[Asterisk-Users] Announcement: FOP 0.26 released

[asterisk at frameweb.it]

I checked the online documentation
everything seems again correct.
So I tryed to start  ./op_server.pl with debug =7
The output seems to be correct, ie when a call arrive it is reported in the
output; if i call from 557 the extension 567, I see a lot of these "rooms"

127.0.0.1       <- Event: Newcallerid
127.0.0.1       <- Privilege: call,all
127.0.0.1       <- Channel: SIP/557-7907
127.0.0.1       <- CallerID: 557
127.0.0.1       <- CallerIDName: SNOM Andrea Lanza
127.0.0.1       <- Uniqueid: 1147786308.115
127.0.0.1       <- CID-CallingPres: 0 (Presentation Allowed, Not Screened)
127.0.0.1       <- Server: 0

127.0.0.1       <- Event: Newexten
127.0.0.1       <- Privilege: call,all
127.0.0.1       <- Channel: SIP/557-7907
127.0.0.1       <- Context: macro-user-callerid
127.0.0.1       <- Extension: s
127.0.0.1       <- Priority: 9
127.0.0.1       <- Application: NoOp
127.0.0.1       <- AppData: Using CallerID "SNOM Andrea Lanza" <557>
127.0.0.1       <- Uniqueid: 1147786308.115
127.0.0.1       <- Server: 0

So I start thinking that the problem is the flash player on the browser !!!
Another strange thing is that if I turn off op_server.pl, the flash player
still shows the same thing....I think it sounds not good !!!

I installed Flash Player 8; I will try to understand how to remove it and
how to reinstall


Andrea



                                                                           
             "Nicolás Gudiño"                                              
             <asternic at gmail.c                                             
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             16/05/2006 09.47          Re: [Asterisk-Users] Announcement:  
                                       FOP 0.26 released                   
                                                                           
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> I am using amp since lasto August, and I am happy with it and its new
> version FreePBX
>
> Unfortunately, in all the asterisk servers I installed so far (about 10)
I
> was never able to make FOP correctly running.
>
> I see the extensions, I see the queue, sometimes I also see the trunks,
if
> they are zap or iax or sip.
>
> But I ever saw a custom misdn trunk, or  an extensions "speaking", nor I
> obviously succeded in passing calls.
>
> So, of course, I am sistematically doing something wrong, but all the
other
> things work fine.
>
> I checked all the README and so on, and everithing seems to be OK.

Well, FOP is a standalone product. FreePBX (former AMP) has a script
that automatically generates FOP's config files (as well as Asterisk
conf files).

If AMP config generator is not good enough for your requirements, you
should edit the config files manually. You have the *_custom.cfg files
in FreePBX to do just that.

Some channel drivers have their "particular" way of working, like
OH323 and MiSDN.

For monitoring MiSDN channels you will have to use CLID buttons, you
can use the regular channel names but it will monitor in one direction
only due to the way isdn works and how the channel names are crafted.

Read the documentation, not the readme, at http://www.asternic.org to
get an idea of the button types and how to configure them. There is
also a low traffic FOP mailing list you can subscribe from the same
page.

Regards,


--
Nicolás Gudiño
Buenos Aires - Argentina
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