If the output.txt never exists, use either later two approaches, (b) or (c). The (b) approach will create an empty output.txt file prior the make >& output.txt & tail -f output.txt. The (c) approach employs a sleep 1; to issue a sleep for 1 second so as to induce a 1 second delay for the next process (tail -f output.txt). Otherwise, the tail -f output.txt will fail to locate the output.txt file.
4. Once the above is done, one can log out from the ssh session. When reconnected with a new ssh session, try to use tail -f output.txt and/or less output.txtto see the compilation status. If the compilation process is still on going, I think it can be brought back using the fg [job_id] (use ps utility to list the job_id of the above make command). In this case, even though the make process is brought back to a foreground, the screen will show nothing except hanging there until the compilation gets completed. The reason is simply because the compilation output has been piped to the output.txt file.

I gave this a try on a pogoplug following twinclouds tutorial. I also used the oauth2 instructions over on dslreports. I think i successfully compiled Asterisk 14.7.2 and then used the oauth2 script for google voice. When I run asterisk and enter the CLI, it shows me successfully connected to the obi under 'sip show peers'. When I try to dial out I get this error showing up in the CLI:

[Nov 22 08:44:19] WARNING[5477][C-00000001]: channel.c:6261 ast_request: No channel type registered for 'Motif'
[Nov 22 08:44:19] WARNING[5477][C-00000001]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'Motif' (cause 66 - Channel not implemented)

Not really sure what that means.